diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 394d7d378dc7..841a9365d5fd 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -757,6 +757,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     model	- force the model name
     position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
     probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
+    probe_only	- Only probing and no codec initialization (default=off);
+		  Useful to check the initial codec status for debugging
     bdl_pos_adj	- Specifies the DMA IRQ timing delay in samples.
 		Passing -1 will make the driver to choose the appropriate
 		value based on the controller chip.
@@ -772,327 +774,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    See Documentation/sound/alsa/HD-Audio.txt for more details about
+    HD-audio driver.
+
     Each codec may have a model table for different configurations.
     If your machine isn't listed there, the default (usually minimal)
     configuration is set up.  You can pass "model=<name>" option to
     specify a certain model in such a case.  There are different
-    models depending on the codec chip.
-
-	  Model name	Description
-	  ----------    -----------
-	ALC880
-	  3stack	3-jack in back and a headphone out
-	  3stack-digout	3-jack in back, a HP out and a SPDIF out
-	  5stack	5-jack in back, 2-jack in front
-	  5stack-digout	5-jack in back, 2-jack in front, a SPDIF out
-	  6stack	6-jack in back, 2-jack in front
-	  6stack-digout	6-jack with a SPDIF out
-	  w810		3-jack
-	  z71v		3-jack (HP shared SPDIF)
-	  asus		3-jack (ASUS Mobo)
-	  asus-w1v	ASUS W1V
-	  asus-dig	ASUS with SPDIF out
-	  asus-dig2	ASUS with SPDIF out (using GPIO2)
-	  uniwill	3-jack
-	  fujitsu	Fujitsu Laptops (Pi1536)
-	  F1734		2-jack
-	  lg		LG laptop (m1 express dual)
-	  lg-lw		LG LW20/LW25 laptop
-	  tcl		TCL S700
-	  clevo		Clevo laptops (m520G, m665n)
-	  medion	Medion Rim 2150
-	  test		for testing/debugging purpose, almost all controls can be
-			adjusted.  Appearing only when compiled with
-			$CONFIG_SND_DEBUG=y
-	  auto		auto-config reading BIOS (default)
-
-	ALC260
-	  hp		HP machines
-	  hp-3013	HP machines (3013-variant)
-	  hp-dc7600	HP DC7600
-	  fujitsu	Fujitsu S7020
-	  acer		Acer TravelMate
-	  will		Will laptops (PB V7900)
-	  replacer	Replacer 672V
-	  basic		fixed pin assignment (old default model)
-	  test		for testing/debugging purpose, almost all controls can
-			adjusted.  Appearing only when compiled with
-			$CONFIG_SND_DEBUG=y
-	  auto		auto-config reading BIOS (default)
-
-	ALC262
-	  fujitsu	Fujitsu Laptop
-	  hp-bpc	HP xw4400/6400/8400/9400 laptops
-	  hp-bpc-d7000	HP BPC D7000
-	  hp-tc-t5735	HP Thin Client T5735
-	  hp-rp5700	HP RP5700
-	  benq		Benq ED8
-	  benq-t31	Benq T31
-	  hippo		Hippo (ATI) with jack detection, Sony UX-90s
-	  hippo_1	Hippo (Benq) with jack detection
-	  sony-assamd	Sony ASSAMD
-	  toshiba-s06	Toshiba S06
-	  toshiba-rx1	Toshiba RX1
-	  ultra		Samsung Q1 Ultra Vista model
-	  lenovo-3000	Lenovo 3000 y410
-	  nec		NEC Versa S9100
-	  basic		fixed pin assignment w/o SPDIF
-	  auto		auto-config reading BIOS (default)
-
-	ALC267/268
-	  quanta-il1	Quanta IL1 mini-notebook
-	  3stack	3-stack model
-	  toshiba	Toshiba A205
-	  acer		Acer laptops
-	  acer-aspire	Acer Aspire One
-	  dell		Dell OEM laptops (Vostro 1200)
-	  zepto		Zepto laptops
-	  test		for testing/debugging purpose, almost all controls can
-			adjusted.  Appearing only when compiled with
-			$CONFIG_SND_DEBUG=y
-	  auto		auto-config reading BIOS (default)
-
-	ALC269
-	  basic		Basic preset
-	  quanta	Quanta FL1
-	  eeepc-p703	ASUS Eeepc P703 P900A
-	  eeepc-p901	ASUS Eeepc P901 S101
-
-	ALC662/663
-	  3stack-dig	3-stack (2-channel) with SPDIF
-	  3stack-6ch	 3-stack (6-channel)
-	  3stack-6ch-dig 3-stack (6-channel) with SPDIF
-	  6stack-dig	 6-stack with SPDIF
-	  lenovo-101e	 Lenovo laptop
-	  eeepc-p701	ASUS Eeepc P701
-	  eeepc-ep20	ASUS Eeepc EP20
-	  ecs		ECS/Foxconn mobo
-	  m51va		ASUS M51VA
-	  g71v		ASUS G71V
-	  h13		ASUS H13
-	  g50v		ASUS G50V
-	  asus-mode1	ASUS
-	  asus-mode2	ASUS
-	  asus-mode3	ASUS
-	  asus-mode4	ASUS
-	  asus-mode5	ASUS
-	  asus-mode6	ASUS
-	  auto		auto-config reading BIOS (default)
-
-	ALC882/885
-	  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
-	  mbp3		Macbook Pro rev3
-	  imac24	iMac 24'' with jack detection
-	  w2jc		ASUS W2JC
-	  auto		auto-config reading BIOS (default)
-
-	ALC883/888
-	  3stack-dig	3-jack with SPDIF I/O
-	  6stack-dig	6-jack digital with SPDIF I/O
-	  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
-	  medion	Medion Laptops
-	  medion-md2	Medion MD2
-	  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)
-	  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-m720	Clevo M720 laptop series
-	  fujitsu-pi2515 Fujitsu AMILO Pi2515
-	  3stack-6ch-intel Intel DG33* boards
-	  auto		auto-config reading BIOS (default)
-
-	ALC861/660
-	  3stack	3-jack
-	  3stack-dig	3-jack with SPDIF I/O
-	  6stack-dig	6-jack with SPDIF I/O
-	  3stack-660	3-jack (for ALC660)
-	  uniwill-m31	Uniwill M31 laptop
-	  toshiba	Toshiba laptop support
-	  asus		Asus laptop support
-	  asus-laptop	ASUS F2/F3 laptops
-	  auto		auto-config reading BIOS (default)
-
-	ALC861VD/660VD
-	  3stack	3-jack
-	  3stack-dig	3-jack with SPDIF OUT
-	  6stack-dig	6-jack with SPDIF OUT
-	  3stack-660	3-jack (for ALC660VD)
-	  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
-	  lenovo	Lenovo 3000 C200
-	  dallas	Dallas laptops
-	  hp		HP TX1000
-	  auto		auto-config reading BIOS (default)
-
-	CMI9880
-	  minimal	3-jack in back
-	  min_fp	3-jack in back, 2-jack in front
-	  full		6-jack in back, 2-jack in front
-	  full_dig	6-jack in back, 2-jack in front, SPDIF I/O
-	  allout	5-jack in back, 2-jack in front, SPDIF out
-	  auto		auto-config reading BIOS (default)
-
-	AD1882 / AD1882A
-	  3stack	3-stack mode (default)
-	  6stack	6-stack mode
-
-	AD1884A / AD1883 / AD1984A / AD1984B
-	  desktop	3-stack desktop (default)
-	  laptop	laptop with HP jack sensing
-	  mobile	mobile devices with HP jack sensing
-	  thinkpad	Lenovo Thinkpad X300
-
-	AD1884
-	  N/A
-
-	AD1981
-	  basic		3-jack (default)
-	  hp		HP nx6320
-	  thinkpad	Lenovo Thinkpad T60/X60/Z60
-	  toshiba	Toshiba U205
-
-	AD1983
-	  N/A
-
-	AD1984
-	  basic		default configuration
-	  thinkpad	Lenovo Thinkpad T61/X61
-	  dell		Dell T3400
-
-	AD1986A
-	  6stack	6-jack, separate surrounds (default)
-	  3stack	3-stack, shared surrounds
-	  laptop	2-channel only (FSC V2060, Samsung M50)
-	  laptop-eapd	2-channel with EAPD (Samsung R65, ASUS A6J)
-	  laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
-	  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
-
-	AD1988/AD1988B/AD1989A/AD1989B
-	  6stack	6-jack
-	  6stack-dig	ditto with SPDIF
-	  3stack	3-jack
-	  3stack-dig	ditto with SPDIF
-	  laptop	3-jack with hp-jack automute
-	  laptop-dig	ditto with SPDIF
-	  auto		auto-config reading BIOS (default)
-	
-	Conexant 5045
-	  laptop-hpsense    Laptop with HP sense (old model laptop)
-	  laptop-micsense   Laptop with Mic sense (old model fujitsu)
-	  laptop-hpmicsense Laptop with HP and Mic senses
-	  benq		Benq R55E
-	  test		for testing/debugging purpose, almost all controls
-			can be adjusted.  Appearing only when compiled with
-			$CONFIG_SND_DEBUG=y
-
-	Conexant 5047
-	  laptop	Basic Laptop config 
-	  laptop-hp	Laptop config for some HP models (subdevice 30A5)
-	  laptop-eapd	Laptop config with EAPD support
-	  test		for testing/debugging purpose, almost all controls
-			can be adjusted.  Appearing only when compiled with
-			$CONFIG_SND_DEBUG=y
-
-	Conexant 5051
-	  laptop	Basic Laptop config (default)
-	  hp		HP Spartan laptop
-
-	STAC9200
-	  ref		Reference board
-	  dell-d21	Dell (unknown)
-	  dell-d22	Dell (unknown)
-	  dell-d23	Dell (unknown)
-	  dell-m21	Dell Inspiron 630m, Dell Inspiron 640m
-	  dell-m22	Dell Latitude D620, Dell Latitude D820
-	  dell-m23	Dell XPS M1710, Dell Precision M90
-	  dell-m24	Dell Latitude 120L
-	  dell-m25	Dell Inspiron E1505n
-	  dell-m26	Dell Inspiron 1501
-	  dell-m27	Dell Inspiron E1705/9400
-	  gateway	Gateway laptops with EAPD control
-	  panasonic	Panasonic CF-74
-
-	STAC9205/9254
-	  ref		Reference board
-	  dell-m42	Dell (unknown)
-	  dell-m43	Dell Precision
-	  dell-m44	Dell Inspiron
-
-	STAC9220/9221
-	  ref		Reference board
-	  3stack	D945 3stack
-	  5stack	D945 5stack + SPDIF
-	  intel-mac-v1	Intel Mac Type 1
-	  intel-mac-v2	Intel Mac Type 2
-	  intel-mac-v3	Intel Mac Type 3
-	  intel-mac-v4	Intel Mac Type 4
-	  intel-mac-v5	Intel Mac Type 5
-	  intel-mac-auto Intel Mac (detect type according to subsystem id)
-	  macmini	Intel Mac Mini (equivalent with type 3)
-	  macbook	Intel Mac Book (eq. type 5)
-	  macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
-	  macbook-pro	Intel Mac Book Pro 2nd generation (eq. type 3)
-	  imac-intel	Intel iMac (eq. type 2)
-	  imac-intel-20	Intel iMac (newer version) (eq. type 3)
-	  dell-d81	Dell (unknown)
-	  dell-d82	Dell (unknown)
-	  dell-m81	Dell (unknown)
-	  dell-m82	Dell XPS M1210
-
-	STAC9202/9250/9251
-	  ref		Reference board, base config
-	  m2-2		Some Gateway MX series laptops
-	  m6		Some Gateway NX series laptops
-	  pa6		Gateway NX860 series
-
-	STAC9227/9228/9229/927x
-	  ref		Reference board
-	  ref-no-jd	Reference board without HP/Mic jack detection
-	  3stack	D965 3stack
-	  5stack	D965 5stack + SPDIF
-	  dell-3stack	Dell Dimension E520
-	  dell-bios	Fixes with Dell BIOS setup
-
-	STAC92HD71B*
-	  ref		Reference board
-	  dell-m4-1	Dell desktops
-	  dell-m4-2	Dell desktops
-	  dell-m4-3	Dell desktops
-
-	STAC92HD73*
-	  ref		Reference board
-	  no-jd		BIOS setup but without jack-detection
-	  dell-m6-amic	Dell desktops/laptops with analog mics
-	  dell-m6-dmic	Dell desktops/laptops with digital mics
-	  dell-m6	Dell desktops/laptops with both type of mics
-
-	STAC9872
-	  vaio		Setup for VAIO FE550G/SZ110
-	  vaio-ar Setup for VAIO AR
+    models depending on the codec chip.  The list of available models
+    is found in HD-Audio-Models.txt
 
     The model name "genric" is treated as a special case.  When this
     model is given, the driver uses the generic codec parser without
     "codec-patch".  It's sometimes good for testing and debugging.
 
     If the default configuration doesn't work and one of the above
-    matches with your device, report it together with the PCI
-    subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
+    matches with your device, report it together with alsa-info.sh
+    output (with --no-upload option) to kernel bugzilla or alsa-devel
     ML (see the section "Links and Addresses").
 
     power_save and power_save_controller options are for power-saving
@@ -1652,7 +1350,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     * AuzenTech X-Meridian
     * Bgears b-Enspirer
     * Club3D Theatron DTS
-    * HT-Omega Claro
+    * HT-Omega Claro (plus)
+    * HT-Omega Claro halo (XT)
     * Razer Barracuda AC-1
     * Sondigo Inferno
 
@@ -2409,8 +2108,11 @@ Links and Addresses
   ALSA project homepage
        http://www.alsa-project.org
 
-  ALSA Bug Tracking System
-       https://bugtrack.alsa-project.org/bugs/
+  Kernel Bugzilla
+       http://bugzilla.kernel.org/
 
   ALSA Developers ML
        mailto:alsa-devel@alsa-project.org
+
+  alsa-info.sh script
+       http://www.alsa-project.org/alsa-info.sh
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
new file mode 100644
index 000000000000..4b7ac21ea9eb
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -0,0 +1,348 @@
+  Model name	Description
+  ----------    -----------
+ALC880
+======
+  3stack	3-jack in back and a headphone out
+  3stack-digout	3-jack in back, a HP out and a SPDIF out
+  5stack	5-jack in back, 2-jack in front
+  5stack-digout	5-jack in back, 2-jack in front, a SPDIF out
+  6stack	6-jack in back, 2-jack in front
+  6stack-digout	6-jack with a SPDIF out
+  w810		3-jack
+  z71v		3-jack (HP shared SPDIF)
+  asus		3-jack (ASUS Mobo)
+  asus-w1v	ASUS W1V
+  asus-dig	ASUS with SPDIF out
+  asus-dig2	ASUS with SPDIF out (using GPIO2)
+  uniwill	3-jack
+  fujitsu	Fujitsu Laptops (Pi1536)
+  F1734		2-jack
+  lg		LG laptop (m1 express dual)
+  lg-lw		LG LW20/LW25 laptop
+  tcl		TCL S700
+  clevo		Clevo laptops (m520G, m665n)
+  medion	Medion Rim 2150
+  test		for testing/debugging purpose, almost all controls can be
+		adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+  auto		auto-config reading BIOS (default)
+
+ALC260
+======
+  hp		HP machines
+  hp-3013	HP machines (3013-variant)
+  hp-dc7600	HP DC7600
+  fujitsu	Fujitsu S7020
+  acer		Acer TravelMate
+  will		Will laptops (PB V7900)
+  replacer	Replacer 672V
+  basic		fixed pin assignment (old default model)
+  test		for testing/debugging purpose, almost all controls can
+		adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+  auto		auto-config reading BIOS (default)
+
+ALC262
+======
+  fujitsu	Fujitsu Laptop
+  hp-bpc	HP xw4400/6400/8400/9400 laptops
+  hp-bpc-d7000	HP BPC D7000
+  hp-tc-t5735	HP Thin Client T5735
+  hp-rp5700	HP RP5700
+  benq		Benq ED8
+  benq-t31	Benq T31
+  hippo		Hippo (ATI) with jack detection, Sony UX-90s
+  hippo_1	Hippo (Benq) with jack detection
+  sony-assamd	Sony ASSAMD
+  toshiba-s06	Toshiba S06
+  toshiba-rx1	Toshiba RX1
+  ultra		Samsung Q1 Ultra Vista model
+  lenovo-3000	Lenovo 3000 y410
+  nec		NEC Versa S9100
+  basic		fixed pin assignment w/o SPDIF
+  auto		auto-config reading BIOS (default)
+
+ALC267/268
+==========
+  quanta-il1	Quanta IL1 mini-notebook
+  3stack	3-stack model
+  toshiba	Toshiba A205
+  acer		Acer laptops
+  acer-dmic	Acer laptops with digital-mic
+  acer-aspire	Acer Aspire One
+  dell		Dell OEM laptops (Vostro 1200)
+  zepto		Zepto laptops
+  test		for testing/debugging purpose, almost all controls can
+		adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+  auto		auto-config reading BIOS (default)
+
+ALC269
+======
+  basic		Basic preset
+  quanta	Quanta FL1
+  eeepc-p703	ASUS Eeepc P703 P900A
+  eeepc-p901	ASUS Eeepc P901 S101
+  fujitsu	FSC Amilo
+  auto		auto-config reading BIOS (default)
+
+ALC662/663
+==========
+  3stack-dig	3-stack (2-channel) with SPDIF
+  3stack-6ch	 3-stack (6-channel)
+  3stack-6ch-dig 3-stack (6-channel) with SPDIF
+  6stack-dig	 6-stack with SPDIF
+  lenovo-101e	 Lenovo laptop
+  eeepc-p701	ASUS Eeepc P701
+  eeepc-ep20	ASUS Eeepc EP20
+  ecs		ECS/Foxconn mobo
+  m51va		ASUS M51VA
+  g71v		ASUS G71V
+  h13		ASUS H13
+  g50v		ASUS G50V
+  asus-mode1	ASUS
+  asus-mode2	ASUS
+  asus-mode3	ASUS
+  asus-mode4	ASUS
+  asus-mode5	ASUS
+  asus-mode6	ASUS
+  auto		auto-config reading BIOS (default)
+
+ALC882/885
+==========
+  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
+  mbp3		Macbook Pro rev3
+  imac24	iMac 24'' with jack detection
+  w2jc		ASUS W2JC
+  auto		auto-config reading BIOS (default)
+
+ALC883/888
+==========
+  3stack-dig	3-jack with SPDIF I/O
+  6stack-dig	6-jack digital with SPDIF I/O
+  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
+  medion	Medion Laptops
+  medion-md2	Medion MD2
+  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)
+  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-m720	Clevo M720 laptop series
+  fujitsu-pi2515 Fujitsu AMILO Pi2515
+  fujitsu-xa3530 Fujitsu AMILO XA3530
+  3stack-6ch-intel Intel DG33* boards
+  auto		auto-config reading BIOS (default)
+
+ALC861/660
+==========
+  3stack	3-jack
+  3stack-dig	3-jack with SPDIF I/O
+  6stack-dig	6-jack with SPDIF I/O
+  3stack-660	3-jack (for ALC660)
+  uniwill-m31	Uniwill M31 laptop
+  toshiba	Toshiba laptop support
+  asus		Asus laptop support
+  asus-laptop	ASUS F2/F3 laptops
+  auto		auto-config reading BIOS (default)
+
+ALC861VD/660VD
+==============
+  3stack	3-jack
+  3stack-dig	3-jack with SPDIF OUT
+  6stack-dig	6-jack with SPDIF OUT
+  3stack-660	3-jack (for ALC660VD)
+  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
+  lenovo	Lenovo 3000 C200
+  dallas	Dallas laptops
+  hp		HP TX1000
+  asus-v1s	ASUS V1Sn
+  auto		auto-config reading BIOS (default)
+
+CMI9880
+=======
+  minimal	3-jack in back
+  min_fp	3-jack in back, 2-jack in front
+  full		6-jack in back, 2-jack in front
+  full_dig	6-jack in back, 2-jack in front, SPDIF I/O
+  allout	5-jack in back, 2-jack in front, SPDIF out
+  auto		auto-config reading BIOS (default)
+
+AD1882 / AD1882A
+================
+  3stack	3-stack mode (default)
+  6stack	6-stack mode
+
+AD1884A / AD1883 / AD1984A / AD1984B
+====================================
+  desktop	3-stack desktop (default)
+  laptop	laptop with HP jack sensing
+  mobile	mobile devices with HP jack sensing
+  thinkpad	Lenovo Thinkpad X300
+
+AD1884
+======
+  N/A
+
+AD1981
+======
+  basic		3-jack (default)
+  hp		HP nx6320
+  thinkpad	Lenovo Thinkpad T60/X60/Z60
+  toshiba	Toshiba U205
+
+AD1983
+======
+  N/A
+
+AD1984
+======
+  basic		default configuration
+  thinkpad	Lenovo Thinkpad T61/X61
+  dell		Dell T3400
+
+AD1986A
+=======
+  6stack	6-jack, separate surrounds (default)
+  3stack	3-stack, shared surrounds
+  laptop	2-channel only (FSC V2060, Samsung M50)
+  laptop-eapd	2-channel with EAPD (ASUS A6J)
+  laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
+  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
+  samsung	2-channel with EAPD (Samsung R65)
+
+AD1988/AD1988B/AD1989A/AD1989B
+==============================
+  6stack	6-jack
+  6stack-dig	ditto with SPDIF
+  3stack	3-jack
+  3stack-dig	ditto with SPDIF
+  laptop	3-jack with hp-jack automute
+  laptop-dig	ditto with SPDIF
+  auto		auto-config reading BIOS (default)
+
+Conexant 5045
+=============
+  laptop-hpsense    Laptop with HP sense (old model laptop)
+  laptop-micsense   Laptop with Mic sense (old model fujitsu)
+  laptop-hpmicsense Laptop with HP and Mic senses
+  benq		Benq R55E
+  test		for testing/debugging purpose, almost all controls
+		can be adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+
+Conexant 5047
+=============
+  laptop	Basic Laptop config 
+  laptop-hp	Laptop config for some HP models (subdevice 30A5)
+  laptop-eapd	Laptop config with EAPD support
+  test		for testing/debugging purpose, almost all controls
+		can be adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+
+Conexant 5051
+=============
+  laptop	Basic Laptop config (default)
+  hp		HP Spartan laptop
+
+STAC9200
+========
+  ref		Reference board
+  dell-d21	Dell (unknown)
+  dell-d22	Dell (unknown)
+  dell-d23	Dell (unknown)
+  dell-m21	Dell Inspiron 630m, Dell Inspiron 640m
+  dell-m22	Dell Latitude D620, Dell Latitude D820
+  dell-m23	Dell XPS M1710, Dell Precision M90
+  dell-m24	Dell Latitude 120L
+  dell-m25	Dell Inspiron E1505n
+  dell-m26	Dell Inspiron 1501
+  dell-m27	Dell Inspiron E1705/9400
+  gateway	Gateway laptops with EAPD control
+  panasonic	Panasonic CF-74
+
+STAC9205/9254
+=============
+  ref		Reference board
+  dell-m42	Dell (unknown)
+  dell-m43	Dell Precision
+  dell-m44	Dell Inspiron
+
+STAC9220/9221
+=============
+  ref		Reference board
+  3stack	D945 3stack
+  5stack	D945 5stack + SPDIF
+  intel-mac-v1	Intel Mac Type 1
+  intel-mac-v2	Intel Mac Type 2
+  intel-mac-v3	Intel Mac Type 3
+  intel-mac-v4	Intel Mac Type 4
+  intel-mac-v5	Intel Mac Type 5
+  intel-mac-auto Intel Mac (detect type according to subsystem id)
+  macmini	Intel Mac Mini (equivalent with type 3)
+  macbook	Intel Mac Book (eq. type 5)
+  macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
+  macbook-pro	Intel Mac Book Pro 2nd generation (eq. type 3)
+  imac-intel	Intel iMac (eq. type 2)
+  imac-intel-20	Intel iMac (newer version) (eq. type 3)
+  dell-d81	Dell (unknown)
+  dell-d82	Dell (unknown)
+  dell-m81	Dell (unknown)
+  dell-m82	Dell XPS M1210
+
+STAC9202/9250/9251
+==================
+  ref		Reference board, base config
+  m2-2		Some Gateway MX series laptops
+  m6		Some Gateway NX series laptops
+  pa6		Gateway NX860 series
+
+STAC9227/9228/9229/927x
+=======================
+  ref		Reference board
+  ref-no-jd	Reference board without HP/Mic jack detection
+  3stack	D965 3stack
+  5stack	D965 5stack + SPDIF
+  dell-3stack	Dell Dimension E520
+  dell-bios	Fixes with Dell BIOS setup
+
+STAC92HD71B*
+============
+  ref		Reference board
+  dell-m4-1	Dell desktops
+  dell-m4-2	Dell desktops
+  dell-m4-3	Dell desktops
+
+STAC92HD73*
+===========
+  ref		Reference board
+  no-jd		BIOS setup but without jack-detection
+  dell-m6-amic	Dell desktops/laptops with analog mics
+  dell-m6-dmic	Dell desktops/laptops with digital mics
+  dell-m6	Dell desktops/laptops with both type of mics
+
+STAC92HD83*
+===========
+  ref		Reference board
+
+STAC9872
+========
+  vaio		Setup for VAIO FE550G/SZ110
+  vaio-ar Setup for VAIO AR
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
new file mode 100644
index 000000000000..8d68fff71839
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -0,0 +1,577 @@
+MORE NOTES ON HD-AUDIO DRIVER
+=============================
+					Takashi Iwai <tiwai@suse.de>
+
+
+GENERAL
+-------
+
+HD-audio is the new standard on-board audio component on modern PCs
+after AC97.  Although Linux has been supporting HD-audio since long
+time ago, there are often problems with new machines.  A part of the
+problem is broken BIOS, and the rest is the driver implementation.
+This document explains the brief trouble-shooting and debugging
+methods for the	HD-audio hardware.
+
+The HD-audio component consists of two parts: the controller chip and 
+the codec chips on the HD-audio bus.  Linux provides a single driver
+for all controllers, snd-hda-intel.  Although the driver name contains
+a word of a well-known harware vendor, it's not specific to it but for
+all controller chips by other companies.  Since the HD-audio
+controllers are supposed to be compatible, the single snd-hda-driver
+should work in most cases.  But, not surprisingly, there are known
+bugs and issues specific to each controller type.  The snd-hda-intel
+driver has a bunch of workarounds for these as described below.
+
+A controller may have multiple codecs.  Usually you have one audio
+codec and optionally one modem codec.  In theory, there might be
+multiple audio codecs, e.g. for analog and digital outputs, and the
+driver might not work properly because of conflict of mixer elements.
+This should be fixed in future if such hardware really exists.
+
+The snd-hda-intel driver has several different codec parsers depending
+on the codec.  It has a generic parser as a fallback, but this
+functionality is fairly limited until now.  Instead of the generic
+parser, usually the codec-specific parser (coded in patch_*.c) is used
+for the codec-specific implementations.  The details about the
+codec-specific problems are explained in the later sections.
+
+If you are interested in the deep debugging of HD-audio, read the
+HD-audio specification at first.  The specification is found on
+Intel's web page, for example:
+
+- http://www.intel.com/standards/hdaudio/
+
+
+HD-AUDIO CONTROLLER
+-------------------
+
+DMA-Position Problem
+~~~~~~~~~~~~~~~~~~~~
+The most common problem of the controller is the inaccurate DMA
+pointer reporting.  The DMA pointer for playback and capture can be
+read in two ways, either via a LPIB register or via a position-buffer
+map.  As default the driver tries to read from the io-mapped
+position-buffer, and falls back to LPIB if the position-buffer appears
+dead.  However, this detection isn't perfect on some devices.  In such
+a case, you can change the default method via `position_fix` option.
+
+`position_fix=1` means to use LPIB method explicitly.
+`position_fix=2` means to use the position-buffer.  0 is the default
+value, the automatic check and fallback to LPIB as described in the
+above.  If you get a problem of repeated sounds, this option might
+help.
+
+In addition to that, every controller is known to be broken regarding
+the wake-up timing.  It wakes up a few samples before actually
+processing the data on the buffer.  This caused a lot of problems, for
+example, with ALSA dmix or JACK.  Since 2.6.27 kernel, the driver puts
+an artificial delay to the wake up timing.  This delay is controlled
+via `bdl_pos_adj` option. 
+
+When `bdl_pos_adj` is a negative value (as default), it's assigned to
+an appropriate value depending on the controller chip.  For Intel
+chips, it'd be 1 while it'd be 32 for others.  Usually this works.
+Only in case it doesn't work and you get warning messages, you should
+change this parameter to other values.
+
+
+Codec-Probing Problem
+~~~~~~~~~~~~~~~~~~~~~
+A less often but a more severe problem is the codec probing.  When
+BIOS reports the available codec slots wrongly, the driver gets
+confused and tries to access the non-existing codec slot.  This often
+results in the total screw-up, and destructs the further communication
+with the codec chips.  The symptom appears usually as error messages
+like:
+------------------------------------------------------------------------
+  hda_intel: azx_get_response timeout, switching to polling mode:
+        last cmd=0x12345678
+  hda_intel: azx_get_response timeout, switching to single_cmd mode:
+        last cmd=0x12345678
+------------------------------------------------------------------------
+
+The first line is a warning, and this is usually relatively harmless.
+It means that the codec response isn't notified via an IRQ.  The
+driver uses explicit polling method to read the response.  It gives
+very slight CPU overhead, but you'd unlikely notice it.
+
+The second line is, however, a fatal error.  If this happens, usually
+it means that something is really wrong.  Most likely you are
+accessing a non-existing codec slot.
+
+Thus, if the second error message appears, try to narrow the probed
+codec slots via `probe_mask` option.  It's a bitmask, and each bit
+corresponds to the codec slot.  For example, to probe only the first
+slot, pass `probe_mask=1`.  For the first and the third slots, pass
+`probe_mask=5` (where 5 = 1 | 4), and so on.
+
+Since 2.6.29 kernel, the driver has a more robust probing method, so
+this error might happen rarely, though.
+
+
+Interrupt Handling
+~~~~~~~~~~~~~~~~~~
+In rare but some cases, the interrupt isn't properly handled as
+default.  You would notice this by the DMA transfer error reported by
+ALSA PCM core, for example.  Using MSI might help in such a case.
+Pass `enable_msi=1` option for enabling MSI.
+
+
+HD-AUDIO CODEC
+--------------
+
+Model Option
+~~~~~~~~~~~~
+The most common problem regarding the HD-audio driver is the
+unsupported codec features or the mismatched device configuration.
+Most of codec-specific code has several preset models, either to
+override the BIOS setup or to provide more comprehensive features.
+
+The driver checks PCI SSID and looks through the static configuration
+table until any matching entry is found.  If you have a new machine,
+you may see a message like below:
+------------------------------------------------------------------------
+    hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...
+------------------------------------------------------------------------
+Even if you see such a message, DON'T PANIC.  Take a deep breath and
+keep your towel.  First of all, it's an informational message, no
+warning, no error.  This means that the PCI SSID of your device isn't
+listed in the known preset model (white-)list.  But, this doesn't mean
+that the driver is broken.  Many codec-drivers provide the automatic
+configuration mechanism based on the BIOS setup.
+
+The HD-audio codec has usually "pin" widgets, and BIOS sets the default
+configuration of each pin, which indicates the location, the
+connection type, the jack color, etc.  The HD-audio driver can guess
+the right connection judging from these default configuration values.
+However -- some codec-support codes, such as patch_analog.c, don't
+support the automatic probing (yet as of 2.6.28).  And, BIOS is often,
+yes, pretty often broken.  It sets up wrong values and screws up the
+driver.
+
+The preset model is provided basically to overcome such a situation.
+When the matching preset model is found in the white-list, the driver
+assumes the static configuration of that preset and builds the mixer
+elements and PCM streams based on the static information.  Thus, if
+you have a newer machine with a slightly different PCI SSID from the
+existing one, you may have a good chance to re-use the same model.
+You can pass the `model` option to specify the preset model instead of
+PCI SSID look-up.
+
+What `model` option values are available depends on the codec chip.
+Check your codec chip from the codec proc file (see "Codec Proc-File"
+section below).  It will show the vendor/product name of your codec
+chip.  Then, see Documentation/sound/alsa/HD-Audio-Modelstxt file,
+the section of HD-audio driver.  You can find a list of codecs
+and `model` options belonging to each codec.  For example, for Realtek
+ALC262 codec chip, pass `model=ultra` for devices that are compatible
+with Samsung Q1 Ultra.
+
+Thus, the first thing you can do for any brand-new, unsupported and
+non-working HD-audio hardware is to check HD-audio codec and several
+different `model` option values.  If you have a luck, some of them
+might suit with your device well.
+
+Some codecs such as ALC880 have a special model option `model=test`.
+This configures the driver to provide as many mixer controls as
+possible for every single pin feature except for the unsolicited
+events (and maybe some other specials).  Adjust each mixer element and
+try the I/O in the way of trial-and-error until figuring out the whole
+I/O pin mappings.
+
+Note that `model=generic` has a special meaning.  It means to use the
+generic parser regardless of the codec.  Usually the codec-specific
+parser is much better than the generic parser (as now).  Thus this
+option is more about the debugging purpose.
+
+
+Speaker and Headphone Output
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+One of the most frequent (and obvious) bugs with HD-audio is the
+silent output from either or both of a built-in speaker and a
+headphone jack.  In general, you should try a headphone output at
+first.  A speaker output often requires more additional controls like
+the external amplifier bits.  Thus a headphone output has a slightly
+better chance.
+
+Before making a bug report, double-check whether the mixer is set up
+correctly.  The recent version of snd-hda-intel driver provides mostly
+"Master" volume control as well as "Front" volume (where Front
+indicates the front-channels).  In addition, there can be individual
+"Headphone" and "Speaker" controls.
+
+Ditto for the speaker output.  There can be "External Amplifier"
+switch on some codecs.  Turn on this if present.
+
+Another related problem is the automatic mute of speaker output by
+headphone plugging.  This feature is implemented in most cases, but
+not on every preset model or codec-support code.
+
+In anyway, try a different model option if you have such a problem.
+Some other models may match better and give you more matching
+functionality.  If none of the available models works, send a bug
+report.  See the bug report section for details.
+
+If you are masochistic enough to debug the driver problem, note the
+following:
+
+- The speaker (and the headphone, too) output often requires the
+  external amplifier.  This can be set usually via EAPD verb or a
+  certain GPIO.  If the codec pin supports EAPD, you have a better
+  chance via SET_EAPD_BTL verb (0x70c).  On others, GPIO pin (mostly
+  it's either GPIO0 or GPIO1) may turn on/off EAPD.
+- Some Realtek codecs require special vendor-specific coefficients to
+  turn on the amplifier.  See patch_realtek.c.
+- IDT codecs may have extra power-enable/disable controls on each
+  analog pin.  See patch_sigmatel.c.
+- Very rare but some devices don't accept the pin-detection verb until
+  triggered.  Issuing GET_PIN_SENSE verb (0xf09) may result in the
+  codec-communication stall.  Some examples are found in
+  patch_realtek.c.
+
+
+Capture Problems
+~~~~~~~~~~~~~~~~
+The capture problems are often because of missing setups of mixers.
+Thus, before submitting a bug report, make sure that you set up the
+mixer correctly.  For example, both "Capture Volume" and "Capture
+Switch" have to be set properly in addition to the right "Capture
+Source" or "Input Source" selection.  Some devices have "Mic Boost"
+volume or switch.
+
+When the PCM device is opened via "default" PCM (without pulse-audio
+plugin), you'll likely have "Digital Capture Volume" control as well.
+This is provided for the extra gain/attenuation of the signal in
+software, especially for the inputs without the hardware volume
+control such as digital microphones.  Unless really needed, this
+should be set to exactly 50%, corresponding to 0dB -- neither extra
+gain nor attenuation.  When you use "hw" PCM, i.e., a raw access PCM,
+this control will have no influence, though.
+
+It's known that some codecs / devices have fairly bad analog circuits,
+and the recorded sound contains a certain DC-offset.  This is no bug
+of the driver.
+
+Most of modern laptops have no analog CD-input connection.  Thus, the
+recording from CD input won't work in many cases although the driver
+provides it as the capture source.  Use CDDA instead.
+
+The automatic switching of the built-in and external mic per plugging
+is implemented on some codec models but not on every model.  Partly
+because of my laziness but mostly lack of testers.  Feel free to
+submit the improvement patch to the author.
+
+
+Direct Debugging
+~~~~~~~~~~~~~~~~
+If no model option gives you a better result, and you are a tough guy
+to fight against evil, try debugging via hitting the raw HD-audio
+codec verbs to the device.  Some tools are available: hda-emu and
+hda-analyzer.  The detailed description is found in the sections
+below.  You'd need to enable hwdep for using these tools.  See "Kernel
+Configuration" section.
+
+
+OTHER ISSUES
+------------
+
+Kernel Configuration
+~~~~~~~~~~~~~~~~~~~~
+In general, I recommend you to enable the sound debug option,
+`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not.
+This enables snd_printd() macro and others, and you'll get additional
+kernel messages at probing.
+
+In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`.  But this
+will give you far more messages.  Thus turn this on only when you are
+sure to want it.
+
+Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*`
+options.  Note that each of them corresponds to the codec chip, not
+the controller chip.  Thus, even if lspci shows the Nvidia controller,
+you may need to choose the option for other vendors.  If you are
+unsure, just select all yes.
+
+`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver.
+When this is enabled, the driver creates hardware-dependent devices
+(one per each codec), and you have a raw access to the device via
+these device files.  For example, `hwC0D2` will be created for the
+codec slot #2 of the first card (#0).  For debug-tools such as
+hda-verb and hda-analyzer, the hwdep device has to be enabled.
+Thus, it'd be better to turn this on always.
+
+`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the
+hwdep option above.  When enabled, you'll have some sysfs files under
+the corresponding hwdep directory.  See "HD-audio reconfiguration"
+section below.
+
+`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature.
+See "Power-saving" section below.
+
+
+Codec Proc-File
+~~~~~~~~~~~~~~~
+The codec proc-file is a treasure-chest for debugging HD-audio.
+It shows most of useful information of each codec widget.
+
+The proc file is located in /proc/asound/card*/codec#*, one file per
+each codec slot.  You can know the codec vendor, product id and
+names, the type of each widget, capabilities and so on.
+This file, however, doesn't show the jack sensing state, so far.  This
+is because the jack-sensing might be depending on the trigger state.
+
+This file will be picked up by the debug tools, and also it can be fed
+to the emulator as the primary codec information.  See the debug tools
+section below.
+
+This proc file can be also used to check whether the generic parser is
+used.  When the generic parser is used, the vendor/product ID name
+will appear as "Realtek ID 0262", instead of "Realtek ALC262".
+
+
+HD-Audio Reconfiguration
+~~~~~~~~~~~~~~~~~~~~~~~~
+This is an experimental feature to allow you re-configure the HD-audio
+codec dynamically without reloading the driver.  The following sysfs
+files are available under each codec-hwdep device directory (e.g. 
+/sys/class/sound/hwC0D0):
+
+vendor_id::
+  Shows the 32bit codec vendor-id hex number.  You can change the
+  vendor-id value by writing to this file.
+subsystem_id::
+  Shows the 32bit codec subsystem-id hex number.  You can change the
+  subsystem-id value by writing to this file.
+revision_id::
+  Shows the 32bit codec revision-id hex number.  You can change the
+  revision-id value by writing to this file.
+afg::
+  Shows the AFG ID.  This is read-only.
+mfg::
+  Shows the MFG ID.  This is read-only.
+name::
+  Shows the codec name string.  Can be changed by writing to this
+  file.
+modelname::
+  Shows the currently set `model` option.  Can be changed by writing
+  to this file.
+init_verbs::
+  The extra verbs to execute at initialization.  You can add a verb by
+  writing to this file.  Pass tree numbers, nid, verb and parameter.
+hints::
+  Shows hint strings for codec parsers for any use.  Right now it's
+  not used.
+reconfig::
+  Triggers the codec re-configuration.  When any value is written to
+  this file, the driver re-initialize and parses the codec tree
+  again.  All the changes done by the sysfs entries above are taken
+  into account.
+clear::
+  Resets the codec, removes the mixer elements and PCM stuff of the
+  specified codec, and clear all init verbs and hints.
+
+
+Power-Saving
+~~~~~~~~~~~~
+The power-saving is a kind of auto-suspend of the device.  When the
+device is inactive for a certain time, the device is automatically
+turned off to save the power.  The time to go down is specified via
+`power_save` module option, and this option can be changed dynamically
+via sysfs.
+
+The power-saving won't work when the analog loopback is enabled on
+some codecs.  Make sure that you mute all unneeded signal routes when
+you want the power-saving.
+
+The power-saving feature might cause audible click noises at each
+power-down/up depending on the device.  Some of them might be
+solvable, but some are hard, I'm afraid.  Some distros such as
+openSUSE enables the power-saving feature automatically when the power
+cable is unplugged.  Thus, if you hear noises, suspect first the
+power-saving.  See /sys/module/snd_hda_intel/parameters/power_save to
+check the current value.  If it's non-zero, the feature is turned on.
+
+
+Development Tree
+~~~~~~~~~~~~~~~~
+The latest development codes for HD-audio are found on sound git tree:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
+
+The master branch or for-next branches can be used as the main
+development branches in general while the HD-audio specific patches
+are committed in topic/hda branch.
+
+If you are using the latest Linus tree, it'd be better to pull the
+above GIT tree onto it.  If you are using the older kernels, an easy
+way to try the latest ALSA code is to build from the snapshot
+tarball.  There are daily tarballs and the latest snapshot tarball.
+All can be built just like normal alsa-driver release packages, that
+is, installed via the usual spells: configure, make and make
+install(-modules).  See INSTALL in the package.  The snapshot tarballs
+are found at:
+
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/
+
+
+Sending a Bug Report
+~~~~~~~~~~~~~~~~~~~~
+If any model or module options don't work for your device, it's time
+to send a bug report to the developers.  Give the following in your
+bug report:
+
+- Hardware vendor, product and model names
+- Kernel version (and ALSA-driver version if you built externally)
+- `alsa-info.sh` output; run with `--no-upload` option.  See the
+  section below about alsa-info
+
+If it's a regression, at best, send alsa-info outputs of both working
+and non-working kernels.  This is really helpful because we can
+compare the codec registers directly.
+
+Send a bug report either the followings:
+
+kernel-bugzilla::
+  http://bugme.linux-foundation.org/
+alsa-devel ML::
+  alsa-devel@alsa-project.org
+
+
+DEBUG TOOLS
+-----------
+
+This section describes some tools available for debugging HD-audio
+problems.
+
+alsa-info
+~~~~~~~~~
+The script `alsa-info.sh` is a very useful tool to gather the audio
+device information.  You can fetch the latest version from:
+
+- http://www.alsa-project.org/alsa-info.sh
+
+Run this script as root, and it will gather the important information
+such as the module lists, module parameters, proc file contents
+including the codec proc files, mixer outputs and the control
+elements.  As default, it will store the information onto a web server
+on alsa-project.org.  But, if you send a bug report, it'd be better to
+run with `--no-upload` option, and attach the generated file.
+
+There are some other useful options.  See `--help` option output for
+details.
+
+
+hda-verb
+~~~~~~~~
+hda-verb is a tiny program that allows you to access the HD-audio
+codec directly.  You can execute a raw HD-audio codec verb with this.
+This program accesses the hwdep device, thus you need to enable the
+kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand.
+
+The hda-verb program takes four arguments: the hwdep device file, the
+widget NID, the verb and the parameter.  When you access to the codec
+on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first
+argument, typically.  (However, the real path name depends on the
+system.)
+
+The second parameter is the widget number-id to access.  The third
+parameter can be either a hex/digit number or a string corresponding
+to a verb.  Similarly, the last parameter is the value to write, or
+can be a string for the parameter type.
+
+------------------------------------------------------------------------
+  % hda-verb /dev/snd/hwC0D0 0x12 0x701 2
+  nid = 0x12, verb = 0x701, param = 0x2
+  value = 0x0
+
+  % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID
+  nid = 0x0, verb = 0xf00, param = 0x0
+  value = 0x10ec0262
+
+  % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080
+  nid = 0x2, verb = 0x300, param = 0xb080
+  value = 0x0
+------------------------------------------------------------------------
+
+Although you can issue any verbs with this program, the driver state
+won't be always updated.  For example, the volume values are usually
+cached in the driver, and thus changing the widget amp value directly
+via hda-verb won't change the mixer value.
+
+The hda-verb program is found in the ftp directory:
+
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
+
+Also a git repository is available:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git
+
+See README file in the tarball for more details about hda-verb
+program.
+
+
+hda-analyzer
+~~~~~~~~~~~~
+hda-analyzer provides a graphical interface to access the raw HD-audio
+control, based on pyGTK2 binding.  It's a more powerful version of
+hda-verb.  The program gives you an easy-to-use GUI stuff for showing
+the widget information and adjusting the amp values, as well as the
+proc-compatible output.
+
+The hda-analyzer is a part of alsa.git repository in
+alsa-project.org:
+
+- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
+
+
+Codecgraph
+~~~~~~~~~~
+Codecgraph is a utility program to generate a graph and visualizes the
+codec-node connection of a codec chip.  It's especially useful when
+you analyze or debug a codec without a proper datasheet.  The program
+parses the given codec proc file and converts to SVG via graphiz
+program.
+
+The tarball and GIT trees are found in the web page at:
+
+- http://helllabs.org/codecgraph/
+
+
+hda-emu
+~~~~~~~
+hda-emu is an HD-audio emulator.  The main purpose of this program is
+to debug an HD-audio codec without the real hardware.  Thus, it
+doesn't emulate the behavior with the real audio I/O, but it just
+dumps the codec register changes and the ALSA-driver internal changes
+at probing and operating the HD-audio driver.
+
+The program requires a codec proc-file to simulate.  Get a proc file
+for the target codec beforehand, or pick up an example codec from the
+codec proc collections in the tarball.  Then, run the program with the
+proc file, and the hda-emu program will start parsing the codec file
+and simulates the HD-audio driver:
+
+------------------------------------------------------------------------
+  % hda-emu codecs/stac9200-dell-d820-laptop
+  # Parsing..
+  hda_codec: Unknown model for STAC9200, using BIOS defaults
+  hda_codec: pin nid 08 bios pin config 40c003fa
+  ....
+------------------------------------------------------------------------
+
+The program gives you only a very dumb command-line interface.  You
+can get a proc-file dump at the current state, get a list of control
+(mixer) elements, set/get the control element value, simulate the PCM
+operation, the jack plugging simulation, etc.
+
+The package is found in:
+
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
+
+A git repository is available:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
+
+See README file in the tarball for more details about hda-emu
+program.
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index f738b296440a..bba2dbb79d81 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -153,6 +153,16 @@ card*/codec#*
 	Shows the general codec information and the attribute of each
 	widget node.
 
+card*/eld#*
+	Available for HDMI or DisplayPort interfaces.
+	Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink,
+	and describes its audio capabilities and configurations.
+
+	Some ELD fields may be modified by doing `echo name hex_value > eld#*`.
+	Only do this if you are sure the HDMI sink provided value is wrong.
+	And if that makes your HDMI audio work, please report to us so that we
+	can fix it in future kernel releases.
+
 
 Sequencer Information
 ---------------------
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index f370e7db86af..bab7711ce963 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by
 the following struct:-
 
 /* SoC machine */
-struct snd_soc_machine {
+struct snd_soc_card {
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -67,10 +67,10 @@ static struct snd_soc_dai_link corgi_dai = {
 	.ops = &corgi_ops,
 };
 
-struct snd_soc_machine then sets up the machine with it's DAIs. e.g.
+struct snd_soc_card then sets up the machine with it's DAIs. e.g.
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
 	.dai_link = &corgi_dai,
 	.num_links = 1,
@@ -90,7 +90,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
+	.machine = &snd_soc_corgi,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
diff --git a/MAINTAINERS b/MAINTAINERS
index fbc8fa58d56d..701025701514 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3977,7 +3977,7 @@ M:	tiwai@suse.de
 L:	alsa-devel@alsa-project.org (subscribers-only)
 S:	Maintained
 
-SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
+SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
 P:	Liam Girdwood
 M:	lrg@slimlogic.co.uk
 P:	Mark Brown
diff --git a/arch/arm/mach-pxa/include/mach/palmasoc.h b/arch/arm/mach-pxa/include/mach/palmasoc.h
new file mode 100644
index 000000000000..6c4b1f7de20a
--- /dev/null
+++ b/arch/arm/mach-pxa/include/mach/palmasoc.h
@@ -0,0 +1,13 @@
+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+	int	jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
+#else
+static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {}
+#endif
+
+#endif
diff --git a/include/linux/input.h b/include/linux/input.h
index 5341e8251f8c..9a6355f74db2 100644
--- a/include/linux/input.h
+++ b/include/linux/input.h
@@ -659,6 +659,8 @@ struct input_absinfo {
 #define SW_RADIO		SW_RFKILL_ALL	/* deprecated */
 #define SW_MICROPHONE_INSERT	0x04  /* set = inserted */
 #define SW_DOCK			0x05  /* set = plugged into dock */
+#define SW_LINEOUT_INSERT	0x06  /* set = inserted */
+#define SW_JACK_PHYSICAL_INSERT 0x07  /* set = mechanical switch set */
 #define SW_MAX			0x0f
 #define SW_CNT			(SW_MAX+1)
 
diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index 217bb22ebb8e..af95a1d2f3a1 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -1,7 +1,7 @@
 /*
  * audio.h  --  Audio Driver for Wolfson WM8350 PMIC
  *
- * Copyright 2007 Wolfson Microelectronics PLC
+ * Copyright 2007, 2008 Wolfson Microelectronics PLC
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -70,9 +70,9 @@
 #define WM8350_CODEC_ISEL_0_5                   3	/* x0.5 */
 
 #define WM8350_VMID_OFF                         0
-#define WM8350_VMID_500K                        1
-#define WM8350_VMID_100K                        2
-#define WM8350_VMID_10K                         3
+#define WM8350_VMID_300K                        1
+#define WM8350_VMID_50K                         2
+#define WM8350_VMID_5K                          3
 
 /*
  * R40 (0x28) - Clock Control 1
@@ -591,8 +591,38 @@
 #define WM8350_IRQ_CODEC_MICSCD			41
 #define WM8350_IRQ_CODEC_MICD			42
 
+/*
+ * WM8350 Platform data.
+ *
+ * This must be initialised per platform for best audio performance.
+ * Please see WM8350 datasheet for information.
+ */
+struct wm8350_audio_platform_data {
+	int vmid_discharge_msecs;	/* VMID --> OFF discharge time */
+	int drain_msecs;	/* OFF drain time */
+	int cap_discharge_msecs;	/* Cap ON (from OFF) discharge time */
+	int vmid_charge_msecs;	/* vmid power up time */
+	u32 vmid_s_curve:2;	/* vmid enable s curve speed */
+	u32 dis_out4:2;		/* out4 discharge speed */
+	u32 dis_out3:2;		/* out3 discharge speed */
+	u32 dis_out2:2;		/* out2 discharge speed */
+	u32 dis_out1:2;		/* out1 discharge speed */
+	u32 vroi_out4:1;	/* out4 tie off */
+	u32 vroi_out3:1;	/* out3 tie off */
+	u32 vroi_out2:1;	/* out2 tie off */
+	u32 vroi_out1:1;	/* out1 tie off */
+	u32 vroi_enable:1;	/* enable tie off */
+	u32 codec_current_on:2;	/* current level ON */
+	u32 codec_current_standby:2;	/* current level STANDBY */
+	u32 codec_current_charge:2;	/* codec current @ vmid charge */
+};
+
+struct snd_soc_codec;
+
 struct wm8350_codec {
 	struct platform_device *pdev;
+	struct snd_soc_codec *codec;
+	struct wm8350_audio_platform_data *platform_data;
 };
 
 #endif
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 9c309daf492b..251fc1cd5002 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -281,10 +281,12 @@
 /* specific - Analog Devices */
 #define AC97_AD_TEST		0x5a	/* test register */
 #define AC97_AD_TEST2		0x5c	/* undocumented test register 2 */
+#define AC97_AD_HPFD_SHIFT	12	/* High Pass Filter Disable */
 #define AC97_AD_CODEC_CFG	0x70	/* codec configuration */
 #define AC97_AD_JACK_SPDIF	0x72	/* Jack Sense & S/PDIF */
 #define AC97_AD_SERIAL_CFG	0x74	/* Serial Configuration */
 #define AC97_AD_MISC		0x76	/* Misc Control Bits */
+#define AC97_AD_VREFD_SHIFT	2	/* V_REFOUT Disable (AD1888) */
 
 /* specific - Cirrus Logic */
 #define AC97_CSR_ACMODE		0x5e	/* AC Mode Register */
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 2c4dc908a54a..1c02ed1d7c4a 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -575,6 +575,7 @@ enum {
 #define SNDRV_TIMER_GLOBAL_SYSTEM	0
 #define SNDRV_TIMER_GLOBAL_RTC		1
 #define SNDRV_TIMER_GLOBAL_HPET		2
+#define SNDRV_TIMER_GLOBAL_HRTIMER	3
 
 /* info flags */
 #define SNDRV_TIMER_FLG_SLAVE		(1<<0)	/* cannot be controlled */
diff --git a/include/sound/core.h b/include/sound/core.h
index 1508c4ec1ba9..f632484bc743 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -353,7 +353,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
  * snd_printk - printk wrapper
  * @fmt: format string
  *
- * Works like print() but prints the file and the line of the caller
+ * Works like printk() but prints the file and the line of the caller
  * when configured with CONFIG_SND_VERBOSE_PRINTK.
  */
 #define snd_printk(fmt, args...) \
@@ -380,18 +380,40 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 	printk(fmt ,##args)
 #endif
 
+/**
+ * snd_BUG - give a BUG warning message and stack trace
+ *
+ * Calls WARN() if CONFIG_SND_DEBUG is set.
+ * Ignored when CONFIG_SND_DEBUG is not set.
+ */
 #define snd_BUG()		WARN(1, "BUG?\n")
+
+/**
+ * snd_BUG_ON - debugging check macro
+ * @cond: condition to evaluate
+ *
+ * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition,
+ * and call WARN() and returns the value if it's non-zero.
+ * 
+ * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given
+ * condition is ignored.
+ *
+ * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n.
+ * Thus, don't put any statement that influences on the code behavior,
+ * such as pre/post increment, to the argument of this macro.
+ * If you want to evaluate and give a warning, use standard WARN_ON().
+ */
 #define snd_BUG_ON(cond)	WARN((cond), "BUG? (%s)\n", __stringify(cond))
 
 #else /* !CONFIG_SND_DEBUG */
 
 #define snd_printd(fmt, args...)	do { } while (0)
 #define snd_BUG()			do { } while (0)
-static inline int __snd_bug_on(void)
+static inline int __snd_bug_on(int cond)
 {
 	return 0;
 }
-#define snd_BUG_ON(cond)		__snd_bug_on()  /* always false */
+#define snd_BUG_ON(cond)	__snd_bug_on(0 && (cond))  /* always false */
 
 #endif /* CONFIG_SND_DEBUG */
 
diff --git a/include/sound/info.h b/include/sound/info.h
index 8ae72e74f898..7c2ee1a21b00 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -40,30 +40,34 @@ struct snd_info_buffer {
 struct snd_info_entry;
 
 struct snd_info_entry_text {
-	void (*read) (struct snd_info_entry *entry, struct snd_info_buffer *buffer);
-	void (*write) (struct snd_info_entry *entry, struct snd_info_buffer *buffer);
+	void (*read)(struct snd_info_entry *entry,
+		     struct snd_info_buffer *buffer);
+	void (*write)(struct snd_info_entry *entry,
+		      struct snd_info_buffer *buffer);
 };
 
 struct snd_info_entry_ops {
-	int (*open) (struct snd_info_entry *entry,
-		     unsigned short mode, void **file_private_data);
-	int (*release) (struct snd_info_entry * entry,
-			unsigned short mode, void *file_private_data);
-	long (*read) (struct snd_info_entry *entry, void *file_private_data,
-		      struct file * file, char __user *buf,
+	int (*open)(struct snd_info_entry *entry,
+		    unsigned short mode, void **file_private_data);
+	int (*release)(struct snd_info_entry *entry,
+		       unsigned short mode, void *file_private_data);
+	long (*read)(struct snd_info_entry *entry, void *file_private_data,
+		     struct file *file, char __user *buf,
+		     unsigned long count, unsigned long pos);
+	long (*write)(struct snd_info_entry *entry, void *file_private_data,
+		      struct file *file, const char __user *buf,
 		      unsigned long count, unsigned long pos);
-	long (*write) (struct snd_info_entry *entry, void *file_private_data,
-		       struct file * file, const char __user *buf,
-		       unsigned long count, unsigned long pos);
-	long long (*llseek) (struct snd_info_entry *entry, void *file_private_data,
-			    struct file * file, long long offset, int orig);
-	unsigned int (*poll) (struct snd_info_entry *entry, void *file_private_data,
-			      struct file * file, poll_table * wait);
-	int (*ioctl) (struct snd_info_entry *entry, void *file_private_data,
-		      struct file * file, unsigned int cmd, unsigned long arg);
-	int (*mmap) (struct snd_info_entry *entry, void *file_private_data,
-		     struct inode * inode, struct file * file,
-		     struct vm_area_struct * vma);
+	long long (*llseek)(struct snd_info_entry *entry,
+			    void *file_private_data, struct file *file,
+			    long long offset, int orig);
+	unsigned int(*poll)(struct snd_info_entry *entry,
+			    void *file_private_data, struct file *file,
+			    poll_table *wait);
+	int (*ioctl)(struct snd_info_entry *entry, void *file_private_data,
+		     struct file *file, unsigned int cmd, unsigned long arg);
+	int (*mmap)(struct snd_info_entry *entry, void *file_private_data,
+		    struct inode *inode, struct file *file,
+		    struct vm_area_struct *vma);
 };
 
 struct snd_info_entry {
@@ -106,34 +110,37 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer);
 static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
 #endif
 
-int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3)));
+int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \
+				__attribute__ ((format (printf, 2, 3)));
 int snd_info_init(void);
 int snd_info_done(void);
 
-int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len);
+int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
 char *snd_info_get_str(char *dest, char *src, int len);
-struct snd_info_entry *snd_info_create_module_entry(struct module * module,
+struct snd_info_entry *snd_info_create_module_entry(struct module *module,
 					       const char *name,
-					       struct snd_info_entry * parent);
-struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card,
+					       struct snd_info_entry *parent);
+struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
 					     const char *name,
-					     struct snd_info_entry * parent);
-void snd_info_free_entry(struct snd_info_entry * entry);
-int snd_info_store_text(struct snd_info_entry * entry);
-int snd_info_restore_text(struct snd_info_entry * entry);
+					     struct snd_info_entry *parent);
+void snd_info_free_entry(struct snd_info_entry *entry);
+int snd_info_store_text(struct snd_info_entry *entry);
+int snd_info_restore_text(struct snd_info_entry *entry);
 
-int snd_info_card_create(struct snd_card * card);
-int snd_info_card_register(struct snd_card * card);
-int snd_info_card_free(struct snd_card * card);
-void snd_info_card_disconnect(struct snd_card * card);
-int snd_info_register(struct snd_info_entry * entry);
+int snd_info_card_create(struct snd_card *card);
+int snd_info_card_register(struct snd_card *card);
+int snd_info_card_free(struct snd_card *card);
+void snd_info_card_disconnect(struct snd_card *card);
+void snd_info_card_id_change(struct snd_card *card);
+int snd_info_register(struct snd_info_entry *entry);
 
 /* for card drivers */
-int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp);
+int snd_card_proc_new(struct snd_card *card, const char *name,
+		      struct snd_info_entry **entryp);
 
 static inline void snd_info_set_text_ops(struct snd_info_entry *entry, 
-					 void *private_data,
-					 void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
+	void *private_data,
+	void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
 {
 	entry->private_data = private_data;
 	entry->c.text.read = read;
@@ -146,21 +153,22 @@ int snd_info_check_reserved_words(const char *str);
 #define snd_seq_root NULL
 #define snd_oss_root NULL
 
-static inline int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) { return 0; }
+static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; }
 static inline int snd_info_init(void) { return 0; }
 static inline int snd_info_done(void) { return 0; }
 
-static inline int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len) { return 0; }
+static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; }
 static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; }
-static inline struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry * parent) { return NULL; }
-static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, const char *name, struct snd_info_entry * parent) { return NULL; }
-static inline void snd_info_free_entry(struct snd_info_entry * entry) { ; }
+static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; }
+static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; }
+static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; }
 
-static inline int snd_info_card_create(struct snd_card * card) { return 0; }
-static inline int snd_info_card_register(struct snd_card * card) { return 0; }
-static inline int snd_info_card_free(struct snd_card * card) { return 0; }
-static inline void snd_info_card_disconnect(struct snd_card * card) { }
-static inline int snd_info_register(struct snd_info_entry * entry) { return 0; }
+static inline int snd_info_card_create(struct snd_card *card) { return 0; }
+static inline int snd_info_card_register(struct snd_card *card) { return 0; }
+static inline int snd_info_card_free(struct snd_card *card) { return 0; }
+static inline void snd_info_card_disconnect(struct snd_card *card) { }
+static inline void snd_info_card_id_change(struct snd_card *card) { }
+static inline int snd_info_register(struct snd_info_entry *entry) { return 0; }
 
 static inline int snd_card_proc_new(struct snd_card *card, const char *name,
 				    struct snd_info_entry **entryp) { return -EINVAL; }
diff --git a/include/sound/jack.h b/include/sound/jack.h
index b1b2b8b59adb..2e0315cdd0d6 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -35,6 +35,8 @@ enum snd_jack_types {
 	SND_JACK_HEADPHONE	= 0x0001,
 	SND_JACK_MICROPHONE	= 0x0002,
 	SND_JACK_HEADSET	= SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
+	SND_JACK_LINEOUT	= 0x0004,
+	SND_JACK_MECHANICAL	= 0x0008, /* If detected separately */
 };
 
 struct snd_jack {
diff --git a/include/sound/l3.h b/include/sound/l3.h
new file mode 100644
index 000000000000..423a08f0f1b0
--- /dev/null
+++ b/include/sound/l3.h
@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+	void (*setdat)(int);
+	void (*setclk)(int);
+	void (*setmode)(int);
+	int data_hold;
+	int data_setup;
+	int clock_high;
+	int mode_hold;
+	int mode;
+	int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 000000000000..33df4cb909d3
--- /dev/null
+++ b/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+	int l3_clk;
+	int l3_mode;
+	int l3_data;
+	void (*power) (int);
+	int model;
+};
+
+#endif
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 000000000000..24247f763608
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,231 @@
+/*
+ * linux/sound/soc-dai.h -- ALSA SoC Layer
+ *
+ * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Digital Audio Interface (DAI) API.
+ */
+
+#ifndef __LINUX_SND_SOC_DAI_H
+#define __LINUX_SND_SOC_DAI_H
+
+
+#include <linux/list.h>
+
+struct snd_pcm_substream;
+
+/*
+ * DAI hardware audio formats.
+ *
+ * Describes the physical PCM data formating and clocking. Add new formats
+ * to the end.
+ */
+#define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
+#define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
+#define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
+#define SND_SOC_DAIFMT_DSP_A		3 /* L data msb after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B		4 /* L data msb during FRM LRC */
+#define SND_SOC_DAIFMT_AC97		5 /* AC97 */
+
+/* left and right justified also known as MSB and LSB respectively */
+#define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
+#define SND_SOC_DAIFMT_LSB		SND_SOC_DAIFMT_RIGHT_J
+
+/*
+ * DAI Clock gating.
+ *
+ * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * sending or receiving PCM data in a frame. This can be used to save power.
+ */
+#define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */
+
+/*
+ * DAI Left/Right Clocks.
+ *
+ * Specifies whether the DAI can support different samples for similtanious
+ * playback and capture. This usually requires a seperate physical frame
+ * clock for playback and capture.
+ */
+#define SND_SOC_DAIFMT_SYNC		(0 << 5) /* Tx FRM = Rx FRM */
+#define SND_SOC_DAIFMT_ASYNC		(1 << 5) /* Tx FRM ~ Rx FRM */
+
+/*
+ * TDM
+ *
+ * Time Division Multiplexing. Allows PCM data to be multplexed with other
+ * data on the DAI.
+ */
+#define SND_SOC_DAIFMT_TDM		(1 << 6)
+
+/*
+ * DAI hardware signal inversions.
+ *
+ * Specifies whether the DAI can also support inverted clocks for the specified
+ * format.
+ */
+#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal bclk + inv frm */
+#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert bclk + nor frm */
+#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert bclk + frm */
+
+/*
+ * DAI hardware clock masters.
+ *
+ * This is wrt the codec, the inverse is true for the interface
+ * i.e. if the codec is clk and frm master then the interface is
+ * clk and frame slave.
+ */
+#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & frm master */
+#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & frm master */
+#define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & frm slave */
+
+#define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
+#define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
+#define SND_SOC_DAIFMT_INV_MASK		0x0f00
+#define SND_SOC_DAIFMT_MASTER_MASK	0xf000
+
+/*
+ * Master Clock Directions
+ */
+#define SND_SOC_CLOCK_IN		0
+#define SND_SOC_CLOCK_OUT		1
+
+struct snd_soc_dai_ops;
+struct snd_soc_dai;
+struct snd_ac97_bus_ops;
+
+/* Digital Audio Interface registration */
+int snd_soc_register_dai(struct snd_soc_dai *dai);
+void snd_soc_unregister_dai(struct snd_soc_dai *dai);
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
+
+/* Digital Audio Interface clocking API.*/
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+	unsigned int freq, int dir);
+
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+	int div_id, int div);
+
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+	int pll_id, unsigned int freq_in, unsigned int freq_out);
+
+/* Digital Audio interface formatting */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
+
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+	unsigned int mask, int slots);
+
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
+
+/* Digital Audio Interface mute */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+
+/*
+ * Digital Audio Interface.
+ *
+ * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
+ * operations an capabilities. Codec and platfom drivers will register a this
+ * structure for every DAI they have.
+ *
+ * This structure covers the clocking, formating and ALSA operations for each
+ * interface a
+ */
+struct snd_soc_dai_ops {
+	/*
+	 * DAI clocking configuration, all optional.
+	 * Called by soc_card drivers, normally in their hw_params.
+	 */
+	int (*set_sysclk)(struct snd_soc_dai *dai,
+		int clk_id, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_dai *dai,
+		int pll_id, unsigned int freq_in, unsigned int freq_out);
+	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
+
+	/*
+	 * DAI format configuration
+	 * Called by soc_card drivers, normally in their hw_params.
+	 */
+	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+	int (*set_tdm_slot)(struct snd_soc_dai *dai,
+		unsigned int mask, int slots);
+	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
+
+	/*
+	 * DAI digital mute - optional.
+	 * Called by soc-core to minimise any pops.
+	 */
+	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+
+	/*
+	 * ALSA PCM audio operations - all optional.
+	 * Called by soc-core during audio PCM operations.
+	 */
+	int (*startup)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	void (*shutdown)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*hw_params)(struct snd_pcm_substream *,
+		struct snd_pcm_hw_params *, struct snd_soc_dai *);
+	int (*hw_free)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*prepare)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+	/* DAI description */
+	char *name;
+	unsigned int id;
+	int ac97_control;
+
+	struct device *dev;
+
+	/* DAI callbacks */
+	int (*probe)(struct platform_device *pdev,
+		     struct snd_soc_dai *dai);
+	void (*remove)(struct platform_device *pdev,
+		       struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
+
+	/* ops */
+	struct snd_soc_dai_ops ops;
+
+	/* DAI capabilities */
+	struct snd_soc_pcm_stream capture;
+	struct snd_soc_pcm_stream playback;
+
+	/* DAI runtime info */
+	struct snd_pcm_runtime *runtime;
+	struct snd_soc_codec *codec;
+	unsigned int active;
+	unsigned char pop_wait:1;
+	void *dma_data;
+
+	/* DAI private data */
+	void *private_data;
+
+	/* parent codec/platform */
+	union {
+		struct snd_soc_codec *codec;
+		struct snd_soc_platform *platform;
+	};
+
+	struct list_head list;
+};
+
+#endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index ca699a3017f3..7ee2f70ca42e 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
 	int num);
 
 /* dapm path setup */
-int  __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
-	const char *sink_name, const char *control_name, const char *src_name);
 int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
 void snd_soc_dapm_free(struct snd_soc_device *socdev);
 int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5e0189876afd..f86e455d3828 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -21,8 +21,6 @@
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
 
-#define SND_SOC_VERSION "0.13.2"
-
 /*
  * Convenience kcontrol builders
  */
@@ -145,105 +143,31 @@ enum snd_soc_bias_level {
 	SND_SOC_BIAS_OFF,
 };
 
-/*
- * Digital Audio Interface (DAI) types
- */
-#define SND_SOC_DAI_AC97	0x1
-#define SND_SOC_DAI_I2S		0x2
-#define SND_SOC_DAI_PCM		0x4
-#define SND_SOC_DAI_AC97_BUS	0x8	/* for custom i.e. non ac97_codec.c */
-
-/*
- * DAI hardware audio formats
- */
-#define SND_SOC_DAIFMT_I2S		0	/* I2S mode */
-#define SND_SOC_DAIFMT_RIGHT_J	1	/* Right justified mode */
-#define SND_SOC_DAIFMT_LEFT_J	2	/* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A	3	/* L data msb after FRM or LRC */
-#define SND_SOC_DAIFMT_DSP_B	4	/* L data msb during FRM or LRC */
-#define SND_SOC_DAIFMT_AC97		5	/* AC97 */
-
-#define SND_SOC_DAIFMT_MSB 	SND_SOC_DAIFMT_LEFT_J
-#define SND_SOC_DAIFMT_LSB	SND_SOC_DAIFMT_RIGHT_J
-
-/*
- * DAI Gating
- */
-#define SND_SOC_DAIFMT_CONT			(0 << 4)	/* continuous clock */
-#define SND_SOC_DAIFMT_GATED		(1 << 4)	/* clock is gated when not Tx/Rx */
-
-/*
- * DAI Sync
- * Synchronous LR (Left Right) clocks and Frame signals.
- */
-#define SND_SOC_DAIFMT_SYNC		(0 << 5)	/* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC		(1 << 5)	/* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- */
-#define SND_SOC_DAIFMT_TDM		(1 << 6)
-
-/*
- * DAI hardware signal inversions
- */
-#define SND_SOC_DAIFMT_NB_NF		(0 << 8)	/* normal bclk + frm */
-#define SND_SOC_DAIFMT_NB_IF		(1 << 8)	/* normal bclk + inv frm */
-#define SND_SOC_DAIFMT_IB_NF		(2 << 8)	/* invert bclk + nor frm */
-#define SND_SOC_DAIFMT_IB_IF		(3 << 8)	/* invert bclk + frm */
-
-/*
- * DAI hardware clock masters
- * This is wrt the codec, the inverse is true for the interface
- * i.e. if the codec is clk and frm master then the interface is
- * clk and frame slave.
- */
-#define SND_SOC_DAIFMT_CBM_CFM	(0 << 12) /* codec clk & frm master */
-#define SND_SOC_DAIFMT_CBS_CFM	(1 << 12) /* codec clk slave & frm master */
-#define SND_SOC_DAIFMT_CBM_CFS	(2 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS	(3 << 12) /* codec clk & frm slave */
-
-#define SND_SOC_DAIFMT_FORMAT_MASK		0x000f
-#define SND_SOC_DAIFMT_CLOCK_MASK		0x00f0
-#define SND_SOC_DAIFMT_INV_MASK			0x0f00
-#define SND_SOC_DAIFMT_MASTER_MASK		0xf000
-
-
-/*
- * Master Clock Directions
- */
-#define SND_SOC_CLOCK_IN	0
-#define SND_SOC_CLOCK_OUT	1
-
-/*
- * AC97 codec ID's bitmask
- */
-#define SND_SOC_DAI_AC97_ID0	(1 << 0)
-#define SND_SOC_DAI_AC97_ID1	(1 << 1)
-#define SND_SOC_DAI_AC97_ID2	(1 << 2)
-#define SND_SOC_DAI_AC97_ID3	(1 << 3)
-
 struct snd_soc_device;
 struct snd_soc_pcm_stream;
 struct snd_soc_ops;
 struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
 struct snd_soc_dai;
+struct snd_soc_platform;
 struct snd_soc_codec;
-struct snd_soc_machine_config;
 struct soc_enum;
 struct snd_soc_ac97_ops;
-struct snd_soc_clock_info;
 
 typedef int (*hw_write_t)(void *,const char* ,int);
 typedef int (*hw_read_t)(void *,char* ,int);
 
 extern struct snd_ac97_bus_ops soc_ac97_ops;
 
+int snd_soc_register_platform(struct snd_soc_platform *platform);
+void snd_soc_unregister_platform(struct snd_soc_platform *platform);
+int snd_soc_register_codec(struct snd_soc_codec *codec);
+void snd_soc_unregister_codec(struct snd_soc_codec *codec);
+
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_register_card(struct snd_soc_device *socdev);
+int snd_soc_init_card(struct snd_soc_device *socdev);
 
 /* set runtime hw params */
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@@ -263,27 +187,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	struct snd_ac97_bus_ops *ops, int num);
 void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
 
-/* Digital Audio Interface clocking API.*/
-int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
-	unsigned int freq, int dir);
-
-int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
-	int div_id, int div);
-
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
-	int pll_id, unsigned int freq_in, unsigned int freq_out);
-
-/* Digital Audio interface formatting */
-int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
-
-int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
-	unsigned int mask, int slots);
-
-int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
-
-/* Digital Audio Interface mute */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
-
 /*
  *Controls
  */
@@ -341,66 +244,14 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-/* ASoC DAI ops */
-struct snd_soc_dai_ops {
-	/* DAI clocking configuration */
-	int (*set_sysclk)(struct snd_soc_dai *dai,
-		int clk_id, unsigned int freq, int dir);
-	int (*set_pll)(struct snd_soc_dai *dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out);
-	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
-
-	/* DAI format configuration */
-	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
-	int (*set_tdm_slot)(struct snd_soc_dai *dai,
-		unsigned int mask, int slots);
-	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
-
-	/* digital mute */
-	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
-};
-
-/* SoC  DAI (Digital Audio Interface) */
-struct snd_soc_dai {
-	/* DAI description */
-	char *name;
-	unsigned int id;
-	unsigned char type;
-
-	/* DAI callbacks */
-	int (*probe)(struct platform_device *pdev,
-		     struct snd_soc_dai *dai);
-	void (*remove)(struct platform_device *pdev,
-		       struct snd_soc_dai *dai);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-
-	/* ops */
-	struct snd_soc_ops ops;
-	struct snd_soc_dai_ops dai_ops;
-
-	/* DAI capabilities */
-	struct snd_soc_pcm_stream capture;
-	struct snd_soc_pcm_stream playback;
-
-	/* DAI runtime info */
-	struct snd_pcm_runtime *runtime;
-	struct snd_soc_codec *codec;
-	unsigned int active;
-	unsigned char pop_wait:1;
-	void *dma_data;
-
-	/* DAI private data */
-	void *private_data;
-};
-
 /* SoC Audio Codec */
 struct snd_soc_codec {
 	char *name;
 	struct module *owner;
 	struct mutex mutex;
+	struct device *dev;
+
+	struct list_head list;
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_codec *,
@@ -426,6 +277,7 @@ struct snd_soc_codec {
 	short reg_cache_step;
 
 	/* dapm */
+	u32 pop_time;
 	struct list_head dapm_widgets;
 	struct list_head dapm_paths;
 	enum snd_soc_bias_level bias_level;
@@ -435,6 +287,11 @@ struct snd_soc_codec {
 	/* codec DAI's */
 	struct snd_soc_dai *dai;
 	unsigned int num_dai;
+
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_reg;
+	struct dentry *debugfs_pop_time;
+#endif
 };
 
 /* codec device */
@@ -448,13 +305,12 @@ struct snd_soc_codec_device {
 /* SoC platform interface */
 struct snd_soc_platform {
 	char *name;
+	struct list_head list;
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
 	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
@@ -484,9 +340,14 @@ struct snd_soc_dai_link  {
 	struct snd_pcm *pcm;
 };
 
-/* SoC machine */
-struct snd_soc_machine {
+/* SoC card */
+struct snd_soc_card {
 	char *name;
+	struct device *dev;
+
+	struct list_head list;
+
+	int instantiated;
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
@@ -499,23 +360,26 @@ struct snd_soc_machine {
 	int (*resume_post)(struct platform_device *pdev);
 
 	/* callbacks */
-	int (*set_bias_level)(struct snd_soc_machine *,
+	int (*set_bias_level)(struct snd_soc_card *,
 			      enum snd_soc_bias_level level);
 
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;
 	int num_links;
+
+	struct snd_soc_device *socdev;
+
+	struct snd_soc_platform *platform;
+	struct delayed_work delayed_work;
+	struct work_struct deferred_resume_work;
 };
 
 /* SoC Device - the audio subsystem */
 struct snd_soc_device {
 	struct device *dev;
-	struct snd_soc_machine *machine;
-	struct snd_soc_platform *platform;
+	struct snd_soc_card *card;
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
-	struct delayed_work delayed_work;
-	struct work_struct deferred_resume_work;
 	void *codec_data;
 };
 
@@ -542,4 +406,6 @@ struct soc_enum {
 	void *dapm;
 };
 
+#include <sound/soc-dai.h>
+
 #endif
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
new file mode 100644
index 000000000000..475ef8bb7dcd
--- /dev/null
+++ b/include/sound/uda134x.h
@@ -0,0 +1,26 @@
+/*
+ * uda134x.h  --  UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+	struct l3_pins l3;
+	void (*power) (int);
+	int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+};
+
+#endif /* _UDA134X_H */
diff --git a/include/sound/version.h b/include/sound/version.h
index 4aafeda88634..2b48237e23bf 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
 /* include/version.h */
-#define CONFIG_SND_VERSION "1.0.18rc3"
+#define CONFIG_SND_VERSION "1.0.18a"
 #define CONFIG_SND_DATE ""
diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c
index 7fa37e15f196..a351dd0a09c7 100644
--- a/sound/ac97_bus.c
+++ b/sound/ac97_bus.c
@@ -15,6 +15,7 @@
 #include <linux/init.h>
 #include <linux/device.h>
 #include <linux/string.h>
+#include <sound/ac97_codec.h>
 
 /*
  * Let drivers decide whether they want to support given codec from their
diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile
index 31cbe68fd42f..c3ee77fc4b2d 100644
--- a/sound/aoa/codecs/Makefile
+++ b/sound/aoa/codecs/Makefile
@@ -1,3 +1,7 @@
+snd-aoa-codec-onyx-objs := onyx.o
+snd-aoa-codec-tas-objs := tas.o
+snd-aoa-codec-toonie-objs := toonie.o
+
 obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
 obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
 obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o
diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/onyx.c
similarity index 99%
rename from sound/aoa/codecs/snd-aoa-codec-onyx.c
rename to sound/aoa/codecs/onyx.c
index 6a3837d480e5..15500b9d2da0 100644
--- a/sound/aoa/codecs/snd-aoa-codec-onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -37,7 +37,7 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa");
 
-#include "snd-aoa-codec-onyx.h"
+#include "onyx.h"
 #include "../aoa.h"
 #include "../soundbus/soundbus.h"
 
@@ -292,7 +292,7 @@ static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
 static struct snd_kcontrol_new capture_source_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	/* If we name this 'Input Source', it properly shows up in
-	 * alsamixer as a selection, * but it's shown under the 
+	 * alsamixer as a selection, * but it's shown under the
 	 * 'Playback' category.
 	 * If I name it 'Capture Source', it shows up in strange
 	 * ways (two bools of which one can be selected at a
@@ -477,7 +477,7 @@ static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol,
 
 	ucontrol->value.iec958.status[3] = 0x3f;
 	ucontrol->value.iec958.status[4] = 0x0f;
-	
+
 	return 0;
 }
 
@@ -682,7 +682,7 @@ static int onyx_usable(struct codec_info_item *cii,
 	onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
 	spdif_enabled = !!(v & ONYX_SPDIF_ENABLE);
 	onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
-	analog_enabled = 
+	analog_enabled =
 		(v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT))
 		 != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT);
 	mutex_unlock(&onyx->mutex);
@@ -882,7 +882,7 @@ static int onyx_init_codec(struct aoa_codec *codec)
 	msleep(1);
 	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
 	msleep(1);
-	
+
 	if (onyx_register_init(onyx)) {
 		printk(KERN_ERR PFX "failed to initialise onyx registers\n");
 		return -ENODEV;
@@ -1069,7 +1069,7 @@ static int onyx_i2c_attach(struct i2c_adapter *adapter)
 
 	/* if that didn't work, try desperate mode for older
 	 * machines that have stuff missing from the device tree */
-	
+
 	if (!of_device_is_compatible(busnode, "k2-i2c"))
 		return -ENODEV;
 
diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.h b/sound/aoa/codecs/onyx.h
similarity index 100%
rename from sound/aoa/codecs/snd-aoa-codec-onyx.h
rename to sound/aoa/codecs/onyx.h
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h b/sound/aoa/codecs/tas-basstreble.h
similarity index 100%
rename from sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h
rename to sound/aoa/codecs/tas-basstreble.h
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h b/sound/aoa/codecs/tas-gain-table.h
similarity index 100%
rename from sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h
rename to sound/aoa/codecs/tas-gain-table.h
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/tas.c
similarity index 99%
rename from sound/aoa/codecs/snd-aoa-codec-tas.c
rename to sound/aoa/codecs/tas.c
index 6c515b2b8bbd..008e0f85097d 100644
--- a/sound/aoa/codecs/snd-aoa-codec-tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -71,9 +71,9 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("tas codec driver for snd-aoa");
 
-#include "snd-aoa-codec-tas.h"
-#include "snd-aoa-codec-tas-gain-table.h"
-#include "snd-aoa-codec-tas-basstreble.h"
+#include "tas.h"
+#include "tas-gain-table.h"
+#include "tas-basstreble.h"
 #include "../aoa.h"
 #include "../soundbus/soundbus.h"
 
@@ -880,7 +880,7 @@ static void tas_exit_codec(struct aoa_codec *codec)
 		return;
 	tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
 }
-	
+
 
 static struct i2c_driver tas_driver;
 
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.h b/sound/aoa/codecs/tas.h
similarity index 100%
rename from sound/aoa/codecs/snd-aoa-codec-tas.h
rename to sound/aoa/codecs/tas.h
diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/toonie.c
similarity index 98%
rename from sound/aoa/codecs/snd-aoa-codec-toonie.c
rename to sound/aoa/codecs/toonie.c
index 3c7d1d8a9a6f..f13827e17562 100644
--- a/sound/aoa/codecs/snd-aoa-codec-toonie.c
+++ b/sound/aoa/codecs/toonie.c
@@ -131,7 +131,7 @@ static int __init toonie_init(void)
 	toonie->codec.owner = THIS_MODULE;
 	toonie->codec.init = toonie_init_codec;
 	toonie->codec.exit = toonie_exit_codec;
-                                        
+
 	if (aoa_codec_register(&toonie->codec)) {
 		kfree(toonie);
 		return -EINVAL;
diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile
index 62dc7287f663..a1596e88c718 100644
--- a/sound/aoa/core/Makefile
+++ b/sound/aoa/core/Makefile
@@ -1,5 +1,5 @@
 obj-$(CONFIG_SND_AOA) += snd-aoa.o
-snd-aoa-objs := snd-aoa-core.o \
-		snd-aoa-alsa.o \
-		snd-aoa-gpio-pmf.o \
-		snd-aoa-gpio-feature.o
+snd-aoa-objs := core.o \
+		alsa.o \
+		gpio-pmf.o \
+		gpio-feature.o
diff --git a/sound/aoa/core/snd-aoa-alsa.c b/sound/aoa/core/alsa.c
similarity index 98%
rename from sound/aoa/core/snd-aoa-alsa.c
rename to sound/aoa/core/alsa.c
index 17fe689ed287..617850463582 100644
--- a/sound/aoa/core/snd-aoa-alsa.c
+++ b/sound/aoa/core/alsa.c
@@ -6,7 +6,7 @@
  * GPL v2, can be found in COPYING.
  */
 #include <linux/module.h>
-#include "snd-aoa-alsa.h"
+#include "alsa.h"
 
 static int index = -1;
 module_param(index, int, 0444);
@@ -64,7 +64,7 @@ int aoa_snd_device_new(snd_device_type_t type,
 {
 	struct snd_card *card = aoa_get_card();
 	int err;
-	
+
 	if (!card) return -ENOMEM;
 
 	err = snd_device_new(card, type, device_data, ops);
diff --git a/sound/aoa/core/snd-aoa-alsa.h b/sound/aoa/core/alsa.h
similarity index 100%
rename from sound/aoa/core/snd-aoa-alsa.h
rename to sound/aoa/core/alsa.h
diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/core.c
similarity index 99%
rename from sound/aoa/core/snd-aoa-core.c
rename to sound/aoa/core/core.c
index 19fdae400687..10bec6c61382 100644
--- a/sound/aoa/core/snd-aoa-core.c
+++ b/sound/aoa/core/core.c
@@ -10,7 +10,7 @@
 #include <linux/module.h>
 #include <linux/list.h>
 #include "../aoa.h"
-#include "snd-aoa-alsa.h"
+#include "alsa.h"
 
 MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver");
 MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/gpio-feature.c
similarity index 99%
rename from sound/aoa/core/snd-aoa-gpio-feature.c
rename to sound/aoa/core/gpio-feature.c
index 805dcbff2257..c93ad5dec66b 100644
--- a/sound/aoa/core/snd-aoa-gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -5,7 +5,7 @@
  *
  * GPL v2, can be found in COPYING.
  *
- * This file contains the GPIO control routines for 
+ * This file contains the GPIO control routines for
  * direct (through feature calls) access to the GPIO
  * registers.
  */
diff --git a/sound/aoa/core/snd-aoa-gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
similarity index 100%
rename from sound/aoa/core/snd-aoa-gpio-pmf.c
rename to sound/aoa/core/gpio-pmf.c
diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile
index 55fc5e7e52cf..da37c10eca51 100644
--- a/sound/aoa/fabrics/Makefile
+++ b/sound/aoa/fabrics/Makefile
@@ -1 +1,3 @@
+snd-aoa-fabric-layout-objs += layout.o
+
 obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/layout.c
similarity index 99%
rename from sound/aoa/fabrics/snd-aoa-fabric-layout.c
rename to sound/aoa/fabrics/layout.c
index dea7abb082cd..ad60f5d10e82 100644
--- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -66,7 +66,7 @@ struct layout {
 	unsigned int layout_id;
 	struct codec_connect_info codecs[MAX_CODECS_PER_BUS];
 	int flags;
-	
+
 	/* if busname is not assigned, we use 'Master' below,
 	 * so that our layout table doesn't need to be filled
 	 * too much.
diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile
index e57a5cf65655..1b949b2a4028 100644
--- a/sound/aoa/soundbus/i2sbus/Makefile
+++ b/sound/aoa/soundbus/i2sbus/Makefile
@@ -1,2 +1,2 @@
 obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
-snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o
+snd-aoa-i2sbus-objs := core.o pcm.o control.o
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-control.c b/sound/aoa/soundbus/i2sbus/control.c
similarity index 100%
rename from sound/aoa/soundbus/i2sbus/i2sbus-control.c
rename to sound/aoa/soundbus/i2sbus/control.c
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/core.c
similarity index 99%
rename from sound/aoa/soundbus/i2sbus/i2sbus-core.c
rename to sound/aoa/soundbus/i2sbus/core.c
index b4590df07466..be468edf3ecb 100644
--- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -64,7 +64,7 @@ static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
 				       struct dbdma_command_mem *r)
 {
 	if (!r->space) return;
-	
+
 	dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev,
 			    r->size, r->space, r->bus_addr);
 }
@@ -247,7 +247,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 		 * but request_resource doesn't know about parents and
 		 * contained resources...
 		 */
-		dev->allocated_resource[i] = 
+		dev->allocated_resource[i] =
 			request_mem_region(dev->resources[i].start,
 					   dev->resources[i].end -
 					   dev->resources[i].start + 1,
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus.h b/sound/aoa/soundbus/i2sbus/i2sbus.h
index ff29654782c9..befefd99e271 100644
--- a/sound/aoa/soundbus/i2sbus/i2sbus.h
+++ b/sound/aoa/soundbus/i2sbus/i2sbus.h
@@ -18,7 +18,7 @@
 #include <asm/pmac_feature.h>
 #include <asm/dbdma.h>
 
-#include "i2sbus-interface.h"
+#include "interface.h"
 #include "../soundbus.h"
 
 struct i2sbus_control {
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h b/sound/aoa/soundbus/i2sbus/interface.h
similarity index 100%
rename from sound/aoa/soundbus/i2sbus/i2sbus-interface.h
rename to sound/aoa/soundbus/i2sbus/interface.h
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c
similarity index 100%
rename from sound/aoa/soundbus/i2sbus/i2sbus-pcm.c
rename to sound/aoa/soundbus/i2sbus/pcm.c
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 66348c92f88d..7bbdda041a99 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -95,6 +95,26 @@ config SND_SEQUENCER_OSS
 	  this will be compiled as a module. The module will be called
 	  snd-seq-oss.
 
+config SND_HRTIMER
+	tristate "HR-timer backend support"
+	depends on HIGH_RES_TIMERS
+	select SND_TIMER
+	help
+	  Say Y here to enable HR-timer backend for ALSA timer.  ALSA uses
+	  the hrtimer as a precise timing source. The ALSA sequencer code
+	  also can use this timing source.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-hrtimer.
+
+config SND_SEQ_HRTIMER_DEFAULT
+	bool "Use HR-timer as default sequencer timer"
+	depends on SND_HRTIMER && SND_SEQUENCER
+	default y
+	help
+	  Say Y here to use the HR-timer backend as the default sequencer
+	  timer.
+
 config SND_RTCTIMER
 	tristate "RTC Timer support"
 	depends on RTC
@@ -114,6 +134,7 @@ config SND_RTCTIMER
 config SND_SEQ_RTCTIMER_DEFAULT
 	bool "Use RTC as default sequencer timer"
 	depends on SND_RTCTIMER && SND_SEQUENCER
+	depends on !SND_SEQ_HRTIMER_DEFAULT
 	default y
 	help
 	  Say Y here to use the RTC timer as the default sequencer
diff --git a/sound/core/Makefile b/sound/core/Makefile
index d57125a5687d..4229052e7b91 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -17,12 +17,14 @@ snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o
 
 snd-rawmidi-objs  := rawmidi.o
 snd-timer-objs    := timer.o
+snd-hrtimer-objs  := hrtimer.o
 snd-rtctimer-objs := rtctimer.o
 snd-hwdep-objs    := hwdep.o
 
 obj-$(CONFIG_SND) 		+= snd.o
 obj-$(CONFIG_SND_HWDEP)		+= snd-hwdep.o
 obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
+obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
 obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
 obj-$(CONFIG_SND_PCM)		+= snd-pcm.o snd-page-alloc.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o
diff --git a/sound/core/device.c b/sound/core/device.c
index c58d8227254c..a67dfac08c03 100644
--- a/sound/core/device.c
+++ b/sound/core/device.c
@@ -98,7 +98,7 @@ int snd_device_free(struct snd_card *card, void *device_data)
 		kfree(dev);
 		return 0;
 	}
-	snd_printd("device free %p (from %p), not found\n", device_data,
+	snd_printd("device free %p (from %pF), not found\n", device_data,
 		   __builtin_return_address(0));
 	return -ENXIO;
 }
@@ -135,7 +135,7 @@ int snd_device_disconnect(struct snd_card *card, void *device_data)
 		}
 		return 0;
 	}
-	snd_printd("device disconnect %p (from %p), not found\n", device_data,
+	snd_printd("device disconnect %p (from %pF), not found\n", device_data,
 		   __builtin_return_address(0));
 	return -ENXIO;
 }
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
new file mode 100644
index 000000000000..c1d285921f80
--- /dev/null
+++ b/sound/core/hrtimer.c
@@ -0,0 +1,155 @@
+/*
+ * ALSA timer back-end using hrtimer
+ * Copyright (C) 2008 Takashi Iwai
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/hrtimer.h>
+#include <sound/core.h>
+#include <sound/timer.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ALSA hrtimer backend");
+MODULE_LICENSE("GPL");
+
+MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_HRTIMER));
+
+#define NANO_SEC	1000000000UL	/* 10^9 in sec */
+static unsigned int resolution;
+
+struct snd_hrtimer {
+	struct snd_timer *timer;
+	struct hrtimer hrt;
+};
+
+static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
+{
+	struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
+	struct snd_timer *t = stime->timer;
+	hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+	snd_timer_interrupt(stime->timer, t->sticks);
+	return HRTIMER_RESTART;
+}
+
+static int snd_hrtimer_open(struct snd_timer *t)
+{
+	struct snd_hrtimer *stime;
+
+	stime = kmalloc(sizeof(*stime), GFP_KERNEL);
+	if (!stime)
+		return -ENOMEM;
+	hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+	stime->timer = t;
+	stime->hrt.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED;
+	stime->hrt.function = snd_hrtimer_callback;
+	t->private_data = stime;
+	return 0;
+}
+
+static int snd_hrtimer_close(struct snd_timer *t)
+{
+	struct snd_hrtimer *stime = t->private_data;
+
+	if (stime) {
+		hrtimer_cancel(&stime->hrt);
+		kfree(stime);
+		t->private_data = NULL;
+	}
+	return 0;
+}
+
+static int snd_hrtimer_start(struct snd_timer *t)
+{
+	struct snd_hrtimer *stime = t->private_data;
+
+	hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution),
+		      HRTIMER_MODE_REL);
+	return 0;
+}
+
+static int snd_hrtimer_stop(struct snd_timer *t)
+{
+	struct snd_hrtimer *stime = t->private_data;
+
+	hrtimer_cancel(&stime->hrt);
+	return 0;
+}
+
+static struct snd_timer_hardware hrtimer_hw = {
+	.flags =	SNDRV_TIMER_HW_AUTO,
+	.open =		snd_hrtimer_open,
+	.close =	snd_hrtimer_close,
+	.start =	snd_hrtimer_start,
+	.stop =		snd_hrtimer_stop,
+};
+
+/*
+ * entry functions
+ */
+
+static struct snd_timer *mytimer;
+
+static int __init snd_hrtimer_init(void)
+{
+	struct snd_timer *timer;
+	struct timespec tp;
+	int err;
+
+	hrtimer_get_res(CLOCK_MONOTONIC, &tp);
+	if (tp.tv_sec > 0 || !tp.tv_nsec) {
+		snd_printk(KERN_ERR
+			   "snd-hrtimer: Invalid resolution %u.%09u",
+			   (unsigned)tp.tv_sec, (unsigned)tp.tv_nsec);
+		return -EINVAL;
+	}
+	resolution = tp.tv_nsec;
+
+	/* Create a new timer and set up the fields */
+	err = snd_timer_global_new("hrtimer", SNDRV_TIMER_GLOBAL_HRTIMER,
+				   &timer);
+	if (err < 0)
+		return err;
+
+	timer->module = THIS_MODULE;
+	strcpy(timer->name, "HR timer");
+	timer->hw = hrtimer_hw;
+	timer->hw.resolution = resolution;
+	timer->hw.ticks = NANO_SEC / resolution;
+
+	err = snd_timer_global_register(timer);
+	if (err < 0) {
+		snd_timer_global_free(timer);
+		return err;
+	}
+	mytimer = timer; /* remember this */
+
+	return 0;
+}
+
+static void __exit snd_hrtimer_exit(void)
+{
+	if (mytimer) {
+		snd_timer_global_free(mytimer);
+		mytimer = NULL;
+	}
+}
+
+module_init(snd_hrtimer_init);
+module_exit(snd_hrtimer_exit);
diff --git a/sound/core/info.c b/sound/core/info.c
index 527b207462b0..70fa87189f36 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -652,6 +652,23 @@ int snd_info_card_register(struct snd_card *card)
 	return 0;
 }
 
+/*
+ * called on card->id change
+ */
+void snd_info_card_id_change(struct snd_card *card)
+{
+	mutex_lock(&info_mutex);
+	if (card->proc_root_link) {
+		snd_remove_proc_entry(snd_proc_root, card->proc_root_link);
+		card->proc_root_link = NULL;
+	}
+	if (strcmp(card->id, card->proc_root->name))
+		card->proc_root_link = proc_symlink(card->id,
+						    snd_proc_root,
+						    card->proc_root->name);
+	mutex_unlock(&info_mutex);
+}
+
 /*
  * de-register the card proc file
  * called from init.c
diff --git a/sound/core/init.c b/sound/core/init.c
index b47ff8b44be8..0d5520c415d3 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -533,6 +533,65 @@ static void choose_default_id(struct snd_card *card)
 	}
 }
 
+#ifndef CONFIG_SYSFS_DEPRECATED
+static ssize_t
+card_id_show_attr(struct device *dev,
+		  struct device_attribute *attr, char *buf)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)");
+}
+
+static ssize_t
+card_id_store_attr(struct device *dev, struct device_attribute *attr,
+		   const char *buf, size_t count)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	char buf1[sizeof(card->id)];
+	size_t copy = count > sizeof(card->id) - 1 ?
+					sizeof(card->id) - 1 : count;
+	size_t idx;
+	int c;
+
+	for (idx = 0; idx < copy; idx++) {
+		c = buf[idx];
+		if (!isalnum(c) && c != '_' && c != '-')
+			return -EINVAL;
+	}
+	memcpy(buf1, buf, copy);
+	buf1[copy] = '\0';
+	mutex_lock(&snd_card_mutex);
+	if (!snd_info_check_reserved_words(buf1)) {
+	     __exist:
+		mutex_unlock(&snd_card_mutex);
+		return -EEXIST;
+	}
+	for (idx = 0; idx < snd_ecards_limit; idx++) {
+		if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1))
+			goto __exist;
+	}
+	strcpy(card->id, buf1);
+	snd_info_card_id_change(card);
+	mutex_unlock(&snd_card_mutex);
+
+	return count;
+}
+
+static struct device_attribute card_id_attrs =
+	__ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr);
+
+static ssize_t
+card_number_show_attr(struct device *dev,
+		     struct device_attribute *attr, char *buf)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1);
+}
+
+static struct device_attribute card_number_attrs =
+	__ATTR(number, S_IRUGO, card_number_show_attr, NULL);
+#endif /* CONFIG_SYSFS_DEPRECATED */
+
 /**
  *  snd_card_register - register the soundcard
  *  @card: soundcard structure
@@ -553,7 +612,7 @@ int snd_card_register(struct snd_card *card)
 #ifndef CONFIG_SYSFS_DEPRECATED
 	if (!card->card_dev) {
 		card->card_dev = device_create(sound_class, card->dev,
-					       MKDEV(0, 0), NULL,
+					       MKDEV(0, 0), card,
 					       "card%i", card->number);
 		if (IS_ERR(card->card_dev))
 			card->card_dev = NULL;
@@ -575,6 +634,16 @@ int snd_card_register(struct snd_card *card)
 #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
 	if (snd_mixer_oss_notify_callback)
 		snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER);
+#endif
+#ifndef CONFIG_SYSFS_DEPRECATED
+	if (card->card_dev) {
+		err = device_create_file(card->card_dev, &card_id_attrs);
+		if (err < 0)
+			return err;
+		err = device_create_file(card->card_dev, &card_number_attrs);
+		if (err < 0)
+			return err;
+	}
 #endif
 	return 0;
 }
diff --git a/sound/core/jack.c b/sound/core/jack.c
index bd2d9e6b55e9..dd4a12dc09aa 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device)
 	else
 		input_free_device(jack->input_dev);
 
+	kfree(jack->id);
 	kfree(jack);
 
 	return 0;
@@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 	if (jack == NULL)
 		return -ENOMEM;
 
-	jack->id = id;
+	jack->id = kstrdup(id, GFP_KERNEL);
 
 	jack->input_dev = input_allocate_device();
 	if (jack->input_dev == NULL) {
@@ -102,9 +103,15 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 	if (type & SND_JACK_HEADPHONE)
 		input_set_capability(jack->input_dev, EV_SW,
 				     SW_HEADPHONE_INSERT);
+	if (type & SND_JACK_LINEOUT)
+		input_set_capability(jack->input_dev, EV_SW,
+				     SW_LINEOUT_INSERT);
 	if (type & SND_JACK_MICROPHONE)
 		input_set_capability(jack->input_dev, EV_SW,
 				     SW_MICROPHONE_INSERT);
+	if (type & SND_JACK_MECHANICAL)
+		input_set_capability(jack->input_dev, EV_SW,
+				     SW_JACK_PHYSICAL_INSERT);
 
 	err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops);
 	if (err < 0)
@@ -153,9 +160,15 @@ void snd_jack_report(struct snd_jack *jack, int status)
 	if (jack->type & SND_JACK_HEADPHONE)
 		input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
 				    status & SND_JACK_HEADPHONE);
+	if (jack->type & SND_JACK_LINEOUT)
+		input_report_switch(jack->input_dev, SW_LINEOUT_INSERT,
+				    status & SND_JACK_LINEOUT);
 	if (jack->type & SND_JACK_MICROPHONE)
 		input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT,
 				    status & SND_JACK_MICROPHONE);
+	if (jack->type & SND_JACK_MECHANICAL)
+		input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT,
+				    status & SND_JACK_MECHANICAL);
 
 	input_sync(jack->input_dev);
 }
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 39672f68ce5d..002777ba336a 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -151,7 +151,7 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
 	if (!substream->opened)
 		return;
 	if (up) {
-		tasklet_hi_schedule(&substream->runtime->tasklet);
+		tasklet_schedule(&substream->runtime->tasklet);
 	} else {
 		tasklet_kill(&substream->runtime->tasklet);
 		substream->ops->trigger(substream, 0);
@@ -908,7 +908,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
 	}
 	if (result > 0) {
 		if (runtime->event)
-			tasklet_hi_schedule(&runtime->tasklet);
+			tasklet_schedule(&runtime->tasklet);
 		else if (snd_rawmidi_ready(substream))
 			wake_up(&runtime->sleep);
 	}
diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
index 51e64e30dd3b..0851cd13e303 100644
--- a/sound/core/rtctimer.c
+++ b/sound/core/rtctimer.c
@@ -118,7 +118,7 @@ static void rtctimer_tasklet(unsigned long data)
  */
 static void rtctimer_interrupt(void *private_data)
 {
-	tasklet_hi_schedule(private_data);
+	tasklet_schedule(private_data);
 }
 
 
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index ee0f8405ab35..bf09a5ad1865 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -43,7 +43,9 @@ int seq_default_timer_class = SNDRV_TIMER_CLASS_GLOBAL;
 int seq_default_timer_sclass = SNDRV_TIMER_SCLASS_NONE;
 int seq_default_timer_card = -1;
 int seq_default_timer_device =
-#ifdef CONFIG_SND_SEQ_RTCTIMER_DEFAULT
+#ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT
+	SNDRV_TIMER_GLOBAL_HRTIMER
+#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT)
 	SNDRV_TIMER_GLOBAL_RTC
 #else
 	SNDRV_TIMER_GLOBAL_SYSTEM
diff --git a/sound/core/timer.c b/sound/core/timer.c
index c584408c9f17..796532081e81 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -743,7 +743,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 	spin_unlock_irqrestore(&timer->lock, flags);
 
 	if (use_tasklet)
-		tasklet_hi_schedule(&timer->task_queue);
+		tasklet_schedule(&timer->task_queue);
 }
 
 /*
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index 255fd18b9aec..0bcf14640fde 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -163,7 +163,7 @@ config SND_ML403_AC97CR
 
 config SND_AC97_POWER_SAVE
 	bool "AC97 Power-Saving Mode"
-	depends on SND_AC97_CODEC && EXPERIMENTAL
+	depends on SND_AC97_CODEC
 	default n
 	help
 	  Say Y here to enable the aggressive power-saving support of
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 1899cf0685bc..2a02f704f366 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -96,7 +96,7 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev)
 		return -EINVAL;
 
 	hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
-	pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ;
+	pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED;
 	pcsp_chip.timer.function = pcsp_do_timer;
 
 	card = snd_card_new(index, id, THIS_MODULE, 0);
@@ -188,10 +188,8 @@ static int __devexit pcsp_remove(struct platform_device *dev)
 
 static void pcsp_stop_beep(struct snd_pcsp *chip)
 {
-	spin_lock_irq(&chip->substream_lock);
-	if (!chip->playback_substream)
-		pcspkr_stop_sound();
-	spin_unlock_irq(&chip->substream_lock);
+	pcsp_sync_stop(chip);
+	pcspkr_stop_sound();
 }
 
 #ifdef CONFIG_PM
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index 1d661f795e8c..cdef2664218f 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -62,6 +62,8 @@ struct snd_pcsp {
 	unsigned short port, irq, dma;
 	spinlock_t substream_lock;
 	struct snd_pcm_substream *playback_substream;
+	unsigned int fmt_size;
+	unsigned int is_signed;
 	size_t playback_ptr;
 	size_t period_ptr;
 	atomic_t timer_active;
@@ -77,6 +79,7 @@ struct snd_pcsp {
 extern struct snd_pcsp pcsp_chip;
 
 extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle);
+extern void pcsp_sync_stop(struct snd_pcsp *chip);
 
 extern int snd_pcsp_new_pcm(struct snd_pcsp *chip);
 extern int snd_pcsp_new_mixer(struct snd_pcsp *chip);
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 1f42e4063118..84cc2658c05b 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -8,6 +8,7 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
+#include <linux/interrupt.h>
 #include <sound/pcm.h>
 #include <asm/io.h>
 #include "pcsp.h"
@@ -19,61 +20,57 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
 
 #define DMIX_WANTS_S16	1
 
-enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+/*
+ * Call snd_pcm_period_elapsed in a tasklet
+ * This avoids spinlock messes and long-running irq contexts
+ */
+static void pcsp_call_pcm_elapsed(unsigned long priv)
+{
+	if (atomic_read(&pcsp_chip.timer_active)) {
+		struct snd_pcm_substream *substream;
+		substream = pcsp_chip.playback_substream;
+		if (substream)
+			snd_pcm_period_elapsed(substream);
+	}
+}
+
+static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
+
+/* write the port and returns the next expire time in ns;
+ * called at the trigger-start and in hrtimer callback
+ */
+static unsigned long pcsp_timer_update(struct hrtimer *handle)
 {
 	unsigned char timer_cnt, val;
-	int fmt_size, periods_elapsed;
 	u64 ns;
-	size_t period_bytes, buffer_bytes;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_runtime *runtime;
 	struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
+	unsigned long flags;
 
 	if (chip->thalf) {
 		outb(chip->val61, 0x61);
 		chip->thalf = 0;
 		if (!atomic_read(&chip->timer_active))
-			return HRTIMER_NORESTART;
-		hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
-				ktime_set(0, chip->ns_rem));
-		return HRTIMER_RESTART;
+			return 0;
+		return chip->ns_rem;
 	}
 
-	spin_lock_irq(&chip->substream_lock);
-	/* Takashi Iwai says regarding this extra lock:
-
-	If the irq handler handles some data on the DMA buffer, it should
-	do snd_pcm_stream_lock().
-	That protects basically against all races among PCM callbacks, yes.
-	However, there are two remaining issues:
-	1. The substream pointer you try to lock isn't protected _before_
-	  this lock yet.
-	2. snd_pcm_period_elapsed() itself acquires the lock.
-	The requirement of another lock is because of 1.  When you get
-	chip->playback_substream, it's not protected.
-	Keeping this lock while snd_pcm_period_elapsed() assures the substream
-	is still protected (at least, not released).  And the other status is
-	handled properly inside snd_pcm_stream_lock() in
-	snd_pcm_period_elapsed().
-
-	*/
-	if (!chip->playback_substream)
-		goto exit_nr_unlock1;
-	substream = chip->playback_substream;
-	snd_pcm_stream_lock(substream);
 	if (!atomic_read(&chip->timer_active))
-		goto exit_nr_unlock2;
+		return 0;
+	substream = chip->playback_substream;
+	if (!substream)
+		return 0;
 
 	runtime = substream->runtime;
-	fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3;
 	/* assume it is mono! */
-	val = runtime->dma_area[chip->playback_ptr + fmt_size - 1];
-	if (snd_pcm_format_signed(runtime->format))
+	val = runtime->dma_area[chip->playback_ptr + chip->fmt_size - 1];
+	if (chip->is_signed)
 		val ^= 0x80;
 	timer_cnt = val * CUR_DIV() / 256;
 
 	if (timer_cnt && chip->enable) {
-		spin_lock(&i8253_lock);
+		spin_lock_irqsave(&i8253_lock, flags);
 		if (!nforce_wa) {
 			outb_p(chip->val61, 0x61);
 			outb_p(timer_cnt, 0x42);
@@ -82,12 +79,39 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 			outb(chip->val61 ^ 2, 0x61);
 			chip->thalf = 1;
 		}
-		spin_unlock(&i8253_lock);
+		spin_unlock_irqrestore(&i8253_lock, flags);
 	}
 
+	chip->ns_rem = PCSP_PERIOD_NS();
+	ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem);
+	chip->ns_rem -= ns;
+	return ns;
+}
+
+enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+{
+	struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
+	struct snd_pcm_substream *substream;
+	int periods_elapsed, pointer_update;
+	size_t period_bytes, buffer_bytes;
+	unsigned long ns;
+	unsigned long flags;
+
+	pointer_update = !chip->thalf;
+	ns = pcsp_timer_update(handle);
+	if (!ns)
+		return HRTIMER_NORESTART;
+
+	/* update the playback position */
+	substream = chip->playback_substream;
+	if (!substream)
+		return HRTIMER_NORESTART;
+
 	period_bytes = snd_pcm_lib_period_bytes(substream);
 	buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
-	chip->playback_ptr += PCSP_INDEX_INC() * fmt_size;
+
+	spin_lock_irqsave(&chip->substream_lock, flags);
+	chip->playback_ptr += PCSP_INDEX_INC() * chip->fmt_size;
 	periods_elapsed = chip->playback_ptr - chip->period_ptr;
 	if (periods_elapsed < 0) {
 #if PCSP_DEBUG
@@ -102,41 +126,30 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 	 * or ALSA will BUG on us. */
 	chip->playback_ptr %= buffer_bytes;
 
-	snd_pcm_stream_unlock(substream);
-
 	if (periods_elapsed) {
-		snd_pcm_period_elapsed(substream);
 		chip->period_ptr += periods_elapsed * period_bytes;
 		chip->period_ptr %= buffer_bytes;
 	}
+	spin_unlock_irqrestore(&chip->substream_lock, flags);
 
-	spin_unlock_irq(&chip->substream_lock);
+	if (periods_elapsed)
+		tasklet_schedule(&pcsp_pcm_tasklet);
 
-	if (!atomic_read(&chip->timer_active))
-		return HRTIMER_NORESTART;
+	hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
 
-	chip->ns_rem = PCSP_PERIOD_NS();
-	ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem);
-	chip->ns_rem -= ns;
-	hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
-							ktime_set(0, ns));
 	return HRTIMER_RESTART;
-
-exit_nr_unlock2:
-	snd_pcm_stream_unlock(substream);
-exit_nr_unlock1:
-	spin_unlock_irq(&chip->substream_lock);
-	return HRTIMER_NORESTART;
 }
 
-static void pcsp_start_playing(struct snd_pcsp *chip)
+static int pcsp_start_playing(struct snd_pcsp *chip)
 {
+	unsigned long ns;
+
 #if PCSP_DEBUG
 	printk(KERN_INFO "PCSP: start_playing called\n");
 #endif
 	if (atomic_read(&chip->timer_active)) {
 		printk(KERN_ERR "PCSP: Timer already active\n");
-		return;
+		return -EIO;
 	}
 
 	spin_lock(&i8253_lock);
@@ -146,7 +159,12 @@ static void pcsp_start_playing(struct snd_pcsp *chip)
 	atomic_set(&chip->timer_active, 1);
 	chip->thalf = 0;
 
-	hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
+	ns = pcsp_timer_update(&pcsp_chip.timer);
+	if (!ns)
+		return -EIO;
+
+	hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL);
+	return 0;
 }
 
 static void pcsp_stop_playing(struct snd_pcsp *chip)
@@ -165,26 +183,35 @@ static void pcsp_stop_playing(struct snd_pcsp *chip)
 	spin_unlock(&i8253_lock);
 }
 
+/*
+ * Force to stop and sync the stream
+ */
+void pcsp_sync_stop(struct snd_pcsp *chip)
+{
+	local_irq_disable();
+	pcsp_stop_playing(chip);
+	local_irq_enable();
+	hrtimer_cancel(&chip->timer);
+	tasklet_kill(&pcsp_pcm_tasklet);
+}
+
 static int snd_pcsp_playback_close(struct snd_pcm_substream *substream)
 {
 	struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
 #if PCSP_DEBUG
 	printk(KERN_INFO "PCSP: close called\n");
 #endif
-	if (atomic_read(&chip->timer_active)) {
-		printk(KERN_ERR "PCSP: timer still active\n");
-		pcsp_stop_playing(chip);
-	}
-	spin_lock_irq(&chip->substream_lock);
+	pcsp_sync_stop(chip);
 	chip->playback_substream = NULL;
-	spin_unlock_irq(&chip->substream_lock);
 	return 0;
 }
 
 static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream,
 				       struct snd_pcm_hw_params *hw_params)
 {
+	struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
 	int err;
+	pcsp_sync_stop(chip);
 	err = snd_pcm_lib_malloc_pages(substream,
 				      params_buffer_bytes(hw_params));
 	if (err < 0)
@@ -194,9 +221,11 @@ static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream,
 
 static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream)
 {
+	struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
 #if PCSP_DEBUG
 	printk(KERN_INFO "PCSP: hw_free called\n");
 #endif
+	pcsp_sync_stop(chip);
 	return snd_pcm_lib_free_pages(substream);
 }
 
@@ -212,8 +241,12 @@ static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream)
 			snd_pcm_lib_period_bytes(substream),
 			substream->runtime->periods);
 #endif
+	pcsp_sync_stop(chip);
 	chip->playback_ptr = 0;
 	chip->period_ptr = 0;
+	chip->fmt_size =
+		snd_pcm_format_physical_width(substream->runtime->format) >> 3;
+	chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
 	return 0;
 }
 
@@ -226,8 +259,7 @@ static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
-		pcsp_start_playing(chip);
-		break;
+		return pcsp_start_playing(chip);
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		pcsp_stop_playing(chip);
@@ -242,7 +274,11 @@ static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream
 						   *substream)
 {
 	struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
-	return bytes_to_frames(substream->runtime, chip->playback_ptr);
+	unsigned int pos;
+	spin_lock(&chip->substream_lock);
+	pos = chip->playback_ptr;
+	spin_unlock(&chip->substream_lock);
+	return bytes_to_frames(substream->runtime, pos);
 }
 
 static struct snd_pcm_hardware snd_pcsp_playback = {
@@ -279,9 +315,7 @@ static int snd_pcsp_playback_open(struct snd_pcm_substream *substream)
 		return -EBUSY;
 	}
 	runtime->hw = snd_pcsp_playback;
-	spin_lock_irq(&chip->substream_lock);
 	chip->playback_substream = substream;
-	spin_unlock_irq(&chip->substream_lock);
 	return 0;
 }
 
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index 473b07f6ae85..14e3354be43a 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -548,7 +548,7 @@ irqreturn_t snd_vx_irq_handler(int irq, void *dev)
 	    (chip->chip_status & VX_STAT_IS_STALE))
 		return IRQ_NONE;
 	if (! vx_test_and_ack(chip))
-		tasklet_hi_schedule(&chip->tq);
+		tasklet_schedule(&chip->tq);
 	return IRQ_HANDLED;
 }
 
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 27de574c08f7..6644d0034fba 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -823,7 +823,7 @@ static int vx_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
 		 * we trigger the pipe using tasklet, so that the interrupts are
 		 * issued surely after the trigger is completed.
 		 */ 
-		tasklet_hi_schedule(&pipe->start_tq);
+		tasklet_schedule(&pipe->start_tq);
 		chip->pcm_running++;
 		pipe->running = 1;
 		break;
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 667eccc676a4..ea06877be4b1 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -140,8 +140,10 @@ static int __devinit snd_sb8_probe(struct device *pdev, unsigned int dev)
 				break;
 			}
 		}
-		if (i >= ARRAY_SIZE(possible_ports))
+		if (i >= ARRAY_SIZE(possible_ports)) {
+			err = -EINVAL;
 			goto _err;
+		}
 	}
 	acard->chip = chip;
 			
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 7003711f4fcc..6e3a1848447c 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -208,7 +208,8 @@ config SND_OXYGEN
 	   * AuzenTech X-Meridian
 	   * Bgears b-Enspirer
 	   * Club3D Theatron DTS
-	   * HT-Omega Claro
+	   * HT-Omega Claro (plus)
+	   * HT-Omega Claro halo (XT)
 	   * Razer Barracuda AC-1
 	   * Sondigo Inferno
 
@@ -497,129 +498,7 @@ config SND_FM801_TEA575X
 	depends on SND_FM801_TEA575X_BOOL
 	default SND_FM801
 
-config SND_HDA_INTEL
-	tristate "Intel HD Audio"
-	select SND_PCM
-	select SND_VMASTER
-	help
-	  Say Y here to include support for Intel "High Definition
-	  Audio" (Azalia) motherboard devices.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-hda-intel.
-
-config SND_HDA_HWDEP
-	bool "Build hwdep interface for HD-audio driver"
-	depends on SND_HDA_INTEL
-	select SND_HWDEP
-	help
-	  Say Y here to build a hwdep interface for HD-audio driver.
-	  This interface can be used for out-of-band communication
-	  with codecs for debugging purposes.
-
-config SND_HDA_INPUT_BEEP
-	bool "Support digital beep via input layer"
-	depends on SND_HDA_INTEL
-	depends on INPUT=y || INPUT=SND_HDA_INTEL
-	help
-	  Say Y here to build a digital beep interface for HD-audio
-	  driver. This interface is used to generate digital beeps.
-
-config SND_HDA_CODEC_REALTEK
-	bool "Build Realtek HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include Realtek HD-audio codec support in
-	  snd-hda-intel driver, such as ALC880.
-
-config SND_HDA_CODEC_ANALOG
-	bool "Build Analog Device HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include Analog Device HD-audio codec support in
-	  snd-hda-intel driver, such as AD1986A.
-
-config SND_HDA_CODEC_SIGMATEL
-	bool "Build IDT/Sigmatel HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include IDT (Sigmatel) HD-audio codec support in
-	  snd-hda-intel driver, such as STAC9200.
-
-config SND_HDA_CODEC_VIA
-	bool "Build VIA HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include VIA HD-audio codec support in
-	  snd-hda-intel driver, such as VT1708.
-
-config SND_HDA_CODEC_ATIHDMI
-	bool "Build ATI HDMI HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include ATI HDMI HD-audio codec support in
-	  snd-hda-intel driver, such as ATI RS600 HDMI.
-
-config SND_HDA_CODEC_NVHDMI
-	bool "Build NVIDIA HDMI HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include NVIDIA HDMI HD-audio codec support in
-	  snd-hda-intel driver, such as NVIDIA MCP78 HDMI.
-
-config SND_HDA_CODEC_CONEXANT
-	bool "Build Conexant HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include Conexant HD-audio codec support in
-	  snd-hda-intel driver, such as CX20549.
-
-config SND_HDA_CODEC_CMEDIA
-	bool "Build C-Media HD-audio codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include C-Media HD-audio codec support in
-	  snd-hda-intel driver, such as CMI9880.
-
-config SND_HDA_CODEC_SI3054
-	bool "Build Silicon Labs 3054 HD-modem codec support"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to include Silicon Labs 3054 HD-modem codec
-	  (and compatibles) support in snd-hda-intel driver.
-
-config SND_HDA_GENERIC
-	bool "Enable generic HD-audio codec parser"
-	depends on SND_HDA_INTEL
-	default y
-	help
-	  Say Y here to enable the generic HD-audio codec parser
-	  in snd-hda-intel driver.
-
-config SND_HDA_POWER_SAVE
-	bool "Aggressive power-saving on HD-audio"
-	depends on SND_HDA_INTEL && EXPERIMENTAL
-	help
-	  Say Y here to enable more aggressive power-saving mode on
-	  HD-audio driver.  The power-saving timeout can be configured
-	  via power_save option or over sysfs on-the-fly.
-
-config SND_HDA_POWER_SAVE_DEFAULT
-	int "Default time-out for HD-audio power-save mode"
-	depends on SND_HDA_POWER_SAVE
-	default 0
-	help
-	  The default time-out value in seconds for HD-audio automatic
-	  power-save mode.  0 means to disable the power-save mode.
+source "sound/pci/hda/Kconfig"
 
 config SND_HDSP
 	tristate "RME Hammerfall DSP Audio"
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index bd510eceff1f..e2b843b4f9d0 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -175,7 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q",	patch_wolfson04, NULL},
 { 0x574d4C05, 0xffffffff, "WM9705,WM9710",	patch_wolfson05, NULL},
 { 0x574d4C09, 0xffffffff, "WM9709",		NULL,		NULL},
-{ 0x574d4C12, 0xffffffff, "WM9711,WM9712",	patch_wolfson11, NULL},
+{ 0x574d4C12, 0xffffffff, "WM9711,WM9712,WM9715",	patch_wolfson11, NULL},
 { 0x574d4c13, 0xffffffff, "WM9713,WM9714",	patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF},
 { 0x594d4800, 0xffffffff, "YMF743",		patch_yamaha_ymf743,	NULL },
 { 0x594d4802, 0xffffffff, "YMF752",		NULL,		NULL },
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 6e831aff1bd0..81bc93e5f1e3 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -2054,8 +2054,9 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
 		.get = snd_ac97_ad1888_lohpsel_get,
 		.put = snd_ac97_ad1888_lohpsel_put
 	},
-	AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, 2, 1, 1),
-	AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1),
+	AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, AC97_AD_VREFD_SHIFT, 1, 1),
+	AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2,
+			AC97_AD_HPFD_SHIFT, 1, 1),
 	AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -2832,6 +2833,8 @@ static int patch_alc655(struct snd_ac97 * ac97)
 			val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */
 		else
 			val |= (1 << 1); /* Pin 47 is spdif input pin */
+		/* this seems missing on some hardwares */
+		ac97->ext_id |= AC97_EI_SPDIF;
 	}
 	val &= ~(1 << 12); /* vref enable */
 	snd_ac97_write_cache(ac97, 0x7a, val);
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index 74175fc80c7f..14b8d9a91aae 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -664,10 +664,14 @@ struct snd_ca0106_pcm {
 struct snd_ca0106_details {
         u32 serial;
         char * name;
-        int ac97;
-	int gpio_type;
-	int i2c_adc;
-	int spi_dac;
+	int ac97;	/* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
+			   ac97 = 1 -> Default to AC97 in. */
+	int gpio_type;	/* gpio_type = 1 -> shared mic-in/line-in
+			   gpio_type = 2 -> shared side-out/line-in. */
+	int i2c_adc;	/* with i2c_adc=1, the driver adds some capture volume
+			   controls, phone, mic, line-in and aux. */
+	int spi_dac;	/* spi_dac=1 adds the mute switch for each analog
+			   output, front, rear, etc. */
 };
 
 // definition of the chip-specific record
@@ -686,11 +690,12 @@ struct snd_ca0106 {
 	spinlock_t emu_lock;
 
 	struct snd_ac97 *ac97;
-	struct snd_pcm *pcm;
+	struct snd_pcm *pcm[4];
 
 	struct snd_ca0106_channel playback_channels[4];
 	struct snd_ca0106_channel capture_channels[4];
-	u32 spdif_bits[4];             /* s/pdif out setup */
+	u32 spdif_bits[4];             /* s/pdif out default setup */
+	u32 spdif_str_bits[4];         /* s/pdif out per-stream setup */
 	int spdif_enable;
 	int capture_source;
 	int i2c_capture_source;
@@ -703,6 +708,11 @@ struct snd_ca0106 {
 	struct snd_ca_midi midi2;
 
 	u16 spi_dac_reg[16];
+
+#ifdef CONFIG_PM
+#define NUM_SAVED_VOLUMES	9
+	unsigned int saved_vol[NUM_SAVED_VOLUMES];
+#endif
 };
 
 int snd_ca0106_mixer(struct snd_ca0106 *emu);
@@ -721,3 +731,11 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
 
 int snd_ca0106_spi_write(struct snd_ca0106 * emu,
 				   unsigned int data);
+
+#ifdef CONFIG_PM
+void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
+void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
+#else
+#define snd_ca0106_mixer_suspend(chip)	do { } while (0)
+#define snd_ca0106_mixer_resume(chip)	do { } while (0)
+#endif
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 88fbf285d2b7..0e62205d4081 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	   .name   = "MSI K8N Diamond MB",
 	   .gpio_type = 2,
 	   .i2c_adc = 1,
-	   .spi_dac = 2 } ,
+	   .spi_dac = 1 } ,
 	 /* Shuttle XPC SD31P which has an onboard Creative Labs
 	  * Sound Blaster Live! 24-bit EAX
 	  * high-definition 7.1 audio processor".
@@ -305,9 +305,15 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				 SNDRV_PCM_INFO_MMAP_VALID),
 	.formats =		SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+#if 0 /* FIXME: looks like 44.1kHz capture causes noisy output on 48kHz */
 	.rates =		(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
 				 SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
 	.rate_min =		44100,
+#else
+	.rates =		(SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
+	.rate_min =		48000,
+#endif /* FIXME */
 	.rate_max =		192000,
 	.channels_min =		2,
 	.channels_max =		2,
@@ -479,6 +485,15 @@ static const int spi_dacd_bit[] = {
 	[PCM_UNKNOWN_CHANNEL]	= SPI_DACD1_BIT,
 };
 
+static void restore_spdif_bits(struct snd_ca0106 *chip, int idx)
+{
+	if (chip->spdif_str_bits[idx] != chip->spdif_bits[idx]) {
+		chip->spdif_str_bits[idx] = chip->spdif_bits[idx];
+		snd_ca0106_ptr_write(chip, SPCS0 + idx, 0,
+				     chip->spdif_str_bits[idx]);
+	}
+}
+
 /* open_playback callback */
 static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
 						int channel_id)
@@ -524,6 +539,9 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
 		if (err < 0)
 			return err;
 	}
+
+	restore_spdif_bits(chip, channel_id);
+
 	return 0;
 }
 
@@ -535,6 +553,8 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
         struct snd_ca0106_pcm *epcm = runtime->private_data;
 	chip->playback_channels[epcm->channel_id].use = 0;
 
+	restore_spdif_bits(chip, epcm->channel_id);
+
 	if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) {
 		const int reg = spi_dacd_reg[epcm->channel_id];
 
@@ -847,15 +867,18 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
         struct snd_pcm_substream *s;
 	u32 basic = 0;
 	u32 extended = 0;
-	int running=0;
+	u32 bits;
+	int running = 0;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		running=1;
+	case SNDRV_PCM_TRIGGER_RESUME:
+		running = 1;
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	default:
-		running=0;
+		running = 0;
 		break;
 	}
         snd_pcm_group_for_each_entry(s, substream) {
@@ -865,22 +888,32 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
 		runtime = s->runtime;
 		epcm = runtime->private_data;
 		channel = epcm->channel_id;
-		//snd_printk("channel=%d\n",channel);
+		/* snd_printk("channel=%d\n",channel); */
 		epcm->running = running;
-		basic |= (0x1<<channel);
-		extended |= (0x10<<channel);
+		basic |= (0x1 << channel);
+		extended |= (0x10 << channel);
                 snd_pcm_trigger_done(s, substream);
         }
-	//snd_printk("basic=0x%x, extended=0x%x\n",basic, extended);
+	/* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (extended));
-		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(basic));
+	case SNDRV_PCM_TRIGGER_RESUME:
+		bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
+		bits |= extended;
+		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
+		bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
+		bits |= basic;
+		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
-		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(basic));
-		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(extended));
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
+		bits &= ~basic;
+		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
+		bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
+		bits &= ~extended;
+		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
 		break;
 	default:
 		result = -EINVAL;
@@ -1103,21 +1136,13 @@ static int snd_ca0106_ac97(struct snd_ca0106 *chip)
 	return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
 }
 
+static void ca0106_stop_chip(struct snd_ca0106 *chip);
+
 static int snd_ca0106_free(struct snd_ca0106 *chip)
 {
-	if (chip->res_port != NULL) {    /* avoid access to already used hardware */
-		// disable interrupts
-		snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
-		outl(0, chip->port + INTE);
-		snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
-		udelay(1000);
-		// disable audio
-		//outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
-		outl(0, chip->port + HCFG);
-		/* FIXME: We need to stop and DMA transfers here.
-		 *        But as I am not sure how yet, we cannot from the dma pages.
-		 * So we can fix: snd-malloc: Memory leak?  pages not freed = 8
-		 */
+	if (chip->res_port != NULL) {
+		/* avoid access to already used hardware */
+		ca0106_stop_chip(chip);
 	}
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1203,15 +1228,14 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
 	return IRQ_HANDLED;
 }
 
-static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct snd_pcm **rpcm)
+static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device)
 {
 	struct snd_pcm *pcm;
 	struct snd_pcm_substream *substream;
 	int err;
   
-	if (rpcm)
-		*rpcm = NULL;
-	if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0)
+	err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm);
+	if (err < 0)
 		return err;
   
 	pcm->private_data = emu;
@@ -1238,7 +1262,6 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s
 	pcm->info_flags = 0;
 	pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
 	strcpy(pcm->name, "CA0106");
-	emu->pcm = pcm;
 
 	for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; 
 	    substream; 
@@ -1260,8 +1283,7 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s
 			return err;
 	}
   
-	if (rpcm)
-		*rpcm = pcm;
+	emu->pcm[device] = pcm;
   
 	return 0;
 }
@@ -1301,89 +1323,10 @@ static unsigned int i2c_adc_init[][2] = {
 	{ 0x15, ADC_MUX_LINEIN },  /* ADC Mixer control */
 };
 
-static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
-					 struct pci_dev *pci,
-					 struct snd_ca0106 **rchip)
+static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
 {
-	struct snd_ca0106 *chip;
-	struct snd_ca0106_details *c;
-	int err;
 	int ch;
-	static struct snd_device_ops ops = {
-		.dev_free = snd_ca0106_dev_free,
-	};
-  
-	*rchip = NULL;
-  
-	if ((err = pci_enable_device(pci)) < 0)
-		return err;
-	if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
-	    pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
-		printk(KERN_ERR "error to set 32bit mask DMA\n");
-		pci_disable_device(pci);
-		return -ENXIO;
-	}
-  
-	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
-	if (chip == NULL) {
-		pci_disable_device(pci);
-		return -ENOMEM;
-	}
-  
-	chip->card = card;
-	chip->pci = pci;
-	chip->irq = -1;
-
-	spin_lock_init(&chip->emu_lock);
-  
-	chip->port = pci_resource_start(pci, 0);
-	if ((chip->res_port = request_region(chip->port, 0x20,
-					     "snd_ca0106")) == NULL) { 
-		snd_ca0106_free(chip);
-		printk(KERN_ERR "cannot allocate the port\n");
-		return -EBUSY;
-	}
-
-	if (request_irq(pci->irq, snd_ca0106_interrupt,
-			IRQF_SHARED, "snd_ca0106", chip)) {
-		snd_ca0106_free(chip);
-		printk(KERN_ERR "cannot grab irq\n");
-		return -EBUSY;
-	}
-	chip->irq = pci->irq;
-  
- 	/* This stores the periods table. */ 
-	if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) {
-		snd_ca0106_free(chip);
-		return -ENOMEM;
-	}
-
-	pci_set_master(pci);
-	/* read serial */
-	pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
-	pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
-#if 1
-	printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model,
-	       pci->revision, chip->serial);
-#endif
-	strcpy(card->driver, "CA0106");
-	strcpy(card->shortname, "CA0106");
-
-	for (c = ca0106_chip_details; c->serial; c++) {
-		if (subsystem[dev]) {
-			if (c->serial == subsystem[dev])
-				break;
-		} else if (c->serial == chip->serial)
-			break;
-	}
-	chip->details = c;
-	if (subsystem[dev]) {
-		printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n",
-                        c->name, chip->serial, subsystem[dev]);
-	}
-
-	sprintf(card->longname, "%s at 0x%lx irq %i",
-		c->name, chip->port, chip->irq);
+	unsigned int def_bits;
 
 	outl(0, chip->port + INTE);
 
@@ -1401,31 +1344,22 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
 	 *  AN                = 0     (Audio data)
 	 *  P                 = 0     (Consumer)
 	 */
-	snd_ca0106_ptr_write(chip, SPCS0, 0,
-				chip->spdif_bits[0] =
-				SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
-				SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
-				SPCS_GENERATIONSTATUS | 0x00001200 |
-				0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+	def_bits =
+		SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+		SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+		SPCS_GENERATIONSTATUS | 0x00001200 |
+		0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT;
+	if (!resume) {
+		chip->spdif_str_bits[0] = chip->spdif_bits[0] = def_bits;
+		chip->spdif_str_bits[1] = chip->spdif_bits[1] = def_bits;
+		chip->spdif_str_bits[2] = chip->spdif_bits[2] = def_bits;
+		chip->spdif_str_bits[3] = chip->spdif_bits[3] = def_bits;
+	}
 	/* Only SPCS1 has been tested */
-	snd_ca0106_ptr_write(chip, SPCS1, 0,
-				chip->spdif_bits[1] =
-				SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
-				SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
-				SPCS_GENERATIONSTATUS | 0x00001200 |
-				0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
-	snd_ca0106_ptr_write(chip, SPCS2, 0,
-				chip->spdif_bits[2] =
-				SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
-				SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
-				SPCS_GENERATIONSTATUS | 0x00001200 |
-				0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
-	snd_ca0106_ptr_write(chip, SPCS3, 0,
-				chip->spdif_bits[3] =
-				SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
-				SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
-				SPCS_GENERATIONSTATUS | 0x00001200 |
-				0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+	snd_ca0106_ptr_write(chip, SPCS1, 0, chip->spdif_str_bits[1]);
+	snd_ca0106_ptr_write(chip, SPCS0, 0, chip->spdif_str_bits[0]);
+	snd_ca0106_ptr_write(chip, SPCS2, 0, chip->spdif_str_bits[2]);
+	snd_ca0106_ptr_write(chip, SPCS3, 0, chip->spdif_str_bits[3]);
 
         snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
         snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
@@ -1433,92 +1367,124 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
         /* Write 0x8000 to AC97_REC_GAIN to mute it. */
         outb(AC97_REC_GAIN, chip->port + AC97ADDRESS);
         outw(0x8000, chip->port + AC97DATA);
-#if 0
+#if 0 /* FIXME: what are these? */
 	snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006);
 	snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006);
 	snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006);
 	snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006);
 #endif
 
-	//snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
+	/* OSS drivers set this. */
+	/* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); */
+
 	/* Analog or Digital output */
 	snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
-	snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */
-	chip->spdif_enable = 0; /* Set digital SPDIF output off */
-	//snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
-	//snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */
+	/* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers.
+	 * Use 0x000f0000 for surround71
+	 */
+	snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000);
+
+	chip->spdif_enable = 0; /* Set digital SPDIF output off */
+	/*snd_ca0106_ptr_write(chip, 0x45, 0, 0);*/ /* Analogue out */
+	/*snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00);*/ /* Digital out */
+
+	/* goes to 0x40c80000 when doing SPDIF IN/OUT */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000);
+	/* (Mute) CAPTURE feedback into PLAYBACK volume.
+	 * Only lower 16 bits matter.
+	 */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff);
+	/* SPDIF IN Volume */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000);
+	/* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000);
 
-	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */
-	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */
-	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */
-	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
 	snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
 	snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
 	snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
 	snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
-	for(ch = 0; ch < 4; ch++) {
-		snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */
+
+	for (ch = 0; ch < 4; ch++) {
+		/* Only high 16 bits matter */
+		snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030);
 		snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
-		//snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */
-		//snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */
-		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */
-		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */
+#if 0 /* Mute */
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040);
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040);
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff);
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff);
+#endif
 	}
 	if (chip->details->i2c_adc == 1) {
 	        /* Select MIC, Line in, TAD in, AUX in */
 	        snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
 		/* Default to CAPTURE_SOURCE to i2s in */
-		chip->capture_source = 3;
+		if (!resume)
+			chip->capture_source = 3;
 	} else if (chip->details->ac97 == 1) {
 	        /* Default to AC97 in */
 	        snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4);
 		/* Default to CAPTURE_SOURCE to AC97 in */
-		chip->capture_source = 4;
+		if (!resume)
+			chip->capture_source = 4;
 	} else {
 	        /* Select MIC, Line in, TAD in, AUX in */
 	        snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
 		/* Default to Set CAPTURE_SOURCE to i2s in */
-		chip->capture_source = 3;
+		if (!resume)
+			chip->capture_source = 3;
 	}
 
-        if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */
-		/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
+	if (chip->details->gpio_type == 2) {
+		/* The SB0438 use GPIO differently. */
+		/* FIXME: Still need to find out what the other GPIO bits do.
+		 * E.g. For digital spdif out.
+		 */
 		outl(0x0, chip->port+GPIO);
-		//outl(0x00f0e000, chip->port+GPIO); /* Analog */
+		/* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */
 		outl(0x005f5301, chip->port+GPIO); /* Analog */
-	} else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */
-		/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
+	} else if (chip->details->gpio_type == 1) {
+		/* The SB0410 and SB0413 use GPIO differently. */
+		/* FIXME: Still need to find out what the other GPIO bits do.
+		 * E.g. For digital spdif out.
+		 */
 		outl(0x0, chip->port+GPIO);
-		//outl(0x00f0e000, chip->port+GPIO); /* Analog */
+		/* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */
 		outl(0x005f5301, chip->port+GPIO); /* Analog */
 	} else {
 		outl(0x0, chip->port+GPIO);
 		outl(0x005f03a3, chip->port+GPIO); /* Analog */
-		//outl(0x005f02a2, chip->port+GPIO);   /* SPDIF */
+		/* outl(0x005f02a2, chip->port+GPIO); */ /* SPDIF */
 	}
 	snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
 
-	//outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
-	//outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
-	//outl(0x00000009, chip->port+HCFG);
-	outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
+	/* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */
+	/* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
+	/* outl(0x00001409, chip->port+HCFG); */
+	/* outl(0x00000009, chip->port+HCFG); */
+	/* AC97 2.0, Enable outputs. */
+	outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG);
 
-        if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+	if (chip->details->i2c_adc == 1) {
+		/* The SB0410 and SB0413 use I2C to control ADC. */
 		int size, n;
 
 		size = ARRAY_SIZE(i2c_adc_init);
-                //snd_printk("I2C:array size=0x%x\n", size);
-		for (n=0; n < size; n++) {
-			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]);
+		/* snd_printk("I2C:array size=0x%x\n", size); */
+		for (n = 0; n < size; n++)
+			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
+					     i2c_adc_init[n][1]);
+		for (n = 0; n < 4; n++) {
+			chip->i2c_capture_volume[n][0] = 0xcf;
+			chip->i2c_capture_volume[n][1] = 0xcf;
 		}
-		for (n=0; n < 4; n++) {
-			chip->i2c_capture_volume[n][0]= 0xcf;
-			chip->i2c_capture_volume[n][1]= 0xcf;
-		}
-		chip->i2c_capture_source=2; /* Line in */
-	        //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */
+		chip->i2c_capture_source = 2; /* Line in */
+		/* Enable Line-in capture. MIC in currently untested. */
+		/* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */
 	}
-        if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */
+
+	if (chip->details->spi_dac == 1) {
+		/* The SB0570 use SPI to control DAC. */
 		int size, n;
 
 		size = ARRAY_SIZE(spi_dac_init);
@@ -1530,9 +1496,112 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
 				chip->spi_dac_reg[reg] = spi_dac_init[n];
 		}
 	}
+}
 
-	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
-				  chip, &ops)) < 0) {
+static void ca0106_stop_chip(struct snd_ca0106 *chip)
+{
+	/* disable interrupts */
+	snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
+	outl(0, chip->port + INTE);
+	snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
+	udelay(1000);
+	/* disable audio */
+	/* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */
+	outl(0, chip->port + HCFG);
+	/* FIXME: We need to stop and DMA transfers here.
+	 *        But as I am not sure how yet, we cannot from the dma pages.
+	 * So we can fix: snd-malloc: Memory leak?  pages not freed = 8
+	 */
+}
+
+static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
+					 struct pci_dev *pci,
+					 struct snd_ca0106 **rchip)
+{
+	struct snd_ca0106 *chip;
+	struct snd_ca0106_details *c;
+	int err;
+	static struct snd_device_ops ops = {
+		.dev_free = snd_ca0106_dev_free,
+	};
+
+	*rchip = NULL;
+
+	err = pci_enable_device(pci);
+	if (err < 0)
+		return err;
+	if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
+	    pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+		printk(KERN_ERR "error to set 32bit mask DMA\n");
+		pci_disable_device(pci);
+		return -ENXIO;
+	}
+
+	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+	if (chip == NULL) {
+		pci_disable_device(pci);
+		return -ENOMEM;
+	}
+
+	chip->card = card;
+	chip->pci = pci;
+	chip->irq = -1;
+
+	spin_lock_init(&chip->emu_lock);
+
+	chip->port = pci_resource_start(pci, 0);
+	chip->res_port = request_region(chip->port, 0x20, "snd_ca0106");
+	if (!chip->res_port) {
+		snd_ca0106_free(chip);
+		printk(KERN_ERR "cannot allocate the port\n");
+		return -EBUSY;
+	}
+
+	if (request_irq(pci->irq, snd_ca0106_interrupt,
+			IRQF_SHARED, "snd_ca0106", chip)) {
+		snd_ca0106_free(chip);
+		printk(KERN_ERR "cannot grab irq\n");
+		return -EBUSY;
+	}
+	chip->irq = pci->irq;
+
+	/* This stores the periods table. */
+	if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+				1024, &chip->buffer) < 0) {
+		snd_ca0106_free(chip);
+		return -ENOMEM;
+	}
+
+	pci_set_master(pci);
+	/* read serial */
+	pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
+	pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+	printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n",
+	       chip->model, pci->revision, chip->serial);
+	strcpy(card->driver, "CA0106");
+	strcpy(card->shortname, "CA0106");
+
+	for (c = ca0106_chip_details; c->serial; c++) {
+		if (subsystem[dev]) {
+			if (c->serial == subsystem[dev])
+				break;
+		} else if (c->serial == chip->serial)
+			break;
+	}
+	chip->details = c;
+	if (subsystem[dev]) {
+		printk(KERN_INFO "snd-ca0106: Sound card name=%s, "
+		       "subsystem=0x%x. Forced to subsystem=0x%x\n",
+		       c->name, chip->serial, subsystem[dev]);
+	}
+
+	sprintf(card->longname, "%s at 0x%lx irq %i",
+		c->name, chip->port, chip->irq);
+
+	ca0106_init_chip(chip, 0);
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0) {
 		snd_ca0106_free(chip);
 		return err;
 	}
@@ -1629,7 +1698,7 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
 	static int dev;
 	struct snd_card *card;
 	struct snd_ca0106 *chip;
-	int err;
+	int i, err;
 
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
@@ -1642,44 +1711,31 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
 	if (card == NULL)
 		return -ENOMEM;
 
-	if ((err = snd_ca0106_create(dev, card, pci, &chip)) < 0) {
-		snd_card_free(card);
-		return err;
+	err = snd_ca0106_create(dev, card, pci, &chip);
+	if (err < 0)
+		goto error;
+	card->private_data = chip;
+
+	for (i = 0; i < 4; i++) {
+		err = snd_ca0106_pcm(chip, i);
+		if (err < 0)
+			goto error;
 	}
 
-	if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) {
-		snd_card_free(card);
-		return err;
-	}
-	if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) {
-		snd_card_free(card);
-		return err;
-	}
-	if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) {
-		snd_card_free(card);
-		return err;
-	}
-	if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) {
-		snd_card_free(card);
-		return err;
-	}
-        if (chip->details->ac97 == 1) { /* The SB0410 and SB0413 do not have an AC97 chip. */
-		if ((err = snd_ca0106_ac97(chip)) < 0) {
-			snd_card_free(card);
-			return err;
-		}
-	}
-	if ((err = snd_ca0106_mixer(chip)) < 0) {
-		snd_card_free(card);
-		return err;
+	if (chip->details->ac97 == 1) {
+		/* The SB0410 and SB0413 do not have an AC97 chip. */
+		err = snd_ca0106_ac97(chip);
+		if (err < 0)
+			goto error;
 	}
+	err = snd_ca0106_mixer(chip);
+	if (err < 0)
+		goto error;
 
 	snd_printdd("ca0106: probe for MIDI channel A ...");
-	if ((err = snd_ca0106_midi(chip,CA0106_MIDI_CHAN_A)) < 0) {
-		snd_card_free(card);
-		snd_printdd(" failed, err=0x%x\n",err);
-		return err;
-	}
+	err = snd_ca0106_midi(chip, CA0106_MIDI_CHAN_A);
+	if (err < 0)
+		goto error;
 	snd_printdd(" done.\n");
 
 #ifdef CONFIG_PROC_FS
@@ -1688,14 +1744,17 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
 
 	snd_card_set_dev(card, &pci->dev);
 
-	if ((err = snd_card_register(card)) < 0) {
-		snd_card_free(card);
-		return err;
-	}
+	err = snd_card_register(card);
+	if (err < 0)
+		goto error;
 
 	pci_set_drvdata(pci, card);
 	dev++;
 	return 0;
+
+ error:
+	snd_card_free(card);
+	return err;
 }
 
 static void __devexit snd_ca0106_remove(struct pci_dev *pci)
@@ -1704,6 +1763,59 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
+#ifdef CONFIG_PM
+static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_ca0106 *chip = card->private_data;
+	int i;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	for (i = 0; i < 4; i++)
+		snd_pcm_suspend_all(chip->pcm[i]);
+	if (chip->details->ac97)
+		snd_ac97_suspend(chip->ac97);
+	snd_ca0106_mixer_suspend(chip);
+
+	ca0106_stop_chip(chip);
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+	pci_set_power_state(pci, pci_choose_state(pci, state));
+	return 0;
+}
+
+static int snd_ca0106_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_ca0106 *chip = card->private_data;
+	int i;
+
+	pci_set_power_state(pci, PCI_D0);
+	pci_restore_state(pci);
+
+	if (pci_enable_device(pci) < 0) {
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+
+	pci_set_master(pci);
+
+	ca0106_init_chip(chip, 1);
+
+	if (chip->details->ac97)
+		snd_ac97_resume(chip->ac97);
+	snd_ca0106_mixer_resume(chip);
+	if (chip->details->spi_dac) {
+		for (i = 0; i < ARRAY_SIZE(chip->spi_dac_reg); i++)
+			snd_ca0106_spi_write(chip, chip->spi_dac_reg[i]);
+	}
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+#endif
+
 // PCI IDs
 static struct pci_device_id snd_ca0106_ids[] = {
 	{ 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 },	/* Audigy LS or Live 24bit */
@@ -1717,6 +1829,10 @@ static struct pci_driver driver = {
 	.id_table = snd_ca0106_ids,
 	.probe = snd_ca0106_probe,
 	.remove = __devexit_p(snd_ca0106_remove),
+#ifdef CONFIG_PM
+	.suspend = snd_ca0106_suspend,
+	.resume = snd_ca0106_resume,
+#endif
 };
 
 // initialization of the module
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 3025ed1b6e1e..ad2888705d2a 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -75,6 +75,84 @@
 
 #include "ca0106.h"
 
+static void ca0106_spdif_enable(struct snd_ca0106 *emu)
+{
+	unsigned int val;
+
+	if (emu->spdif_enable) {
+		/* Digital */
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
+		val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000;
+		snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
+		val = inl(emu->port + GPIO) & ~0x101;
+		outl(val, emu->port + GPIO);
+
+	} else {
+		/* Analog */
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
+		val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000;
+		snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
+		val = inl(emu->port + GPIO) | 0x101;
+		outl(val, emu->port + GPIO);
+	}
+}
+
+static void ca0106_set_capture_source(struct snd_ca0106 *emu)
+{
+	unsigned int val = emu->capture_source;
+	unsigned int source, mask;
+	source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+	mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
+	snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
+}
+
+static void ca0106_set_i2c_capture_source(struct snd_ca0106 *emu,
+					  unsigned int val, int force)
+{
+	unsigned int ngain, ogain;
+	u32 source;
+
+	snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+	ngain = emu->i2c_capture_volume[val][0]; /* Left */
+	ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
+	if (force || ngain != ogain)
+		snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ngain & 0xff);
+	ngain = emu->i2c_capture_volume[val][1]; /* Right */
+	ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */
+	if (force || ngain != ogain)
+		snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ngain & 0xff);
+	source = 1 << val;
+	snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
+	emu->i2c_capture_source = val;
+}
+
+static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu)
+{
+	u32 tmp;
+
+	if (emu->capture_mic_line_in) {
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
+		tmp = inl(emu->port+GPIO) & ~0x400;
+		tmp = tmp | 0x400;
+		outl(tmp, emu->port+GPIO);
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */
+	} else {
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
+		tmp = inl(emu->port+GPIO) & ~0x400;
+		outl(tmp, emu->port+GPIO);
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */
+	}
+}
+
+static void ca0106_set_spdif_bits(struct snd_ca0106 *emu, int idx)
+{
+	snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, emu->spdif_str_bits[idx]);
+}
+
+/*
+ */
 static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
 static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
 
@@ -95,30 +173,12 @@ static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol,
 	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
 	unsigned int val;
 	int change = 0;
-	u32 mask;
 
 	val = !!ucontrol->value.integer.value[0];
 	change = (emu->spdif_enable != val);
 	if (change) {
 		emu->spdif_enable = val;
-		if (val) {
-			/* Digital */
-			snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
-			snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
-			snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
-				snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000);
-			mask = inl(emu->port + GPIO) & ~0x101;
-			outl(mask, emu->port + GPIO);
-
-		} else {
-			/* Analog */
-			snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
-			snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
-			snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
-				snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
-			mask = inl(emu->port + GPIO) | 0x101;
-			outl(mask, emu->port + GPIO);
-		}
+		ca0106_spdif_enable(emu);
 	}
         return change;
 }
@@ -154,8 +214,6 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
 	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
 	unsigned int val;
 	int change = 0;
-	u32 mask;
-	u32 source;
 
 	val = ucontrol->value.enumerated.item[0] ;
 	if (val >= 6)
@@ -163,9 +221,7 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
 	change = (emu->capture_source != val);
 	if (change) {
 		emu->capture_source = val;
-		source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
-		mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
-		snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
+		ca0106_set_capture_source(emu);
 	}
         return change;
 }
@@ -200,9 +256,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
 	unsigned int source_id;
-	unsigned int ngain, ogain;
 	int change = 0;
-	u32 source;
 	/* If the capture source has changed,
 	 * update the capture volume from the cached value
 	 * for the particular source.
@@ -212,18 +266,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
 		return -EINVAL;
 	change = (emu->i2c_capture_source != source_id);
 	if (change) {
-		snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
-		ngain = emu->i2c_capture_volume[source_id][0]; /* Left */
-		ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
-		if (ngain != ogain)
-			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff));
-		ngain = emu->i2c_capture_volume[source_id][1]; /* Left */
-		ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */
-		if (ngain != ogain)
-			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
-		source = 1 << source_id;
-		snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
-		emu->i2c_capture_source = source_id;
+		ca0106_set_i2c_capture_source(emu, source_id, 0);
 	}
         return change;
 }
@@ -271,7 +314,6 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
 	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
 	unsigned int val;
 	int change = 0;
-	u32 tmp;
 
 	val = ucontrol->value.enumerated.item[0] ;
 	if (val > 1)
@@ -279,18 +321,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
 	change = (emu->capture_mic_line_in != val);
 	if (change) {
 		emu->capture_mic_line_in = val;
-		if (val) {
-			//snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
-			tmp = inl(emu->port+GPIO) & ~0x400;
-			tmp = tmp | 0x400;
-			outl(tmp, emu->port+GPIO);
-			//snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC);
-		} else {
-			//snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
-			tmp = inl(emu->port+GPIO) & ~0x400;
-			outl(tmp, emu->port+GPIO);
-			//snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN);
-		}
+		ca0106_set_capture_mic_line_in(emu);
 	}
         return change;
 }
@@ -322,16 +353,33 @@ static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int snd_ca0106_spdif_get(struct snd_kcontrol *kcontrol,
+static void decode_spdif_bits(unsigned char *status, unsigned int bits)
+{
+	status[0] = (bits >> 0) & 0xff;
+	status[1] = (bits >> 8) & 0xff;
+	status[2] = (bits >> 16) & 0xff;
+	status[3] = (bits >> 24) & 0xff;
+}
+
+static int snd_ca0106_spdif_get_default(struct snd_kcontrol *kcontrol,
                                  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
 	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
 
-	ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
-	ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
-	ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
-	ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
+	decode_spdif_bits(ucontrol->value.iec958.status,
+			  emu->spdif_bits[idx]);
+        return 0;
+}
+
+static int snd_ca0106_spdif_get_stream(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+	decode_spdif_bits(ucontrol->value.iec958.status,
+			  emu->spdif_str_bits[idx]);
         return 0;
 }
 
@@ -345,24 +393,48 @@ static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol,
         return 0;
 }
 
-static int snd_ca0106_spdif_put(struct snd_kcontrol *kcontrol,
+static unsigned int encode_spdif_bits(unsigned char *status)
+{
+	return ((unsigned int)status[0] << 0) |
+		((unsigned int)status[1] << 8) |
+		((unsigned int)status[2] << 16) |
+		((unsigned int)status[3] << 24);
+}
+
+static int snd_ca0106_spdif_put_default(struct snd_kcontrol *kcontrol,
                                  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
 	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	int change;
 	unsigned int val;
 
-	val = (ucontrol->value.iec958.status[0] << 0) |
-	      (ucontrol->value.iec958.status[1] << 8) |
-	      (ucontrol->value.iec958.status[2] << 16) |
-	      (ucontrol->value.iec958.status[3] << 24);
-	change = val != emu->spdif_bits[idx];
-	if (change) {
-		snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val);
+	val = encode_spdif_bits(ucontrol->value.iec958.status);
+	if (val != emu->spdif_bits[idx]) {
 		emu->spdif_bits[idx] = val;
+		/* FIXME: this isn't safe, but needed to keep the compatibility
+		 * with older alsa-lib config
+		 */
+		emu->spdif_str_bits[idx] = val;
+		ca0106_set_spdif_bits(emu, idx);
+		return 1;
 	}
-        return change;
+	return 0;
+}
+
+static int snd_ca0106_spdif_put_stream(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	unsigned int val;
+
+	val = encode_spdif_bits(ucontrol->value.iec958.status);
+	if (val != emu->spdif_str_bits[idx]) {
+		emu->spdif_str_bits[idx] = val;
+		ca0106_set_spdif_bits(emu, idx);
+		return 1;
+	}
+        return 0;
 }
 
 static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol,
@@ -573,8 +645,16 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
 		.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
 		.count =	4,
 		.info =         snd_ca0106_spdif_info,
-		.get =          snd_ca0106_spdif_get,
-		.put =          snd_ca0106_spdif_put
+		.get =          snd_ca0106_spdif_get_default,
+		.put =          snd_ca0106_spdif_put_default
+	},
+	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
+		.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+		.count =	4,
+		.info =         snd_ca0106_spdif_info,
+		.get =          snd_ca0106_spdif_get_stream,
+		.put =          snd_ca0106_spdif_put_stream
 	},
 };
 
@@ -773,3 +853,50 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
         return 0;
 }
 
+#ifdef CONFIG_PM
+struct ca0106_vol_tbl {
+	unsigned int channel_id;
+	unsigned int reg;
+};
+
+static struct ca0106_vol_tbl saved_volumes[NUM_SAVED_VOLUMES] = {
+	{ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1 },
+	{ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1 },
+	{ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1 },
+	{ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1 },
+	{ 1, CAPTURE_CONTROL },
+};
+
+void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip)
+{
+	int i;
+
+	/* save volumes */
+	for (i = 0; i < NUM_SAVED_VOLUMES; i++)
+		chip->saved_vol[i] =
+			snd_ca0106_ptr_read(chip, saved_volumes[i].reg,
+					    saved_volumes[i].channel_id);
+}
+
+void snd_ca0106_mixer_resume(struct snd_ca0106  *chip)
+{
+	int i;
+
+	for (i = 0; i < NUM_SAVED_VOLUMES; i++)
+		snd_ca0106_ptr_write(chip, saved_volumes[i].reg,
+				     saved_volumes[i].channel_id,
+				     chip->saved_vol[i]);
+
+	ca0106_spdif_enable(chip);
+	ca0106_set_capture_source(chip);
+	ca0106_set_i2c_capture_source(chip, chip->i2c_capture_source, 1);
+	for (i = 0; i < 4; i++)
+		ca0106_set_spdif_bits(chip, i);
+	if (chip->details->i2c_adc)
+		ca0106_set_capture_mic_line_in(chip);
+}
+#endif /* CONFIG_PM */
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index fb6dc3980257..8ab07aa63652 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -3640,7 +3640,10 @@ int snd_cs46xx_resume(struct pci_dev *pci)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct snd_cs46xx *chip = card->private_data;
-	int i, amp_saved;
+	int amp_saved;
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+	int i;
+#endif
 
 	pci_set_power_state(pci, PCI_D0);
 	pci_restore_state(pci);
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index bb3d57e6a3cb..fda7a94c992f 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -4,6 +4,9 @@
 
 snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
 snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
+ifdef CONFIG_MGEODE_LX
+snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o
+endif
 
 # Toplevel Module Dependency
 obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 1d8b16052535..826e6dec2e97 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -159,10 +159,14 @@ static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au)
 		return err;
 
 	memset(&ac97, 0, sizeof(ac97));
-	ac97.scaps = AC97_SCAP_AUDIO|AC97_SCAP_SKIP_MODEM;
+	ac97.scaps = AC97_SCAP_AUDIO | AC97_SCAP_SKIP_MODEM
+			| AC97_SCAP_POWER_SAVE;
 	ac97.private_data = cs5535au;
 	ac97.pci = cs5535au->pci;
 
+	/* set any OLPC-specific scaps */
+	olpc_prequirks(card, &ac97);
+
 	if ((err = snd_ac97_mixer(pbus, &ac97, &cs5535au->ac97)) < 0) {
 		snd_printk(KERN_ERR "mixer failed\n");
 		return err;
@@ -170,6 +174,12 @@ static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au)
 
 	snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk);
 
+	err = olpc_quirks(card, cs5535au->ac97);
+	if (err < 0) {
+		snd_printk(KERN_ERR "olpc quirks failed\n");
+		return err;
+	}
+
 	return 0;
 }
 
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 66bae7664193..7a298ac662e3 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -78,6 +78,7 @@ struct cs5535audio_dma {
 	unsigned int buf_addr, buf_bytes;
 	unsigned int period_bytes, periods;
 	u32 saved_prd;
+	int pcm_open_flag;
 };
 
 struct cs5535audio {
@@ -93,8 +94,46 @@ struct cs5535audio {
 	struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS];
 };
 
+#ifdef CONFIG_PM
 int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
 int snd_cs5535audio_resume(struct pci_dev *pci);
+#endif
+
+#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX)
+void __devinit olpc_prequirks(struct snd_card *card,
+		struct snd_ac97_template *ac97);
+int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97);
+void olpc_analog_input(struct snd_ac97 *ac97, int on);
+void olpc_mic_bias(struct snd_ac97 *ac97, int on);
+
+static inline void olpc_capture_open(struct snd_ac97 *ac97)
+{
+	/* default to Analog Input off */
+	olpc_analog_input(ac97, 0);
+	/* enable MIC Bias for recording */
+	olpc_mic_bias(ac97, 1);
+}
+
+static inline void olpc_capture_close(struct snd_ac97 *ac97)
+{
+	/* disable Analog Input */
+	olpc_analog_input(ac97, 0);
+	/* disable the MIC Bias (so the recording LED turns off) */
+	olpc_mic_bias(ac97, 0);
+}
+#else
+static inline void olpc_prequirks(struct snd_card *card,
+		struct snd_ac97_template *ac97) { }
+static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
+{
+	return 0;
+}
+static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { }
+static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { }
+static inline void olpc_capture_open(struct snd_ac97 *ac97) { }
+static inline void olpc_capture_close(struct snd_ac97 *ac97) { }
+#endif
+
 int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio);
 
 #endif /* __SOUND_CS5535AUDIO_H */
diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c
new file mode 100644
index 000000000000..5c6814335cd7
--- /dev/null
+++ b/sound/pci/cs5535audio/cs5535audio_olpc.c
@@ -0,0 +1,179 @@
+/*
+ * OLPC XO-1 additional sound features
+ *
+ * Copyright © 2006  Jaya Kumar <jayakumar.lkml@gmail.com>
+ * Copyright © 2007-2008  Andres Salomon <dilinger@debian.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/ac97_codec.h>
+
+#include <asm/olpc.h>
+#include "cs5535audio.h"
+
+/*
+ * OLPC has an additional feature on top of the regular AD1888 codec features.
+ * It has an Analog Input mode that is switched into (after disabling the
+ * High Pass Filter) via GPIO.  It is supported on B2 and later models.
+ */
+void olpc_analog_input(struct snd_ac97 *ac97, int on)
+{
+	int err;
+
+	if (!machine_is_olpc())
+		return;
+
+	/* update the High Pass Filter (via AC97_AD_TEST2) */
+	err = snd_ac97_update_bits(ac97, AC97_AD_TEST2,
+			1 << AC97_AD_HPFD_SHIFT, on << AC97_AD_HPFD_SHIFT);
+	if (err < 0) {
+		snd_printk(KERN_ERR "setting High Pass Filter - %d\n", err);
+		return;
+	}
+
+	/* set Analog Input through GPIO */
+	if (on)
+		geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
+	else
+		geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
+}
+
+/*
+ * OLPC XO-1's V_REFOUT is a mic bias enable.
+ */
+void olpc_mic_bias(struct snd_ac97 *ac97, int on)
+{
+	int err;
+
+	if (!machine_is_olpc())
+		return;
+
+	on = on ? 0 : 1;
+	err = snd_ac97_update_bits(ac97, AC97_AD_MISC,
+			1 << AC97_AD_VREFD_SHIFT, on << AC97_AD_VREFD_SHIFT);
+	if (err < 0)
+		snd_printk(KERN_ERR "setting MIC Bias - %d\n", err);
+}
+
+static int olpc_dc_info(struct snd_kcontrol *kctl,
+		struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
+{
+	v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC,
+			GPIO_OUTPUT_VAL);
+	return 0;
+}
+
+static int olpc_dc_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
+{
+	struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl);
+
+	olpc_analog_input(cs5535au->ac97, v->value.integer.value[0]);
+	return 1;
+}
+
+static int olpc_mic_info(struct snd_kcontrol *kctl,
+		struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+static int olpc_mic_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
+{
+	struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl);
+	struct snd_ac97 *ac97 = cs5535au->ac97;
+	int i;
+
+	i = (snd_ac97_read(ac97, AC97_AD_MISC) >> AC97_AD_VREFD_SHIFT) & 0x1;
+	v->value.integer.value[0] = i ? 0 : 1;
+	return 0;
+}
+
+static int olpc_mic_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
+{
+	struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl);
+
+	olpc_mic_bias(cs5535au->ac97, v->value.integer.value[0]);
+	return 1;
+}
+
+static struct snd_kcontrol_new olpc_cs5535audio_ctls[] __devinitdata = {
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "DC Mode Enable",
+	.info = olpc_dc_info,
+	.get = olpc_dc_get,
+	.put = olpc_dc_put,
+	.private_value = 0,
+},
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "MIC Bias Enable",
+	.info = olpc_mic_info,
+	.get = olpc_mic_get,
+	.put = olpc_mic_put,
+	.private_value = 0,
+},
+};
+
+void __devinit olpc_prequirks(struct snd_card *card,
+		struct snd_ac97_template *ac97)
+{
+	if (!machine_is_olpc())
+		return;
+
+	/* invert EAPD if on an OLPC B3 or higher */
+	if (olpc_board_at_least(olpc_board_pre(0xb3)))
+		ac97->scaps |= AC97_SCAP_INV_EAPD;
+}
+
+int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
+{
+	struct snd_ctl_elem_id elem;
+	int i, err;
+
+	if (!machine_is_olpc())
+		return 0;
+
+	/* drop the original AD1888 HPF control */
+	memset(&elem, 0, sizeof(elem));
+	elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	strncpy(elem.name, "High Pass Filter Enable", sizeof(elem.name));
+	snd_ctl_remove_id(card, &elem);
+
+	/* drop the original V_REFOUT control */
+	memset(&elem, 0, sizeof(elem));
+	elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	strncpy(elem.name, "V_REFOUT Enable", sizeof(elem.name));
+	snd_ctl_remove_id(card, &elem);
+
+	/* add the OLPC-specific controls */
+	for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) {
+		err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i],
+				ac97->private_data));
+		if (err < 0)
+			return err;
+	}
+
+	/* turn off the mic by default */
+	olpc_mic_bias(ac97, 0);
+	return 0;
+}
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index cdcda87116c3..0f48a871f17b 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -260,6 +260,9 @@ static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream,
 	err = cs5535audio_build_dma_packets(cs5535au, dma, substream,
 					    params_periods(hw_params),
 					    params_period_bytes(hw_params));
+	if (!err)
+		dma->pcm_open_flag = 1;
+
 	return err;
 }
 
@@ -268,6 +271,15 @@ static int snd_cs5535audio_hw_free(struct snd_pcm_substream *substream)
 	struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
 	struct cs5535audio_dma *dma = substream->runtime->private_data;
 
+	if (dma->pcm_open_flag) {
+		if (substream == cs5535au->playback_substream)
+			snd_ac97_update_power(cs5535au->ac97,
+					AC97_PCM_FRONT_DAC_RATE, 0);
+		else
+			snd_ac97_update_power(cs5535au->ac97,
+					AC97_PCM_LR_ADC_RATE, 0);
+		dma->pcm_open_flag = 0;
+	}
 	cs5535audio_clear_dma_packets(cs5535au, dma, substream);
 	return snd_pcm_lib_free_pages(substream);
 }
@@ -351,11 +363,14 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream)
 	if ((err = snd_pcm_hw_constraint_integer(runtime,
 					 SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 		return err;
+	olpc_capture_open(cs5535au->ac97);
 	return 0;
 }
 
 static int snd_cs5535audio_capture_close(struct snd_pcm_substream *substream)
 {
+	struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+	olpc_capture_close(cs5535au->ac97);
 	return 0;
 }
 
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index de5ee8f097f6..7958006a1d66 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -69,7 +69,7 @@ MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME);
  * EMU10K1 init / done
  *************************************************************************/
 
-void snd_emu10k1_voice_init(struct snd_emu10k1 * emu, int ch)
+void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch)
 {
 	snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0);
 	snd_emu10k1_ptr_write(emu, IP, ch, 0);
@@ -151,9 +151,9 @@ static unsigned int i2c_adc_init[][2] = {
 	{ 0x12, 0x32 },  /* ALC Control 3 */
 	{ 0x13, 0x00 },  /* Noise gate control */
 	{ 0x14, 0xa6 },  /* Limiter control */
-	{ 0x15, ADC_MUX_2 },  /* ADC Mixer control. Mic for Audigy 2 ZS Notebook */
+	{ 0x15, ADC_MUX_2 },  /* ADC Mixer control. Mic for A2ZS Notebook */
 };
-	
+
 static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 {
 	unsigned int silent_page;
@@ -161,8 +161,8 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 	u32 tmp;
 
 	/* disable audio and lock cache */
-	outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE,
-	     emu->port + HCFG);
+	outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK |
+		HCFG_MUTEBUTTONENABLE, emu->port + HCFG);
 
 	/* reset recording buffers */
 	snd_emu10k1_ptr_write(emu, MICBS, 0, ADCBS_BUFSIZE_NONE);
@@ -179,7 +179,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 	snd_emu10k1_ptr_write(emu, SOLEL, 0, 0);
 	snd_emu10k1_ptr_write(emu, SOLEH, 0, 0);
 
-	if (emu->audigy){
+	if (emu->audigy) {
 		/* set SPDIF bypass mode */
 		snd_emu10k1_ptr_write(emu, SPBYPASS, 0, SPBYPASS_FORMAT);
 		/* enable rear left + rear right AC97 slots */
@@ -197,12 +197,12 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 
 	if (emu->card_capabilities->ca0151_chip) { /* audigy2 */
 		/* Hacks for Alice3 to work independent of haP16V driver */
-		//Setup SRCMulti_I2S SamplingRate
+		/* Setup SRCMulti_I2S SamplingRate */
 		tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0);
 		tmp &= 0xfffff1ff;
 		tmp |= (0x2<<9);
 		snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp);
-		
+
 		/* Setup SRCSel (Enable Spdif,I2S SRCMulti) */
 		snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14);
 		/* Setup SRCMulti Input Audio Enable */
@@ -217,7 +217,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 	if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */
 		/* Hacks for Alice3 to work independent of haP16V driver */
 		snd_printk(KERN_INFO "Audigy2 value: Special config.\n");
-		//Setup SRCMulti_I2S SamplingRate
+		/* Setup SRCMulti_I2S SamplingRate */
 		tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0);
 		tmp &= 0xfffff1ff;
 		tmp |= (0x2<<9);
@@ -270,13 +270,13 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 		size = ARRAY_SIZE(i2c_adc_init);
 		for (n = 0; n < size; n++)
 			snd_emu10k1_i2c_write(emu, i2c_adc_init[n][0], i2c_adc_init[n][1]);
-		for (n=0; n < 4; n++) {
-			emu->i2c_capture_volume[n][0]= 0xcf;
-			emu->i2c_capture_volume[n][1]= 0xcf;
+		for (n = 0; n < 4; n++) {
+			emu->i2c_capture_volume[n][0] = 0xcf;
+			emu->i2c_capture_volume[n][1] = 0xcf;
 		}
 	}
 
-	
+
 	snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr);
 	snd_emu10k1_ptr_write(emu, TCB, 0, 0);	/* taken from original driver */
 	snd_emu10k1_ptr_write(emu, TCBS, 0, 4);	/* taken from original driver */
@@ -313,7 +313,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 	    (emu->model == 0x21 && emu->revision < 6))
 		outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE, emu->port + HCFG);
 	else
-		// With on-chip joystick
+		/* With on-chip joystick */
 		outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG);
 
 	if (enable_ir) {	/* enable IR for SB Live */
@@ -335,9 +335,9 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
 			outl(reg | HCFG_GPOUT1 | HCFG_GPOUT2, emu->port + HCFG);
 			udelay(100);
 			outl(reg, emu->port + HCFG);
- 		}
+		}
 	}
-	
+
 	if (emu->card_capabilities->emu_model) {
 		;  /* Disable all access to A_IOCFG for the emu1010 */
 	} else if (emu->card_capabilities->i2c_adc) {
@@ -364,7 +364,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu)
 		;  /* Disable A_IOCFG for Audigy 2 ZS Notebook */
 	} else if (emu->audigy) {
 		outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG);
- 
+
 		if (emu->card_capabilities->ca0151_chip) { /* audigy2 */
 			/* Unmute Analog now.  Set GPO6 to 1 for Apollo.
 			 * This has to be done after init ALice3 I2SOut beyond 48KHz.
@@ -378,12 +378,12 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu)
 			outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG);
 		}
 	}
-	
+
 #if 0
 	{
 	unsigned int tmp;
 	/* FIXME: the following routine disables LiveDrive-II !! */
-	// TOSLink detection
+	/* TOSLink detection */
 	emu->tos_link = 0;
 	tmp = inl(emu->port + HCFG);
 	if (tmp & (HCFG_GPINPUT0 | HCFG_GPINPUT1)) {
@@ -400,7 +400,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu)
 	snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE);
 }
 
-int snd_emu10k1_done(struct snd_emu10k1 * emu)
+int snd_emu10k1_done(struct snd_emu10k1 *emu)
 {
 	int ch;
 
@@ -495,7 +495,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu)
 
 #define EC_LAST_PROMFILE_ADDR	0x2f
 
-#define EC_SERIALNUM_ADDR	0x30	/* First word of serial number.  The 
+#define EC_SERIALNUM_ADDR	0x30	/* First word of serial number.  The
 					 * can be up to 30 characters in length
 					 * and is stored as a NULL-terminated
 					 * ASCII string.  Any unused bytes must be
@@ -503,8 +503,8 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu)
 #define EC_CHECKSUM_ADDR	0x3f	/* Location at which checksum is stored */
 
 
-/* Most of this stuff is pretty self-evident.  According to the hardware 
- * dudes, we need to leave the ADCCAL bit low in order to avoid a DC 
+/* Most of this stuff is pretty self-evident.  According to the hardware
+ * dudes, we need to leave the ADCCAL bit low in order to avoid a DC
  * offset problem.  Weird.
  */
 #define EC_RAW_RUN_MODE		(EC_DACMUTEN | EC_ADCRSTN | EC_TRIM_MUTEN | \
@@ -523,7 +523,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu)
  *  register.
  */
 
-static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value)
+static void snd_emu10k1_ecard_write(struct snd_emu10k1 *emu, unsigned int value)
 {
 	unsigned short count;
 	unsigned int data;
@@ -561,7 +561,7 @@ static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value
  * channel.
  */
 
-static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu,
+static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 *emu,
 					 unsigned short gain)
 {
 	unsigned int bit;
@@ -574,7 +574,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu,
 
 	for (bit = (1 << 15); bit; bit >>= 1) {
 		unsigned int value;
-		
+
 		value = emu->ecard_ctrl & ~(EC_TRIM_CSN | EC_TRIM_SDATA);
 
 		if (gain & bit)
@@ -589,7 +589,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu,
 	snd_emu10k1_ecard_write(emu, emu->ecard_ctrl);
 }
 
-static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu)
+static int snd_emu10k1_ecard_init(struct snd_emu10k1 *emu)
 {
 	unsigned int hc_value;
 
@@ -598,7 +598,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu)
 			  EC_SPDIF0_SELECT(EC_DEFAULT_SPDIF0_SEL) |
 			  EC_SPDIF1_SELECT(EC_DEFAULT_SPDIF1_SEL);
 
-	/* Step 0: Set the codec type in the hardware control register 
+	/* Step 0: Set the codec type in the hardware control register
 	 * and enable audio output */
 	hc_value = inl(emu->port + HCFG);
 	outl(hc_value | HCFG_AUDIOENABLE | HCFG_CODECFORMAT_I2S, emu->port + HCFG);
@@ -629,7 +629,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu)
 	return 0;
 }
 
-static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu)
+static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu)
 {
 	unsigned long special_port;
 	unsigned int value;
@@ -656,7 +656,7 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu)
 	return 0;
 }
 
-static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * filename)
+static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filename)
 {
 	int err;
 	int n, i;
@@ -666,11 +666,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
 	unsigned long flags;
 	const struct firmware *fw_entry;
 
-	if ((err = request_firmware(&fw_entry, filename, &emu->pci->dev)) != 0) {
-		snd_printk(KERN_ERR "firmware: %s not found. Err=%d\n",filename, err);
+	err = request_firmware(&fw_entry, filename, &emu->pci->dev);
+	if (err != 0) {
+		snd_printk(KERN_ERR "firmware: %s not found. Err = %d\n", filename, err);
 		return err;
 	}
-	snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size);
+	snd_printk(KERN_INFO "firmware size = 0x%zx\n", fw_entry->size);
 
 	/* The FPGA is a Xilinx Spartan IIE XC2S50E */
 	/* GPIO7 -> FPGA PGMN
@@ -685,13 +686,13 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
 	outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */
 	write_post = inl(emu->port + A_IOCFG);
 	udelay(100); /* Allow FPGA memory to clean */
-	for(n = 0; n < fw_entry->size; n++) {
-		value=fw_entry->data[n];	
-		for(i = 0; i < 8; i++) {
+	for (n = 0; n < fw_entry->size; n++) {
+		value = fw_entry->data[n];
+		for (i = 0; i < 8; i++) {
 			reg = 0x80;
 			if (value & 0x1)
 				reg = reg | 0x20;
-			value = value >> 1;   
+			value = value >> 1;
 			outl(reg, emu->port + A_IOCFG);
 			write_post = inl(emu->port + A_IOCFG);
 			outl(reg | 0x40, emu->port + A_IOCFG);
@@ -703,14 +704,14 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
 	write_post = inl(emu->port + A_IOCFG);
 	spin_unlock_irqrestore(&emu->emu_lock, flags);
 
-        release_firmware(fw_entry);
+	release_firmware(fw_entry);
 	return 0;
 }
 
 static int emu1010_firmware_thread(void *data)
 {
-	struct snd_emu10k1 * emu = data;
-	int tmp,tmp2;
+	struct snd_emu10k1 *emu = data;
+	int tmp, tmp2;
 	int reg;
 	int err;
 
@@ -719,50 +720,50 @@ static int emu1010_firmware_thread(void *data)
 		msleep_interruptible(1000);
 		if (kthread_should_stop())
 			break;
-		snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */
-		snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */
+		snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); /* IRQ Status */
+		snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); /* OPTIONS: Which cards are attached to the EMU */
 		if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) {
 			/* Audio Dock attached */
 			/* Return to Audio Dock programming mode */
 			snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
-			snd_emu1010_fpga_write(emu,  EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
+			snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK);
 			if (emu->card_capabilities->emu_model ==
 			    EMU_MODEL_EMU1010) {
-				if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
+				err = snd_emu1010_load_firmware(emu, DOCK_FILENAME);
+				if (err != 0)
 					continue;
-				}
 			} else if (emu->card_capabilities->emu_model ==
 				   EMU_MODEL_EMU1010B) {
-				if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+				err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME);
+				if (err != 0)
 					continue;
-				}
 			} else if (emu->card_capabilities->emu_model ==
 				   EMU_MODEL_EMU1616) {
-				if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+				err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME);
+				if (err != 0)
 					continue;
-				}
 			}
 
-			snd_emu1010_fpga_write(emu,  EMU_HANA_FPGA_CONFIG, 0 );
-			snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
-			snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
+			snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0);
+			snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg);
+			snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x%x\n", reg);
 			/* ID, should read & 0x7f = 0x55 when FPGA programmed. */
-			snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
-			snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
+			snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg);
+			snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", reg);
 			if ((reg & 0x1f) != 0x15) {
 				/* FPGA failed to be programmed */
-				snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
+				snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg = 0x%x\n", reg);
 				continue;
 			}
 			snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
-			snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp );
-			snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 );
-			snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2);
+			snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp);
+			snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2);
+			snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2);
 			/* Sync clocking between 1010 and Dock */
 			/* Allow DLL to settle */
 			msleep(10);
 			/* Unmute all. Default is muted after a firmware load */
-			snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+			snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE);
 		}
 	}
 	snd_printk(KERN_INFO "emu1010: firmware thread stopping\n");
@@ -800,10 +801,10 @@ static int emu1010_firmware_thread(void *data)
  *		16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops
  * 		16 x 32-bit capture - snd_emu10k1_capture_efx_ops
  */
-static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
+static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
 {
 	unsigned int i;
-	int tmp,tmp2;
+	int tmp, tmp2;
 	int reg;
 	int err;
 	const char *filename = NULL;
@@ -818,7 +819,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 	 * Lock Tank Memory Cache,
 	 * Mute all codecs.
 	 */
-	outl(0x0005a004, emu->port + HCFG); 
+	outl(0x0005a004, emu->port + HCFG);
 	/* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave,
 	 * Mute all codecs.
 	 */
@@ -829,25 +830,25 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 	outl(0x0005a000, emu->port + HCFG);
 
 	/* Disable 48Volt power to Audio Dock */
-	snd_emu1010_fpga_write(emu,  EMU_HANA_DOCK_PWR,  0 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0);
 
 	/* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */
-	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
-	snd_printdd("reg1=0x%x\n",reg);
+	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg);
+	snd_printdd("reg1 = 0x%x\n", reg);
 	if ((reg & 0x3f) == 0x15) {
 		/* FPGA netlist already present so clear it */
 		/* Return to programming mode */
 
-		snd_emu1010_fpga_write(emu,  EMU_HANA_FPGA_CONFIG, 0x02 );
+		snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02);
 	}
-	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
-	snd_printdd("reg2=0x%x\n",reg);
+	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg);
+	snd_printdd("reg2 = 0x%x\n", reg);
 	if ((reg & 0x3f) == 0x15) {
 		/* FPGA failed to return to programming mode */
 		snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n");
 		return -ENODEV;
 	}
-	snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg);
+	snd_printk(KERN_INFO "emu1010: EMU_HANA_ID = 0x%x\n", reg);
 	switch (emu->card_capabilities->emu_model) {
 	case EMU_MODEL_EMU1010:
 		filename = HANA_FILENAME;
@@ -876,25 +877,25 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 	}
 
 	/* ID, should read & 0x7f = 0x55 when FPGA programmed. */
-	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg);
 	if ((reg & 0x3f) != 0x15) {
 		/* FPGA failed to be programmed */
-		snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg);
+		snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg = 0x%x\n", reg);
 		return -ENODEV;
 	}
 
 	snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n");
-	snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp );
-	snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2 );
-	snd_printk("Hana ver:%d.%d\n",tmp ,tmp2);
+	snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp);
+	snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2);
+	snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2);
 	/* Enable 48Volt power to Audio Dock */
-	snd_emu1010_fpga_write(emu,  EMU_HANA_DOCK_PWR,  EMU_HANA_DOCK_PWR_ON );
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON);
 
-	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
-	snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg);
-	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
-	snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg);
-	snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); 
+	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg);
+	snd_printk(KERN_INFO "emu1010: Card options = 0x%x\n", reg);
+	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg);
+	snd_printk(KERN_INFO "emu1010: Card options = 0x%x\n", reg);
+	snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp);
 	/* Optical -> ADAT I/O  */
 	/* 0 : SPDIF
 	 * 1 : ADAT
@@ -904,41 +905,42 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 	tmp = 0;
 	tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) |
 		(emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0);
-	snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp );
-	snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp );
+	snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp);
+	snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp);
 	/* Set no attenuation on Audio Dock pads. */
-	snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00);
 	emu->emu1010.adc_pads = 0x00;
-	snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp );
+	snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp);
 	/* Unmute Audio dock DACs, Headphone source DAC-4. */
-	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 );
-	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 );
-	snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp );
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30);
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);
+	snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp);
 	/* DAC PADs. */
-	snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f );
+	snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f);
 	emu->emu1010.dac_pads = 0x0f;
-	snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp );
-	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 );
-	snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp );
+	snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp);
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30);
+	snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp);
 	/* SPDIF Format. Set Consumer mode, 24bit, copy enable */
-	snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10);
 	/* MIDI routing */
-	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19);
 	/* Unknown. */
-	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c );
-	/* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); // IRQ Enable: All on */
+	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c);
+	/* IRQ Enable: Alll on */
+	/* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); */
 	/* IRQ Enable: All off */
-	snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00);
 
-	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
-	snd_printk(KERN_INFO "emu1010: Card options3=0x%x\n",reg);
+	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg);
+	snd_printk(KERN_INFO "emu1010: Card options3 = 0x%x\n", reg);
 	/* Default WCLK set to 48kHz. */
-	snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00);
 	/* Word Clock source, Internal 48kHz x1 */
-	snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K );
-	//snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X );
+	snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K);
+	/* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */
 	/* Audio Dock LEDs. */
-	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 );
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);
 
 #if 0
 	/* For 96kHz */
@@ -992,7 +994,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 	 * Defaults only, users will set their own values anyways, let's
 	 * just copy/paste.
 	 */
-	
+
 	snd_emu1010_fpga_link_dst_src_write(emu,
 		EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1);
 	snd_emu1010_fpga_link_dst_src_write(emu,
@@ -1037,19 +1039,19 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 	snd_emu1010_fpga_link_dst_src_write(emu,
 		EMU_DST_ALICE2_EMU32_F, EMU_SRC_HAMOA_ADC_LEFT2);
 #endif
-	for (i = 0;i < 0x20; i++ ) {
-		/* AudioDock Elink <-  Silence */
-		snd_emu1010_fpga_link_dst_src_write(emu, 0x0100+i, EMU_SRC_SILENCE);
+	for (i = 0; i < 0x20; i++) {
+		/* AudioDock Elink <- Silence */
+		snd_emu1010_fpga_link_dst_src_write(emu, 0x0100 + i, EMU_SRC_SILENCE);
 	}
-	for (i = 0;i < 4; i++) {
+	for (i = 0; i < 4; i++) {
 		/* Hana SPDIF Out <- Silence */
-		snd_emu1010_fpga_link_dst_src_write(emu, 0x0200+i, EMU_SRC_SILENCE);
+		snd_emu1010_fpga_link_dst_src_write(emu, 0x0200 + i, EMU_SRC_SILENCE);
 	}
-	for (i = 0;i < 7; i++) {
+	for (i = 0; i < 7; i++) {
 		/* Hamoa DAC <- Silence */
-		snd_emu1010_fpga_link_dst_src_write(emu, 0x0300+i, EMU_SRC_SILENCE);
+		snd_emu1010_fpga_link_dst_src_write(emu, 0x0300 + i, EMU_SRC_SILENCE);
 	}
-	for (i = 0;i < 7; i++) {
+	for (i = 0; i < 7; i++) {
 		/* Hana ADAT Out <- Silence */
 		snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HANA_ADAT + i, EMU_SRC_SILENCE);
 	}
@@ -1065,30 +1067,30 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 		EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1);
 	snd_emu1010_fpga_link_dst_src_write(emu,
 		EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1);
-	snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01 ); // Unmute all
+	snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01); /* Unmute all */
+
+	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp);
 
-	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp );
-	
 	/* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave,
 	 * Lock Sound Memory Cache, Lock Tank Memory Cache,
 	 * Mute all codecs.
 	 */
-	outl(0x0000a000, emu->port + HCFG); 
+	outl(0x0000a000, emu->port + HCFG);
 	/* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave,
 	 * Lock Sound Memory Cache, Lock Tank Memory Cache,
 	 * Un-Mute all codecs.
 	 */
 	outl(0x0000a001, emu->port + HCFG);
- 
+
 	/* Initial boot complete. Now patches */
 
-	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp );
-	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */
-	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */
-	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */
-	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */
-	snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); 
-	snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif  (or 0x11 for aes/ebu) */
+	snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp);
+	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */
+	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */
+	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */
+	snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */
+	snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp);
+	snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* SPDIF Format spdif  (or 0x11 for aes/ebu) */
 
 	/* Start Micro/Audio Dock firmware loader thread */
 	if (!emu->emu1010.firmware_thread) {
@@ -1218,20 +1220,20 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
 		emu->emu1010.output_source[23] = 28;
 	}
 	/* TEMP: Select SPDIF in/out */
-	//snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */
+	/* snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); */ /* Output spdif */
 
 	/* TEMP: Select 48kHz SPDIF out */
 	snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */
 	snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */
 	/* Word Clock source, Internal 48kHz x1 */
-	snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K );
-	//snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X );
+	snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K);
+	/* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */
 	emu->emu1010.internal_clock = 1; /* 48000 */
-	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);/* Set LEDs on Audio Dock */
+	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); /* Set LEDs on Audio Dock */
 	snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */
-	//snd_emu1010_fpga_write(emu, 0x7, 0x0); /* Mute all */
-	//snd_emu1010_fpga_write(emu, 0x7, 0x1); /* Unmute all */
-	//snd_emu1010_fpga_write(emu, 0xe, 0x12); /* Set LEDs on Audio Dock */
+	/* snd_emu1010_fpga_write(emu, 0x7, 0x0); */ /* Mute all */
+	/* snd_emu1010_fpga_write(emu, 0x7, 0x1); */ /* Unmute all */
+	/* snd_emu1010_fpga_write(emu, 0xe, 0x12); */ /* Set LEDs on Audio Dock */
 
 	return 0;
 }
@@ -1247,13 +1249,13 @@ static void free_pm_buffer(struct snd_emu10k1 *emu);
 static int snd_emu10k1_free(struct snd_emu10k1 *emu)
 {
 	if (emu->port) {	/* avoid access to already used hardware */
-	       	snd_emu10k1_fx8010_tram_setup(emu, 0);
+		snd_emu10k1_fx8010_tram_setup(emu, 0);
 		snd_emu10k1_done(emu);
 		snd_emu10k1_free_efx(emu);
-       	}
+	}
 	if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) {
 		/* Disable 48Volt power to Audio Dock */
-		snd_emu1010_fpga_write(emu,  EMU_HANA_DOCK_PWR,  0 );
+		snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0);
 	}
 	if (emu->emu1010.firmware_thread)
 		kthread_stop(emu->emu1010.firmware_thread);
@@ -1278,7 +1280,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
 #endif
 	if (emu->port)
 		pci_release_regions(emu->pci);
-	if (emu->card_capabilities->ca0151_chip) /* P16V */	
+	if (emu->card_capabilities->ca0151_chip) /* P16V */
 		snd_p16v_free(emu);
 	pci_disable_device(emu->pci);
 	kfree(emu);
@@ -1292,21 +1294,6 @@ static int snd_emu10k1_dev_free(struct snd_device *device)
 }
 
 static struct snd_emu_chip_details emu_chip_details[] = {
-	/* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/
-	/* Tested by James@superbug.co.uk 3rd July 2005 */
-	/* DSP: CA0108-IAT
-	 * DAC: CS4382-KQ
-	 * ADC: Philips 1361T
-	 * AC97: STAC9750
-	 * CA0151: None
-	 */
-	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102,
-	 .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", 
-	 .id = "Audigy2",
-	 .emu10k2_chip = 1,
-	 .ca0108_chip = 1,
-	 .spk71 = 1,
-	 .ac97_chip = 1} ,
 	/* Audigy4 (Not PRO) SB0610 */
 	/* Tested by James@superbug.co.uk 4th April 2006 */
 	/* A_IOCFG bits
@@ -1346,20 +1333,37 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 * CA0151: None
 	 */
 	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102,
-	 .driver = "Audigy2", .name = "Audigy 4 [SB0610]", 
+	 .driver = "Audigy2", .name = "SB Audigy 4 [SB0610]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
 	 .spk71 = 1,
 	 .adc_1361t = 1,  /* 24 bit capture instead of 16bit */
 	 .ac97_chip = 1} ,
+	/* Audigy 2 Value AC3 out does not work yet.
+	 * Need to find out how to turn off interpolators.
+	 */
+	/* Tested by James@superbug.co.uk 3rd July 2005 */
+	/* DSP: CA0108-IAT
+	 * DAC: CS4382-KQ
+	 * ADC: Philips 1361T
+	 * AC97: STAC9750
+	 * CA0151: None
+	 */
+	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102,
+	 .driver = "Audigy2", .name = "SB Audigy 2 Value [SB0400]",
+	 .id = "Audigy2",
+	 .emu10k2_chip = 1,
+	 .ca0108_chip = 1,
+	 .spk71 = 1,
+	 .ac97_chip = 1} ,
 	/* Audigy 2 ZS Notebook Cardbus card.*/
 	/* Tested by James@superbug.co.uk 6th November 2006 */
 	/* Audio output 7.1/Headphones working.
 	 * Digital output working. (AC3 not checked, only PCM)
 	 * Audio Mic/Line inputs working.
 	 * Digital input not tested.
-	 */ 
+	 */
 	/* DSP: Tina2
 	 * DAC: Wolfson WM8768/WM8568
 	 * ADC: Wolfson WM8775
@@ -1386,7 +1390,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 *
 	 */
 	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102,
-	 .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 ZS Notebook [SB0530]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
@@ -1396,7 +1400,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .spk71 = 1} ,
 	/* Tested by James@superbug.co.uk 4th Nov 2007. */
 	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102,
-	 .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", 
+	 .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]",
 	 .id = "EMU1010",
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
@@ -1404,47 +1408,49 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .spk71 = 1 ,
 	 .emu_model = EMU_MODEL_EMU1616},
 	/* Tested by James@superbug.co.uk 4th Nov 2007. */
+	/* This is MAEM8960, 0202 is MAEM 8980 */
 	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102,
-	 .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", 
+	 .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM8960]",
 	 .id = "EMU1010",
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
 	 .spk71 = 1,
-	 .emu_model = EMU_MODEL_EMU1010B},
+	 .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */
 	/* Tested by James@superbug.co.uk 8th July 2005. */
+	/* This is MAEM8810, 0202 is MAEM8820 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102,
-	 .driver = "Audigy2", .name = "E-mu 1010 [4001]",
+	 .driver = "Audigy2", .name = "E-mu 1010 [MAEM8810]",
 	 .id = "EMU1010",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .spk71 = 1,
-	 .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */
+	 .emu_model = EMU_MODEL_EMU1010}, /* EMU 1010 old revision */
 	/* EMU0404b */
 	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102,
-	 .driver = "Audigy2", .name = "E-mu 0404b [4002]",
+	 .driver = "Audigy2", .name = "E-mu 0404b PCI [MAEM8852]",
 	 .id = "EMU0404",
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
 	 .spk71 = 1,
-	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 new revision */
 	/* Tested by James@superbug.co.uk 20-3-2007. */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102,
-	 .driver = "Audigy2", .name = "E-mu 0404 [4002]",
+	 .driver = "Audigy2", .name = "E-mu 0404 [MAEM8850]",
 	 .id = "EMU0404",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .spk71 = 1,
 	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
-	/* Audigy4 (Not PRO) SB0610 */
-	{.vendor = 0x1102, .device = 0x0008, 
-	 .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", 
+	/* Note that all E-mu cards require kernel 2.6 or newer. */
+	{.vendor = 0x1102, .device = 0x0008,
+	 .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
 	 .ac97_chip = 1} ,
 	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102,
-	 .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", 
+	 .driver = "Audigy2", .name = "SB Audigy 4 PRO [SB0380]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1457,7 +1463,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 * Just like 0x20021102
 	 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20061102,
-	 .driver = "Audigy2", .name = "Audigy 2 [SB0350b]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 [SB0350b]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1467,7 +1473,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102,
-	 .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0350]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1477,7 +1483,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102,
-	 .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0360]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1495,7 +1501,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 * CA0151: Yes
 	 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102,
-	 .driver = "Audigy2", .name = "Audigy 2 [SB0240]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 [SB0240]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1505,7 +1511,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .adc_1361t = 1,  /* 24 bit capture instead of 16bit */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102,
-	 .driver = "Audigy2", .name = "Audigy 2 EX [1005]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 Platinum EX [SB0280]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1515,7 +1521,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	/* Dell OEM/Creative Labs Audigy 2 ZS */
 	/* See ALSA bug#1365 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10031102,
-	 .driver = "Audigy2", .name = "Audigy 2 ZS [SB0353]",
+	 .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0353]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1524,7 +1530,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .spdif_bug = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
-	 .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]", 
+	 .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1535,7 +1541,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .adc_1361t = 1,  /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .revision = 0x04,
-	 .driver = "Audigy2", .name = "Audigy 2 [Unknown]",
+	 .driver = "Audigy2", .name = "SB Audigy 2 [Unknown]",
 	 .id = "Audigy2",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
@@ -1543,78 +1549,79 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .spdif_bug = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102,
-	 .driver = "Audigy", .name = "Audigy 1 [SB0090]", 
+	 .driver = "Audigy", .name = "SB Audigy 1 [SB0092]",
 	 .id = "Audigy",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102,
-	 .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", 
+	 .driver = "Audigy", .name = "SB Audigy 1 ES [SB0160]",
 	 .id = "Audigy",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .spdif_bug = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102,
-	 .driver = "Audigy", .name = "Audigy 1 [SB0090]", 
+	 .driver = "Audigy", .name = "SB Audigy 1 [SB0090]",
 	 .id = "Audigy",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004,
-	 .driver = "Audigy", .name = "Audigy 1 [Unknown]", 
+	 .driver = "Audigy", .name = "Audigy 1 [Unknown]",
 	 .id = "Audigy",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .ac97_chip = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102,
-	 .driver = "EMU10K1", .name = "SBLive! [SB0105]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102,
+	 .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0220]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806b1102,
+	 .driver = "EMU10K1", .name = "SB Live! [SB0105]",
+	 .id = "Live",
+	 .emu10k1_chip = 1,
+	 .ac97_chip = 1,
+	 .sblive51 = 1} ,
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806a1102,
+	 .driver = "EMU10K1", .name = "SB Live! Value [SB0103]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [SB0101]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	/* Tested by ALSA bug#1680 26th December 2005 */
-	/* note: It really has SB0220 written on the card. */
+	/* note: It really has SB0220 written on the card, */
+	/* but it's SB0228 according to kx.inf */
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80661102,
-	 .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]", 
+	 .driver = "EMU10K1", .name = "SB Live! 5.1 Dell OEM [SB0228]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	/* Tested by Thomas Zehetbauer 27th Aug 2005 */
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102,
-	 .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", 
-	 .id = "Live",
-	 .emu10k1_chip = 1,
-	 .ac97_chip = 1,
-	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102,
-	 .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", 
+	 .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0220]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102,
-	 .driver = "EMU10K1", .name = "SB Live 5.1", 
+	 .driver = "EMU10K1", .name = "SB Live! 5.1",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	/* Tested by alsa bugtrack user "hus" bug #1297 12th Aug 2005 */
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
-	 .driver = "EMU10K1", .name = "SBLive 5.1 [SB0060]",
+	 .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0060]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 2, /* ac97 is optional; both SBLive 5.1 and platinum
@@ -1622,78 +1629,78 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 			  */
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4850]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102,
-	 .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", 
+	 .driver = "EMU10K1", .name = "SB Live! Platinum [CT4760P]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4871]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4831]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4831]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4870]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4832]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4830]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102,
-	 .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", 
+	 .driver = "EMU10K1", .name = "SB PCI512 [CT4790]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4780]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102,
-	 .driver = "EMU10K1", .name = "E-mu APS [4001]", 
+	 .driver = "EMU10K1", .name = "E-mu APS [PC545]",
 	 .id = "APS",
 	 .emu10k1_chip = 1,
 	 .ecard = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102,
-	 .driver = "EMU10K1", .name = "SBLive! [CT4620]", 
+	 .driver = "EMU10K1", .name = "SB Live! [CT4620]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", 
+	 .driver = "EMU10K1", .name = "SB Live! Value [CT4670]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
 	{.vendor = 0x1102, .device = 0x0002,
-	 .driver = "EMU10K1", .name = "SB Live [Unknown]", 
+	 .driver = "EMU10K1", .name = "SB Live! [Unknown]",
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
@@ -1702,13 +1709,13 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 };
 
 int __devinit snd_emu10k1_create(struct snd_card *card,
-		       struct pci_dev * pci,
+		       struct pci_dev *pci,
 		       unsigned short extin_mask,
 		       unsigned short extout_mask,
 		       long max_cache_bytes,
 		       int enable_ir,
 		       uint subsystem,
-		       struct snd_emu10k1 ** remu)
+		       struct snd_emu10k1 **remu)
 {
 	struct snd_emu10k1 *emu;
 	int idx, err;
@@ -1718,11 +1725,12 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	static struct snd_device_ops ops = {
 		.dev_free =	snd_emu10k1_dev_free,
 	};
-	
+
 	*remu = NULL;
 
 	/* enable PCI device */
-	if ((err = pci_enable_device(pci)) < 0)
+	err = pci_enable_device(pci);
+	if (err < 0)
 		return err;
 
 	emu = kzalloc(sizeof(*emu), GFP_KERNEL);
@@ -1749,16 +1757,17 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	emu->revision = pci->revision;
 	pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial);
 	pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model);
-	snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model);
+	snd_printdd("vendor = 0x%x, device = 0x%x, subsystem_vendor_id = 0x%x, subsystem_id = 0x%x\n", pci->vendor, pci->device, emu->serial, emu->model);
 
 	for (c = emu_chip_details; c->vendor; c++) {
 		if (c->vendor == pci->vendor && c->device == pci->device) {
 			if (subsystem) {
-				if (c->subsystem && (c->subsystem == subsystem) ) {
+				if (c->subsystem && (c->subsystem == subsystem))
 					break;
-				} else continue;
+				else
+					continue;
 			} else {
-				if (c->subsystem && (c->subsystem != emu->serial) )
+				if (c->subsystem && (c->subsystem != emu->serial))
 					continue;
 				if (c->revision && c->revision != emu->revision)
 					continue;
@@ -1774,14 +1783,18 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	}
 	emu->card_capabilities = c;
 	if (c->subsystem && !subsystem)
-		snd_printdd("Sound card name=%s\n", c->name);
-	else if (subsystem) 
-		snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n",
-		       	c->name, pci->vendor, pci->device, emu->serial, c->subsystem);
-	else 
-		snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n",
-		      	c->name, pci->vendor, pci->device, emu->serial);
-	
+		snd_printdd("Sound card name = %s\n", c->name);
+	else if (subsystem)
+		snd_printdd("Sound card name = %s, "
+			"vendor = 0x%x, device = 0x%x, subsystem = 0x%x. "
+			"Forced to subsytem = 0x%x\n",	c->name,
+			pci->vendor, pci->device, emu->serial, c->subsystem);
+	else
+		snd_printdd("Sound card name = %s, "
+			"vendor = 0x%x, device = 0x%x, subsystem = 0x%x.\n",
+			c->name, pci->vendor, pci->device,
+			emu->serial);
+
 	if (!*card->id && c->id) {
 		int i, n = 0;
 		strlcpy(card->id, c->id, sizeof(card->id));
@@ -1815,7 +1828,8 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	else
 		emu->gpr_base = FXGPREGBASE;
 
-	if ((err = pci_request_regions(pci, "EMU10K1")) < 0) {
+	err = pci_request_regions(pci, "EMU10K1");
+	if (err < 0) {
 		kfree(emu);
 		pci_disable_device(pci);
 		return err;
@@ -1862,21 +1876,25 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	emu->enable_ir = enable_ir;
 
 	if (emu->card_capabilities->ca_cardbus_chip) {
-		if ((err = snd_emu10k1_cardbus_init(emu)) < 0)
+		err = snd_emu10k1_cardbus_init(emu);
+		if (err < 0)
 			goto error;
 	}
 	if (emu->card_capabilities->ecard) {
-		if ((err = snd_emu10k1_ecard_init(emu)) < 0)
+		err = snd_emu10k1_ecard_init(emu);
+		if (err < 0)
 			goto error;
 	} else if (emu->card_capabilities->emu_model) {
- 		if ((err = snd_emu10k1_emu1010_init(emu)) < 0) {
- 			snd_emu10k1_free(emu);
- 			return err;
- 		}
+		err = snd_emu10k1_emu1010_init(emu);
+		if (err < 0) {
+			snd_emu10k1_free(emu);
+			return err;
+		}
 	} else {
 		/* 5.1: Enable the additional AC97 Slots. If the emu10k1 version
 			does not support this, it shouldn't do any harm */
-		snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE);
+		snd_emu10k1_ptr_write(emu, AC97SLOT, 0,
+					AC97SLOT_CNTR|AC97SLOT_LFE);
 	}
 
 	/* initialize TRAM setup */
@@ -1916,7 +1934,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 		snd_emu10k1_synth_alloc(emu, 4096);
 	if (emu->reserved_page)
 		emu->reserved_page->map_locked = 1;
-	
+
 	/* Clear silent pages and set up pointers */
 	memset(emu->silent_page.area, 0, PAGE_SIZE);
 	silent_page = emu->silent_page.addr << 1;
@@ -1929,19 +1947,23 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 		emu->voices[idx].number = idx;
 	}
 
-	if ((err = snd_emu10k1_init(emu, enable_ir, 0)) < 0)
+	err = snd_emu10k1_init(emu, enable_ir, 0);
+	if (err < 0)
 		goto error;
 #ifdef CONFIG_PM
-	if ((err = alloc_pm_buffer(emu)) < 0)
+	err = alloc_pm_buffer(emu);
+	if (err < 0)
 		goto error;
 #endif
 
 	/*  Initialize the effect engine */
-	if ((err = snd_emu10k1_init_efx(emu)) < 0)
+	err = snd_emu10k1_init_efx(emu);
+	if (err < 0)
 		goto error;
 	snd_emu10k1_audio_enable(emu);
 
-	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops)) < 0)
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops);
+	if (err < 0)
 		goto error;
 
 #ifdef CONFIG_PROC_FS
@@ -1981,7 +2003,7 @@ static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu)
 	if (emu->audigy)
 		size += ARRAY_SIZE(saved_regs_audigy);
 	emu->saved_ptr = vmalloc(4 * NUM_G * size);
-	if (! emu->saved_ptr)
+	if (!emu->saved_ptr)
 		return -ENOMEM;
 	if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0)
 		return -ENOMEM;
@@ -2026,7 +2048,7 @@ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu)
 	if (emu->card_capabilities->ecard)
 		snd_emu10k1_ecard_init(emu);
 	else if (emu->card_capabilities->emu_model)
- 		snd_emu10k1_emu1010_init(emu);
+		snd_emu10k1_emu1010_init(emu);
 	else
 		snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE);
 	snd_emu10k1_init(emu, emu->enable_ir, 1);
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index f34bbfb705f5..b0fb6c917c38 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -1639,6 +1639,45 @@ static struct snd_kcontrol_new snd_audigy_shared_spdif __devinitdata =
 	.put =		snd_emu10k1_shared_spdif_put
 };
 
+/* workaround for too low volume on Audigy due to 16bit/24bit conversion */
+
+#define snd_audigy_capture_boost_info	snd_ctl_boolean_mono_info
+
+static int snd_audigy_capture_boost_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int val;
+
+	/* FIXME: better to use a cached version */
+	val = snd_ac97_read(emu->ac97, AC97_REC_GAIN);
+	ucontrol->value.integer.value[0] = !!val;
+	return 0;
+}
+
+static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int val;
+
+	if (ucontrol->value.integer.value[0])
+		val = 0x0f0f;
+	else
+		val = 0;
+	return snd_ac97_update(emu->ac97, AC97_REC_GAIN, val);
+}
+
+static struct snd_kcontrol_new snd_audigy_capture_boost __devinitdata =
+{
+	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name =		"Analog Capture Boost",
+	.info =		snd_audigy_capture_boost_info,
+	.get =		snd_audigy_capture_boost_get,
+	.put =		snd_audigy_capture_boost_put
+};
+
+
 /*
  */
 static void snd_emu10k1_mixer_free_ac97(struct snd_ac97 *ac97)
@@ -2087,5 +2126,12 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 		}
 	}
 		
+	if (emu->card_capabilities->ac97_chip && emu->audigy) {
+		err = snd_ctl_add(card, snd_ctl_new1(&snd_audigy_capture_boost,
+						     emu));
+		if (err < 0)
+			return err;
+	}
+
 	return 0;
 }
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 20ee7599600b..e9c3794bbcb8 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -1953,7 +1953,7 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
 	outw(inw(chip->io_port + 4) & 1, chip->io_port + 4);
 
 	if (event & ESM_HWVOL_IRQ)
-		tasklet_hi_schedule(&chip->hwvol_tq); /* we'll do this later */
+		tasklet_schedule(&chip->hwvol_tq); /* we'll do this later */
 
 	/* else ack 'em all, i imagine */
 	outb(0xFF, chip->io_port + 0x1A);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
new file mode 100644
index 000000000000..eb2a19b894a0
--- /dev/null
+++ b/sound/pci/hda/Kconfig
@@ -0,0 +1,188 @@
+menuconfig SND_HDA_INTEL
+	tristate "Intel HD Audio"
+	select SND_PCM
+	select SND_VMASTER
+	select SND_JACK if INPUT=y || INPUT=SND
+	help
+	  Say Y here to include support for Intel "High Definition
+	  Audio" (Azalia) and its compatible devices.
+
+	  This option enables the HD-audio controller.  Don't forget
+	  to choose the appropriate codec options below.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-hda-intel.
+
+if SND_HDA_INTEL
+
+config SND_HDA_HWDEP
+	bool "Build hwdep interface for HD-audio driver"
+	select SND_HWDEP
+	help
+	  Say Y here to build a hwdep interface for HD-audio driver.
+	  This interface can be used for out-of-band communication
+	  with codecs for debugging purposes.
+
+config SND_HDA_RECONFIG
+	bool "Allow dynamic codec reconfiguration (EXPERIMENTAL)"
+	depends on SND_HDA_HWDEP && EXPERIMENTAL
+	help
+	  Say Y here to enable the HD-audio codec re-configuration feature.
+	  This adds the sysfs interfaces to allow user to clear the whole
+	  codec configuration, change the codec setup, add extra verbs,
+	  and re-configure the codec dynamically.
+
+config SND_HDA_INPUT_BEEP
+	bool "Support digital beep via input layer"
+	depends on INPUT=y || INPUT=SND_HDA_INTEL
+	help
+	  Say Y here to build a digital beep interface for HD-audio
+	  driver. This interface is used to generate digital beeps.
+
+config SND_HDA_CODEC_REALTEK
+	bool "Build Realtek HD-audio codec support"
+	default y
+	help
+	  Say Y here to include Realtek HD-audio codec support in
+	  snd-hda-intel driver, such as ALC880.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-realtek.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_ANALOG
+	bool "Build Analog Device HD-audio codec support"
+	default y
+	help
+	  Say Y here to include Analog Device HD-audio codec support in
+	  snd-hda-intel driver, such as AD1986A.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-analog.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_SIGMATEL
+	bool "Build IDT/Sigmatel HD-audio codec support"
+	default y
+	help
+	  Say Y here to include IDT (Sigmatel) HD-audio codec support in
+	  snd-hda-intel driver, such as STAC9200.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-idt.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_VIA
+	bool "Build VIA HD-audio codec support"
+	default y
+	help
+	  Say Y here to include VIA HD-audio codec support in
+	  snd-hda-intel driver, such as VT1708.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-via.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_ATIHDMI
+	bool "Build ATI HDMI HD-audio codec support"
+	default y
+	help
+	  Say Y here to include ATI HDMI HD-audio codec support in
+	  snd-hda-intel driver, such as ATI RS600 HDMI.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-atihdmi.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_NVHDMI
+	bool "Build NVIDIA HDMI HD-audio codec support"
+	default y
+	help
+	  Say Y here to include NVIDIA HDMI HD-audio codec support in
+	  snd-hda-intel driver, such as NVIDIA MCP78 HDMI.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-nvhdmi.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_INTELHDMI
+	bool "Build INTEL HDMI HD-audio codec support"
+	default y
+	help
+	  Say Y here to include INTEL HDMI HD-audio codec support in
+	  snd-hda-intel driver, such as Eaglelake integrated HDMI.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-intelhdmi.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_ELD
+	def_bool y
+	depends on SND_HDA_CODEC_INTELHDMI
+
+config SND_HDA_CODEC_CONEXANT
+	bool "Build Conexant HD-audio codec support"
+	default y
+	help
+	  Say Y here to include Conexant HD-audio codec support in
+	  snd-hda-intel driver, such as CX20549.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-conexant.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_CMEDIA
+	bool "Build C-Media HD-audio codec support"
+	default y
+	help
+	  Say Y here to include C-Media HD-audio codec support in
+	  snd-hda-intel driver, such as CMI9880.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-cmedia.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_CODEC_SI3054
+	bool "Build Silicon Labs 3054 HD-modem codec support"
+	default y
+	help
+	  Say Y here to include Silicon Labs 3054 HD-modem codec
+	  (and compatibles) support in snd-hda-intel driver.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-si3054.
+	  This module is automatically loaded at probing.
+
+config SND_HDA_GENERIC
+	bool "Enable generic HD-audio codec parser"
+	default y
+	help
+	  Say Y here to enable the generic HD-audio codec parser
+	  in snd-hda-intel driver.
+
+config SND_HDA_POWER_SAVE
+	bool "Aggressive power-saving on HD-audio"
+	help
+	  Say Y here to enable more aggressive power-saving mode on
+	  HD-audio driver.  The power-saving timeout can be configured
+	  via power_save option or over sysfs on-the-fly.
+
+config SND_HDA_POWER_SAVE_DEFAULT
+	int "Default time-out for HD-audio power-save mode"
+	depends on SND_HDA_POWER_SAVE
+	default 0
+	help
+	  The default time-out value in seconds for HD-audio automatic
+	  power-save mode.  0 means to disable the power-save mode.
+
+endif
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 1980c6d207e7..50f9d0967251 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,20 +1,59 @@
-snd-hda-intel-y := hda_intel.o
-# since snd-hda-intel is the only driver using hda-codec,
-# merge it into a single module although it was originally
-# designed to be individual modules
-snd-hda-intel-y += hda_codec.o
-snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
-snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
-snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
-snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o
-snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o
+snd-hda-intel-objs := hda_intel.o
 
+snd-hda-codec-y := hda_codec.o
+snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
+snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
+# snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o
+snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
+snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
+
+snd-hda-codec-realtek-objs :=	patch_realtek.o
+snd-hda-codec-cmedia-objs :=	patch_cmedia.o
+snd-hda-codec-analog-objs :=	patch_analog.o
+snd-hda-codec-idt-objs :=	patch_sigmatel.o
+snd-hda-codec-si3054-objs :=	patch_si3054.o
+snd-hda-codec-atihdmi-objs :=	patch_atihdmi.o
+snd-hda-codec-conexant-objs :=	patch_conexant.o
+snd-hda-codec-via-objs :=	patch_via.o
+snd-hda-codec-nvhdmi-objs :=	patch_nvhdmi.o
+snd-hda-codec-intelhdmi-objs :=	patch_intelhdmi.o hda_eld.o
+
+# common driver
+obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o
+
+# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans)
+ifdef CONFIG_SND_HDA_CODEC_REALTEK
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_CMEDIA
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_ANALOG
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_SI3054
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_CONEXANT
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_VIA
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_NVHDMI
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-nvhdmi.o
+endif
+ifdef CONFIG_SND_HDA_CODEC_INTELHDMI
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-intelhdmi.o
+endif
+
+# this must be the last entry after codec drivers;
+# otherwise the codec patches won't be hooked before the PCI probe
+# when built in kernel
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 3ecd7e797dee..e00421c0d8ba 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -128,6 +128,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device);
 
 void snd_hda_detach_beep_device(struct hda_codec *codec)
 {
@@ -140,3 +141,4 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 		kfree(beep);
 	}
 }
+EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index eb9164176dab..e16cf63821ae 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -31,15 +31,6 @@
 #include <sound/initval.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
-#include "hda_patch.h"	/* codec presets */
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-/* define this option here to hide as static */
-static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
-module_param(power_save, int, 0644);
-MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
-		 "(in second, 0 = disable).");
-#endif
 
 /*
  * vendor / preset table
@@ -55,6 +46,7 @@ static struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x1002, "ATI" },
 	{ 0x1057, "Motorola" },
 	{ 0x1095, "Silicon Image" },
+	{ 0x10de, "Nvidia" },
 	{ 0x10ec, "Realtek" },
 	{ 0x1106, "VIA" },
 	{ 0x111d, "IDT" },
@@ -66,40 +58,31 @@ static struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x1854, "LG" },
 	{ 0x1aec, "Wolfson Microelectronics" },
 	{ 0x434d, "C-Media" },
+	{ 0x8086, "Intel" },
 	{ 0x8384, "SigmaTel" },
 	{} /* terminator */
 };
 
-static const struct hda_codec_preset *hda_preset_tables[] = {
-#ifdef CONFIG_SND_HDA_CODEC_REALTEK
-	snd_hda_preset_realtek,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_CMEDIA
-	snd_hda_preset_cmedia,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_ANALOG
-	snd_hda_preset_analog,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
-	snd_hda_preset_sigmatel,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_SI3054
-	snd_hda_preset_si3054,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
-	snd_hda_preset_atihdmi,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_CONEXANT
-	snd_hda_preset_conexant,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_VIA
-	snd_hda_preset_via,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_NVHDMI
-	snd_hda_preset_nvhdmi,
-#endif
-	NULL
-};
+static DEFINE_MUTEX(preset_mutex);
+static LIST_HEAD(hda_preset_tables);
+
+int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset)
+{
+	mutex_lock(&preset_mutex);
+	list_add_tail(&preset->list, &hda_preset_tables);
+	mutex_unlock(&preset_mutex);
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_add_codec_preset);
+
+int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset)
+{
+	mutex_lock(&preset_mutex);
+	list_del(&preset->list);
+	mutex_unlock(&preset_mutex);
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void hda_power_work(struct work_struct *work);
@@ -108,6 +91,72 @@ static void hda_keep_power_on(struct hda_codec *codec);
 static inline void hda_keep_power_on(struct hda_codec *codec) {}
 #endif
 
+const char *snd_hda_get_jack_location(u32 cfg)
+{
+	static char *bases[7] = {
+		"N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom",
+	};
+	static unsigned char specials_idx[] = {
+		0x07, 0x08,
+		0x17, 0x18, 0x19,
+		0x37, 0x38
+	};
+	static char *specials[] = {
+		"Rear Panel", "Drive Bar",
+		"Riser", "HDMI", "ATAPI",
+		"Mobile-In", "Mobile-Out"
+	};
+	int i;
+	cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT;
+	if ((cfg & 0x0f) < 7)
+		return bases[cfg & 0x0f];
+	for (i = 0; i < ARRAY_SIZE(specials_idx); i++) {
+		if (cfg == specials_idx[i])
+			return specials[i];
+	}
+	return "UNKNOWN";
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_jack_location);
+
+const char *snd_hda_get_jack_connectivity(u32 cfg)
+{
+	static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" };
+
+	return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3];
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_jack_connectivity);
+
+const char *snd_hda_get_jack_type(u32 cfg)
+{
+	static char *jack_types[16] = {
+		"Line Out", "Speaker", "HP Out", "CD",
+		"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
+		"Line In", "Aux", "Mic", "Telephony",
+		"SPDIF In", "Digitial In", "Reserved", "Other"
+	};
+
+	return jack_types[(cfg & AC_DEFCFG_DEVICE)
+				>> AC_DEFCFG_DEVICE_SHIFT];
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_jack_type);
+
+/*
+ * Compose a 32bit command word to be sent to the HD-audio controller
+ */
+static inline unsigned int
+make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
+	       unsigned int verb, unsigned int parm)
+{
+	u32 val;
+
+	val = (u32)(codec->addr & 0x0f) << 28;
+	val |= (u32)direct << 27;
+	val |= (u32)nid << 20;
+	val |= verb << 8;
+	val |= parm;
+	return val;
+}
+
 /**
  * snd_hda_codec_read - send a command and get the response
  * @codec: the HDA codec
@@ -124,17 +173,21 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
 				int direct,
 				unsigned int verb, unsigned int parm)
 {
+	struct hda_bus *bus = codec->bus;
 	unsigned int res;
+
+	res = make_codec_cmd(codec, nid, direct, verb, parm);
 	snd_hda_power_up(codec);
-	mutex_lock(&codec->bus->cmd_mutex);
-	if (!codec->bus->ops.command(codec, nid, direct, verb, parm))
-		res = codec->bus->ops.get_response(codec);
+	mutex_lock(&bus->cmd_mutex);
+	if (!bus->ops.command(bus, res))
+		res = bus->ops.get_response(bus);
 	else
 		res = (unsigned int)-1;
-	mutex_unlock(&codec->bus->cmd_mutex);
+	mutex_unlock(&bus->cmd_mutex);
 	snd_hda_power_down(codec);
 	return res;
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_read);
 
 /**
  * snd_hda_codec_write - send a single command without waiting for response
@@ -151,14 +204,19 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
 int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
 			 unsigned int verb, unsigned int parm)
 {
+	struct hda_bus *bus = codec->bus;
+	unsigned int res;
 	int err;
+
+	res = make_codec_cmd(codec, nid, direct, verb, parm);
 	snd_hda_power_up(codec);
-	mutex_lock(&codec->bus->cmd_mutex);
-	err = codec->bus->ops.command(codec, nid, direct, verb, parm);
-	mutex_unlock(&codec->bus->cmd_mutex);
+	mutex_lock(&bus->cmd_mutex);
+	err = bus->ops.command(bus, res);
+	mutex_unlock(&bus->cmd_mutex);
 	snd_hda_power_down(codec);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_write);
 
 /**
  * snd_hda_sequence_write - sequence writes
@@ -173,6 +231,7 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq)
 	for (; seq->nid; seq++)
 		snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param);
 }
+EXPORT_SYMBOL_HDA(snd_hda_sequence_write);
 
 /**
  * snd_hda_get_sub_nodes - get the range of sub nodes
@@ -194,6 +253,7 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
 	*start_id = (parm >> 16) & 0x7fff;
 	return (int)(parm & 0x7fff);
 }
+EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
 
 /**
  * snd_hda_get_connections - get connection list
@@ -282,6 +342,7 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 	}
 	return conns;
 }
+EXPORT_SYMBOL_HDA(snd_hda_get_connections);
 
 
 /**
@@ -316,6 +377,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
 
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_queue_unsol_event);
 
 /*
  * process queued unsolicited events
@@ -345,7 +407,7 @@ static void process_unsol_events(struct work_struct *work)
 /*
  * initialize unsolicited queue
  */
-static int __devinit init_unsol_queue(struct hda_bus *bus)
+static int init_unsol_queue(struct hda_bus *bus)
 {
 	struct hda_bus_unsolicited *unsol;
 
@@ -391,9 +453,24 @@ static int snd_hda_bus_free(struct hda_bus *bus)
 static int snd_hda_bus_dev_free(struct snd_device *device)
 {
 	struct hda_bus *bus = device->device_data;
+	bus->shutdown = 1;
 	return snd_hda_bus_free(bus);
 }
 
+#ifdef CONFIG_SND_HDA_HWDEP
+static int snd_hda_bus_dev_register(struct snd_device *device)
+{
+	struct hda_bus *bus = device->device_data;
+	struct hda_codec *codec;
+	list_for_each_entry(codec, &bus->codec_list, list) {
+		snd_hda_hwdep_add_sysfs(codec);
+	}
+	return 0;
+}
+#else
+#define snd_hda_bus_dev_register	NULL
+#endif
+
 /**
  * snd_hda_bus_new - create a HDA bus
  * @card: the card entry
@@ -402,13 +479,14 @@ static int snd_hda_bus_dev_free(struct snd_device *device)
  *
  * Returns 0 if successful, or a negative error code.
  */
-int __devinit snd_hda_bus_new(struct snd_card *card,
+int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
 			      const struct hda_bus_template *temp,
 			      struct hda_bus **busp)
 {
 	struct hda_bus *bus;
 	int err;
 	static struct snd_device_ops dev_ops = {
+		.dev_register = snd_hda_bus_dev_register,
 		.dev_free = snd_hda_bus_dev_free,
 	};
 
@@ -430,6 +508,7 @@ int __devinit snd_hda_bus_new(struct snd_card *card,
 	bus->private_data = temp->private_data;
 	bus->pci = temp->pci;
 	bus->modelname = temp->modelname;
+	bus->power_save = temp->power_save;
 	bus->ops = temp->ops;
 
 	mutex_init(&bus->cmd_mutex);
@@ -444,27 +523,42 @@ int __devinit snd_hda_bus_new(struct snd_card *card,
 		*busp = bus;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_bus_new);
 
 #ifdef CONFIG_SND_HDA_GENERIC
 #define is_generic_config(codec) \
-	(codec->bus->modelname && !strcmp(codec->bus->modelname, "generic"))
+	(codec->modelname && !strcmp(codec->modelname, "generic"))
 #else
 #define is_generic_config(codec)	0
 #endif
 
+#ifdef MODULE
+#define HDA_MODREQ_MAX_COUNT	2	/* two request_modules()'s */
+#else
+#define HDA_MODREQ_MAX_COUNT	0	/* all presets are statically linked */
+#endif
+
 /*
  * find a matching codec preset
  */
-static const struct hda_codec_preset __devinit *
+static const struct hda_codec_preset *
 find_codec_preset(struct hda_codec *codec)
 {
-	const struct hda_codec_preset **tbl, *preset;
+	struct hda_codec_preset_list *tbl;
+	const struct hda_codec_preset *preset;
+	int mod_requested = 0;
 
 	if (is_generic_config(codec))
 		return NULL; /* use the generic parser */
 
-	for (tbl = hda_preset_tables; *tbl; tbl++) {
-		for (preset = *tbl; preset->id; preset++) {
+ again:
+	mutex_lock(&preset_mutex);
+	list_for_each_entry(tbl, &hda_preset_tables, list) {
+		if (!try_module_get(tbl->owner)) {
+			snd_printk(KERN_ERR "hda_codec: cannot module_get\n");
+			continue;
+		}
+		for (preset = tbl->preset; preset->id; preset++) {
 			u32 mask = preset->mask;
 			if (preset->afg && preset->afg != codec->afg)
 				continue;
@@ -474,23 +568,40 @@ find_codec_preset(struct hda_codec *codec)
 				mask = ~0;
 			if (preset->id == (codec->vendor_id & mask) &&
 			    (!preset->rev ||
-			     preset->rev == codec->revision_id))
+			     preset->rev == codec->revision_id)) {
+				mutex_unlock(&preset_mutex);
+				codec->owner = tbl->owner;
 				return preset;
+			}
 		}
+		module_put(tbl->owner);
+	}
+	mutex_unlock(&preset_mutex);
+
+	if (mod_requested < HDA_MODREQ_MAX_COUNT) {
+		char name[32];
+		if (!mod_requested)
+			snprintf(name, sizeof(name), "snd-hda-codec-id:%08x",
+				 codec->vendor_id);
+		else
+			snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*",
+				 (codec->vendor_id >> 16) & 0xffff);
+		request_module(name);
+		mod_requested++;
+		goto again;
 	}
 	return NULL;
 }
 
 /*
- * snd_hda_get_codec_name - store the codec name
+ * get_codec_name - store the codec name
  */
-void snd_hda_get_codec_name(struct hda_codec *codec,
-			    char *name, int namelen)
+static int get_codec_name(struct hda_codec *codec)
 {
 	const struct hda_vendor_id *c;
 	const char *vendor = NULL;
 	u16 vendor_id = codec->vendor_id >> 16;
-	char tmp[16];
+	char tmp[16], name[32];
 
 	for (c = hda_vendor_ids; c->id; c++) {
 		if (c->id == vendor_id) {
@@ -503,16 +614,21 @@ void snd_hda_get_codec_name(struct hda_codec *codec,
 		vendor = tmp;
 	}
 	if (codec->preset && codec->preset->name)
-		snprintf(name, namelen, "%s %s", vendor, codec->preset->name);
+		snprintf(name, sizeof(name), "%s %s", vendor,
+			 codec->preset->name);
 	else
-		snprintf(name, namelen, "%s ID %x", vendor,
+		snprintf(name, sizeof(name), "%s ID %x", vendor,
 			 codec->vendor_id & 0xffff);
+	codec->name = kstrdup(name, GFP_KERNEL);
+	if (!codec->name)
+		return -ENOMEM;
+	return 0;
 }
 
 /*
  * look for an AFG and MFG nodes
  */
-static void __devinit setup_fg_nodes(struct hda_codec *codec)
+static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)
 {
 	int i, total_nodes;
 	hda_nid_t nid;
@@ -571,11 +687,15 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	flush_scheduled_work();
 #endif
 	list_del(&codec->list);
+	snd_array_free(&codec->mixers);
 	codec->bus->caddr_tbl[codec->addr] = NULL;
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
+	module_put(codec->owner);
 	free_hda_cache(&codec->amp_cache);
 	free_hda_cache(&codec->cmd_cache);
+	kfree(codec->name);
+	kfree(codec->modelname);
 	kfree(codec->wcaps);
 	kfree(codec);
 }
@@ -588,8 +708,8 @@ static void snd_hda_codec_free(struct hda_codec *codec)
  *
  * Returns 0 if successful, or a negative error code.
  */
-int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
-				struct hda_codec **codecp)
+int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+				    int do_init, struct hda_codec **codecp)
 {
 	struct hda_codec *codec;
 	char component[31];
@@ -617,6 +737,14 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 	mutex_init(&codec->spdif_mutex);
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+	snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32);
+	if (codec->bus->modelname) {
+		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
+		if (!codec->modelname) {
+			snd_hda_codec_free(codec);
+			return -ENODEV;
+		}
+	}
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
@@ -662,12 +790,44 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 			snd_hda_codec_read(codec, nid, 0,
 					   AC_VERB_GET_SUBSYSTEM_ID, 0);
 	}
+	if (bus->modelname)
+		codec->modelname = kstrdup(bus->modelname, GFP_KERNEL);
+
+	if (do_init) {
+		err = snd_hda_codec_configure(codec);
+		if (err < 0) {
+			snd_hda_codec_free(codec);
+			return err;
+		}
+	}
+	snd_hda_codec_proc_new(codec);
+
+	snd_hda_create_hwdep(codec);
+
+	sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id,
+		codec->subsystem_id, codec->revision_id);
+	snd_component_add(codec->bus->card, component);
+
+	if (codecp)
+		*codecp = codec;
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_codec_new);
+
+int snd_hda_codec_configure(struct hda_codec *codec)
+{
+	int err;
 
 	codec->preset = find_codec_preset(codec);
+	if (!codec->name) {
+		err = get_codec_name(codec);
+		if (err < 0)
+			return err;
+	}
 	/* audio codec should override the mixer name */
-	if (codec->afg || !*bus->card->mixername)
-		snd_hda_get_codec_name(codec, bus->card->mixername,
-				       sizeof(bus->card->mixername));
+	if (codec->afg || !*codec->bus->card->mixername)
+		strlcpy(codec->bus->card->mixername, codec->name,
+			sizeof(codec->bus->card->mixername));
 
 	if (is_generic_config(codec)) {
 		err = snd_hda_parse_generic_codec(codec);
@@ -684,25 +844,9 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 		printk(KERN_ERR "hda-codec: No codec parser is available\n");
 
  patched:
-	if (err < 0) {
-		snd_hda_codec_free(codec);
-		return err;
-	}
-
-	if (codec->patch_ops.unsol_event)
-		init_unsol_queue(bus);
-
-	snd_hda_codec_proc_new(codec);
-#ifdef CONFIG_SND_HDA_HWDEP
-	snd_hda_create_hwdep(codec);
-#endif
-
-	sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id);
-	snd_component_add(codec->bus->card, component);
-
-	if (codecp)
-		*codecp = codec;
-	return 0;
+	if (!err && codec->patch_ops.unsol_event)
+		err = init_unsol_queue(codec->bus);
+	return err;
 }
 
 /**
@@ -728,6 +872,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 	msleep(1);
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format);
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream);
 
 void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
 {
@@ -741,6 +886,7 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
 #endif
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream);
 
 /*
  * amp access functions
@@ -752,17 +898,17 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
 #define INFO_AMP_VOL(ch)	(1 << (1 + (ch)))
 
 /* initialize the hash table */
-static void __devinit init_hda_cache(struct hda_cache_rec *cache,
+static void /*__devinit*/ init_hda_cache(struct hda_cache_rec *cache,
 				     unsigned int record_size)
 {
 	memset(cache, 0, sizeof(*cache));
 	memset(cache->hash, 0xff, sizeof(cache->hash));
-	cache->record_size = record_size;
+	snd_array_init(&cache->buf, record_size, 64);
 }
 
 static void free_hda_cache(struct hda_cache_rec *cache)
 {
-	kfree(cache->buffer);
+	snd_array_free(&cache->buf);
 }
 
 /* query the hash.  allocate an entry if not found. */
@@ -774,35 +920,17 @@ static struct hda_cache_head  *get_alloc_hash(struct hda_cache_rec *cache,
 	struct hda_cache_head *info;
 
 	while (cur != 0xffff) {
-		info = (struct hda_cache_head *)(cache->buffer +
-						 cur * cache->record_size);
+		info = snd_array_elem(&cache->buf, cur);
 		if (info->key == key)
 			return info;
 		cur = info->next;
 	}
 
 	/* add a new hash entry */
-	if (cache->num_entries >= cache->size) {
-		/* reallocate the array */
-		unsigned int new_size = cache->size + 64;
-		void *new_buffer;
-		new_buffer = kcalloc(new_size, cache->record_size, GFP_KERNEL);
-		if (!new_buffer) {
-			snd_printk(KERN_ERR "hda_codec: "
-				   "can't malloc amp_info\n");
-			return NULL;
-		}
-		if (cache->buffer) {
-			memcpy(new_buffer, cache->buffer,
-			       cache->size * cache->record_size);
-			kfree(cache->buffer);
-		}
-		cache->size = new_size;
-		cache->buffer = new_buffer;
-	}
-	cur = cache->num_entries++;
-	info = (struct hda_cache_head *)(cache->buffer +
-					 cur * cache->record_size);
+	info = snd_array_new(&cache->buf);
+	if (!info)
+		return NULL;
+	cur = snd_array_index(&cache->buf, info);
 	info->key = key;
 	info->val = 0;
 	info->next = cache->hash[idx];
@@ -840,6 +968,7 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
 	}
 	return info->amp_caps;
 }
+EXPORT_SYMBOL_HDA(query_amp_caps);
 
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps)
@@ -853,6 +982,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 	info->head.val |= INFO_AMP_CAPS;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
 
 /*
  * read the current volume to info
@@ -906,6 +1036,7 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
 		return 0;
 	return get_vol_mute(codec, info, nid, ch, direction, index);
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read);
 
 /*
  * update the AMP value, mask = bit mask to set, val = the value
@@ -925,6 +1056,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 	put_vol_mute(codec, info, nid, ch, direction, idx, val);
 	return 1;
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_amp_update);
 
 /*
  * update the AMP stereo with the same mask and value
@@ -938,15 +1070,16 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 						idx, mask, val);
 	return ret;
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo);
 
 #ifdef SND_HDA_NEEDS_RESUME
 /* resume the all amp commands from the cache */
 void snd_hda_codec_resume_amp(struct hda_codec *codec)
 {
-	struct hda_amp_info *buffer = codec->amp_cache.buffer;
+	struct hda_amp_info *buffer = codec->amp_cache.buf.list;
 	int i;
 
-	for (i = 0; i < codec->amp_cache.size; i++, buffer++) {
+	for (i = 0; i < codec->amp_cache.buf.used; i++, buffer++) {
 		u32 key = buffer->head.key;
 		hda_nid_t nid;
 		unsigned int idx, dir, ch;
@@ -963,6 +1096,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
 		}
 	}
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp);
 #endif /* SND_HDA_NEEDS_RESUME */
 
 /* volume */
@@ -990,6 +1124,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.integer.max = caps;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info);
 
 int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
@@ -1009,6 +1144,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 			& HDA_AMP_VOLMASK;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get);
 
 int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
@@ -1033,6 +1169,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 	snd_hda_power_down(codec);
 	return change;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_put);
 
 int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 			  unsigned int size, unsigned int __user *_tlv)
@@ -1059,6 +1196,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 		return -EFAULT;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_tlv);
 
 /*
  * set (static) TLV for virtual master volume; recalculated as max 0dB
@@ -1078,6 +1216,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
 	tlv[2] = -nums * step;
 	tlv[3] = step;
 }
+EXPORT_SYMBOL_HDA(snd_hda_set_vmaster_tlv);
 
 /* find a mixer control element with the given name */
 static struct snd_kcontrol *
@@ -1097,6 +1236,69 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 {
 	return _snd_hda_find_mixer_ctl(codec, name, 0);
 }
+EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl);
+
+/* Add a control element and assign to the codec */
+int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl)
+{
+	int err;
+	struct snd_kcontrol **knewp;
+
+	err = snd_ctl_add(codec->bus->card, kctl);
+	if (err < 0)
+		return err;
+	knewp = snd_array_new(&codec->mixers);
+	if (!knewp)
+		return -ENOMEM;
+	*knewp = kctl;
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_ctl_add);
+
+#ifdef CONFIG_SND_HDA_RECONFIG
+/* Clear all controls assigned to the given codec */
+void snd_hda_ctls_clear(struct hda_codec *codec)
+{
+	int i;
+	struct snd_kcontrol **kctls = codec->mixers.list;
+	for (i = 0; i < codec->mixers.used; i++)
+		snd_ctl_remove(codec->bus->card, kctls[i]);
+	snd_array_free(&codec->mixers);
+}
+
+void snd_hda_codec_reset(struct hda_codec *codec)
+{
+	int i;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	cancel_delayed_work(&codec->power_work);
+	flush_scheduled_work();
+#endif
+	snd_hda_ctls_clear(codec);
+	/* relase PCMs */
+	for (i = 0; i < codec->num_pcms; i++) {
+		if (codec->pcm_info[i].pcm) {
+			snd_device_free(codec->bus->card,
+					codec->pcm_info[i].pcm);
+			clear_bit(codec->pcm_info[i].device,
+				  codec->bus->pcm_dev_bits);
+		}
+	}
+	if (codec->patch_ops.free)
+		codec->patch_ops.free(codec);
+	codec->proc_widget_hook = NULL;
+	codec->spec = NULL;
+	free_hda_cache(&codec->amp_cache);
+	free_hda_cache(&codec->cmd_cache);
+	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
+	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+	codec->num_pcms = 0;
+	codec->pcm_info = NULL;
+	codec->preset = NULL;
+	module_put(codec->owner);
+	codec->owner = NULL;
+}
+#endif /* CONFIG_SND_HDA_RECONFIG */
 
 /* create a virtual master control and add slaves */
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
@@ -1115,7 +1317,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	kctl = snd_ctl_make_virtual_master(name, tlv);
 	if (!kctl)
 		return -ENOMEM;
-	err = snd_ctl_add(codec->bus->card, kctl);
+	err = snd_hda_ctl_add(codec, kctl);
 	if (err < 0)
 		return err;
 	
@@ -1133,6 +1335,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_add_vmaster);
 
 /* switch */
 int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
@@ -1146,6 +1349,7 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.integer.max = 1;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_info);
 
 int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
@@ -1165,6 +1369,7 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
 			 HDA_AMP_MUTE) ? 0 : 1;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get);
 
 int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
@@ -1195,6 +1400,7 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
 	snd_hda_power_down(codec);
 	return change;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put);
 
 /*
  * bound volume controls
@@ -1220,6 +1426,7 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->spdif_mutex);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get);
 
 int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol)
@@ -1243,6 +1450,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->spdif_mutex);
 	return err < 0 ? err : change;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put);
 
 /*
  * generic bound volume/swtich controls
@@ -1262,6 +1470,7 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->spdif_mutex);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info);
 
 int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
@@ -1278,6 +1487,7 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->spdif_mutex);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get);
 
 int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
@@ -1300,6 +1510,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->spdif_mutex);
 	return err < 0 ? err : change;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put);
 
 int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 			   unsigned int size, unsigned int __user *tlv)
@@ -1316,6 +1527,7 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	mutex_unlock(&codec->spdif_mutex);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_tlv);
 
 struct hda_ctl_ops snd_hda_bind_vol = {
 	.info = snd_hda_mixer_amp_volume_info,
@@ -1323,6 +1535,7 @@ struct hda_ctl_ops snd_hda_bind_vol = {
 	.put = snd_hda_mixer_amp_volume_put,
 	.tlv = snd_hda_mixer_amp_tlv
 };
+EXPORT_SYMBOL_HDA(snd_hda_bind_vol);
 
 struct hda_ctl_ops snd_hda_bind_sw = {
 	.info = snd_hda_mixer_amp_switch_info,
@@ -1330,6 +1543,7 @@ struct hda_ctl_ops snd_hda_bind_sw = {
 	.put = snd_hda_mixer_amp_switch_put,
 	.tlv = snd_hda_mixer_amp_tlv
 };
+EXPORT_SYMBOL_HDA(snd_hda_bind_sw);
 
 /*
  * SPDIF out controls
@@ -1577,9 +1791,11 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
 	}
 	for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
+		if (!kctl)
+			return -ENOMEM;
 		kctl->id.index = idx;
 		kctl->private_value = nid;
-		err = snd_ctl_add(codec->bus->card, kctl);
+		err = snd_hda_ctl_add(codec, kctl);
 		if (err < 0)
 			return err;
 	}
@@ -1589,6 +1805,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
 	codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
 
 /*
  * SPDIF sharing with analog output
@@ -1623,9 +1840,10 @@ int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
 	if (!mout->dig_out_nid)
 		return 0;
 	/* ATTENTION: here mout is passed as private_data, instead of codec */
-	return snd_ctl_add(codec->bus->card,
+	return snd_hda_ctl_add(codec,
 			   snd_ctl_new1(&spdif_share_sw, mout));
 }
+EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw);
 
 /*
  * SPDIF input
@@ -1725,7 +1943,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 	for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
 		kctl->private_value = nid;
-		err = snd_ctl_add(codec->bus->card, kctl);
+		err = snd_hda_ctl_add(codec, kctl);
 		if (err < 0)
 			return err;
 	}
@@ -1735,6 +1953,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 		AC_DIG1_ENABLE;
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls);
 
 #ifdef SND_HDA_NEEDS_RESUME
 /*
@@ -1761,10 +1980,14 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 			      int direct, unsigned int verb, unsigned int parm)
 {
+	struct hda_bus *bus = codec->bus;
+	unsigned int res;
 	int err;
+
+	res = make_codec_cmd(codec, nid, direct, verb, parm);
 	snd_hda_power_up(codec);
-	mutex_lock(&codec->bus->cmd_mutex);
-	err = codec->bus->ops.command(codec, nid, direct, verb, parm);
+	mutex_lock(&bus->cmd_mutex);
+	err = bus->ops.command(bus, res);
 	if (!err) {
 		struct hda_cache_head *c;
 		u32 key = build_cmd_cache_key(nid, verb);
@@ -1772,18 +1995,19 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 		if (c)
 			c->val = parm;
 	}
-	mutex_unlock(&codec->bus->cmd_mutex);
+	mutex_unlock(&bus->cmd_mutex);
 	snd_hda_power_down(codec);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache);
 
 /* resume the all commands from the cache */
 void snd_hda_codec_resume_cache(struct hda_codec *codec)
 {
-	struct hda_cache_head *buffer = codec->cmd_cache.buffer;
+	struct hda_cache_head *buffer = codec->cmd_cache.buf.list;
 	int i;
 
-	for (i = 0; i < codec->cmd_cache.size; i++, buffer++) {
+	for (i = 0; i < codec->cmd_cache.buf.used; i++, buffer++) {
 		u32 key = buffer->key;
 		if (!key)
 			continue;
@@ -1791,6 +2015,7 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec)
 				    get_cmd_cache_cmd(key), buffer->val);
 	}
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_resume_cache);
 
 /**
  * snd_hda_sequence_write_cache - sequence writes with caching
@@ -1808,6 +2033,7 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec,
 		snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb,
 					  seq->param);
 }
+EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache);
 #endif /* SND_HDA_NEEDS_RESUME */
 
 /*
@@ -1868,6 +2094,17 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	}
 }
 
+#ifdef CONFIG_SND_HDA_HWDEP
+/* execute additional init verbs */
+static void hda_exec_init_verbs(struct hda_codec *codec)
+{
+	if (codec->init_verbs.list)
+		snd_hda_sequence_write(codec, codec->init_verbs.list);
+}
+#else
+static inline void hda_exec_init_verbs(struct hda_codec *codec) {}
+#endif
+
 #ifdef SND_HDA_NEEDS_RESUME
 /*
  * call suspend and power-down; used both from PM and power-save
@@ -1894,6 +2131,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D0);
+	hda_exec_init_verbs(codec);
 	if (codec->patch_ops.resume)
 		codec->patch_ops.resume(codec);
 	else {
@@ -1914,28 +2152,37 @@ static void hda_call_codec_resume(struct hda_codec *codec)
  *
  * Returns 0 if successful, otherwise a negative error code.
  */
-int __devinit snd_hda_build_controls(struct hda_bus *bus)
+int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus)
 {
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
-		int err = 0;
-		/* fake as if already powered-on */
-		hda_keep_power_on(codec);
-		/* then fire up */
-		hda_set_power_state(codec,
-				    codec->afg ? codec->afg : codec->mfg,
-				    AC_PWRST_D0);
-		/* continue to initialize... */
-		if (codec->patch_ops.init)
-			err = codec->patch_ops.init(codec);
-		if (!err && codec->patch_ops.build_controls)
-			err = codec->patch_ops.build_controls(codec);
-		snd_hda_power_down(codec);
+		int err = snd_hda_codec_build_controls(codec);
 		if (err < 0)
 			return err;
 	}
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_build_controls);
 
+int snd_hda_codec_build_controls(struct hda_codec *codec)
+{
+	int err = 0;
+	/* fake as if already powered-on */
+	hda_keep_power_on(codec);
+	/* then fire up */
+	hda_set_power_state(codec,
+			    codec->afg ? codec->afg : codec->mfg,
+			    AC_PWRST_D0);
+	hda_exec_init_verbs(codec);
+	/* continue to initialize... */
+	if (codec->patch_ops.init)
+		err = codec->patch_ops.init(codec);
+	if (!err && codec->patch_ops.build_controls)
+		err = codec->patch_ops.build_controls(codec);
+	snd_hda_power_down(codec);
+	if (err < 0)
+		return err;
 	return 0;
 }
 
@@ -2028,6 +2275,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 
 	return val;
 }
+EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
 
 /**
  * snd_hda_query_supported_pcm - query the supported PCM rates and formats
@@ -2042,7 +2290,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
  *
  * Returns 0 if successful, otherwise a negative error code.
  */
-int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
 {
 	int i;
@@ -2207,6 +2455,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 
 	return 1;
 }
+EXPORT_SYMBOL_HDA(snd_hda_is_supported_format);
 
 /*
  * PCM stuff
@@ -2236,8 +2485,8 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static int __devinit set_pcm_default_values(struct hda_codec *codec,
-					    struct hda_pcm_stream *info)
+static int set_pcm_default_values(struct hda_codec *codec,
+				  struct hda_pcm_stream *info)
 {
 	/* query support PCM information from the given NID */
 	if (info->nid && (!info->rates || !info->formats)) {
@@ -2263,6 +2512,110 @@ static int __devinit set_pcm_default_values(struct hda_codec *codec,
 	return 0;
 }
 
+/*
+ * get the empty PCM device number to assign
+ */
+static int get_empty_pcm_device(struct hda_bus *bus, int type)
+{
+	static const char *dev_name[HDA_PCM_NTYPES] = {
+		"Audio", "SPDIF", "HDMI", "Modem"
+	};
+	/* starting device index for each PCM type */
+	static int dev_idx[HDA_PCM_NTYPES] = {
+		[HDA_PCM_TYPE_AUDIO] = 0,
+		[HDA_PCM_TYPE_SPDIF] = 1,
+		[HDA_PCM_TYPE_HDMI] = 3,
+		[HDA_PCM_TYPE_MODEM] = 6
+	};
+	/* normal audio device indices; not linear to keep compatibility */
+	static int audio_idx[4] = { 0, 2, 4, 5 };
+	int i, dev;
+
+	switch (type) {
+	case HDA_PCM_TYPE_AUDIO:
+		for (i = 0; i < ARRAY_SIZE(audio_idx); i++) {
+			dev = audio_idx[i];
+			if (!test_bit(dev, bus->pcm_dev_bits))
+				break;
+		}
+		if (i >= ARRAY_SIZE(audio_idx)) {
+			snd_printk(KERN_WARNING "Too many audio devices\n");
+			return -EAGAIN;
+		}
+		break;
+	case HDA_PCM_TYPE_SPDIF:
+	case HDA_PCM_TYPE_HDMI:
+	case HDA_PCM_TYPE_MODEM:
+		dev = dev_idx[type];
+		if (test_bit(dev, bus->pcm_dev_bits)) {
+			snd_printk(KERN_WARNING "%s already defined\n",
+				   dev_name[type]);
+			return -EAGAIN;
+		}
+		break;
+	default:
+		snd_printk(KERN_WARNING "Invalid PCM type %d\n", type);
+		return -EINVAL;
+	}
+	set_bit(dev, bus->pcm_dev_bits);
+	return dev;
+}
+
+/*
+ * attach a new PCM stream
+ */
+static int snd_hda_attach_pcm(struct hda_codec *codec, struct hda_pcm *pcm)
+{
+	struct hda_bus *bus = codec->bus;
+	struct hda_pcm_stream *info;
+	int stream, err;
+
+	if (snd_BUG_ON(!pcm->name))
+		return -EINVAL;
+	for (stream = 0; stream < 2; stream++) {
+		info = &pcm->stream[stream];
+		if (info->substreams) {
+			err = set_pcm_default_values(codec, info);
+			if (err < 0)
+				return err;
+		}
+	}
+	return bus->ops.attach_pcm(bus, codec, pcm);
+}
+
+/* assign all PCMs of the given codec */
+int snd_hda_codec_build_pcms(struct hda_codec *codec)
+{
+	unsigned int pcm;
+	int err;
+
+	if (!codec->num_pcms) {
+		if (!codec->patch_ops.build_pcms)
+			return 0;
+		err = codec->patch_ops.build_pcms(codec);
+		if (err < 0)
+			return err;
+	}
+	for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+		struct hda_pcm *cpcm = &codec->pcm_info[pcm];
+		int dev;
+
+		if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
+			return 0; /* no substreams assigned */
+
+		if (!cpcm->pcm) {
+			dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type);
+			if (dev < 0)
+				return 0;
+			cpcm->device = dev;
+			err = snd_hda_attach_pcm(codec, cpcm);
+			if (err < 0)
+				return err;
+		}
+	}
+	return 0;
+}
+
 /**
  * snd_hda_build_pcms - build PCM information
  * @bus: the BUS
@@ -2294,27 +2647,13 @@ int __devinit snd_hda_build_pcms(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
-		unsigned int pcm, s;
-		int err;
-		if (!codec->patch_ops.build_pcms)
-			continue;
-		err = codec->patch_ops.build_pcms(codec);
+		int err = snd_hda_codec_build_pcms(codec);
 		if (err < 0)
 			return err;
-		for (pcm = 0; pcm < codec->num_pcms; pcm++) {
-			for (s = 0; s < 2; s++) {
-				struct hda_pcm_stream *info;
-				info = &codec->pcm_info[pcm].stream[s];
-				if (!info->substreams)
-					continue;
-				err = set_pcm_default_values(codec, info);
-				if (err < 0)
-					return err;
-			}
-		}
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_build_pcms);
 
 /**
  * snd_hda_check_board_config - compare the current codec with the config table
@@ -2333,11 +2672,11 @@ int snd_hda_check_board_config(struct hda_codec *codec,
 			       int num_configs, const char **models,
 			       const struct snd_pci_quirk *tbl)
 {
-	if (codec->bus->modelname && models) {
+	if (codec->modelname && models) {
 		int i;
 		for (i = 0; i < num_configs; i++) {
 			if (models[i] &&
-			    !strcmp(codec->bus->modelname, models[i])) {
+			    !strcmp(codec->modelname, models[i])) {
 				snd_printd(KERN_INFO "hda_codec: model '%s' is "
 					   "selected\n", models[i]);
 				return i;
@@ -2370,6 +2709,7 @@ int snd_hda_check_board_config(struct hda_codec *codec,
 	}
 	return -1;
 }
+EXPORT_SYMBOL_HDA(snd_hda_check_board_config);
 
 /**
  * snd_hda_add_new_ctls - create controls from the array
@@ -2390,7 +2730,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
 		kctl = snd_ctl_new1(knew, codec);
 		if (!kctl)
 			return -ENOMEM;
-		err = snd_ctl_add(codec->bus->card, kctl);
+		err = snd_hda_ctl_add(codec, kctl);
 		if (err < 0) {
 			if (!codec->addr)
 				return err;
@@ -2398,13 +2738,14 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
 			if (!kctl)
 				return -ENOMEM;
 			kctl->id.device = codec->addr;
-			err = snd_ctl_add(codec->bus->card, kctl);
+			err = snd_hda_ctl_add(codec, kctl);
 			if (err < 0)
 				return err;
 		}
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls);
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
@@ -2414,6 +2755,7 @@ static void hda_power_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, power_work.work);
+	struct hda_bus *bus = codec->bus;
 
 	if (!codec->power_on || codec->power_count) {
 		codec->power_transition = 0;
@@ -2421,8 +2763,8 @@ static void hda_power_work(struct work_struct *work)
 	}
 
 	hda_call_codec_suspend(codec);
-	if (codec->bus->ops.pm_notify)
-		codec->bus->ops.pm_notify(codec);
+	if (bus->ops.pm_notify)
+		bus->ops.pm_notify(bus);
 }
 
 static void hda_keep_power_on(struct hda_codec *codec)
@@ -2433,29 +2775,39 @@ static void hda_keep_power_on(struct hda_codec *codec)
 
 void snd_hda_power_up(struct hda_codec *codec)
 {
+	struct hda_bus *bus = codec->bus;
+
 	codec->power_count++;
 	if (codec->power_on || codec->power_transition)
 		return;
 
 	codec->power_on = 1;
-	if (codec->bus->ops.pm_notify)
-		codec->bus->ops.pm_notify(codec);
+	if (bus->ops.pm_notify)
+		bus->ops.pm_notify(bus);
 	hda_call_codec_resume(codec);
 	cancel_delayed_work(&codec->power_work);
 	codec->power_transition = 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_power_up);
+
+#define power_save(codec)	\
+	((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
+
+#define power_save(codec)	\
+	((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
 
 void snd_hda_power_down(struct hda_codec *codec)
 {
 	--codec->power_count;
 	if (!codec->power_on || codec->power_count || codec->power_transition)
 		return;
-	if (power_save) {
+	if (power_save(codec)) {
 		codec->power_transition = 1; /* avoid reentrance */
 		schedule_delayed_work(&codec->power_work,
-				      msecs_to_jiffies(power_save * 1000));
+				msecs_to_jiffies(power_save(codec) * 1000));
 	}
 }
+EXPORT_SYMBOL_HDA(snd_hda_power_down);
 
 int snd_hda_check_amp_list_power(struct hda_codec *codec,
 				 struct hda_loopback_check *check,
@@ -2492,6 +2844,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power);
 #endif
 
 /*
@@ -2511,6 +2864,7 @@ int snd_hda_ch_mode_info(struct hda_codec *codec,
 		chmode[uinfo->value.enumerated.item].channels);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_ch_mode_info);
 
 int snd_hda_ch_mode_get(struct hda_codec *codec,
 			struct snd_ctl_elem_value *ucontrol,
@@ -2528,6 +2882,7 @@ int snd_hda_ch_mode_get(struct hda_codec *codec,
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_ch_mode_get);
 
 int snd_hda_ch_mode_put(struct hda_codec *codec,
 			struct snd_ctl_elem_value *ucontrol,
@@ -2548,6 +2903,7 @@ int snd_hda_ch_mode_put(struct hda_codec *codec,
 		snd_hda_sequence_write_cache(codec, chmode[mode].sequence);
 	return 1;
 }
+EXPORT_SYMBOL_HDA(snd_hda_ch_mode_put);
 
 /*
  * input MUX helper
@@ -2568,6 +2924,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux,
 	strcpy(uinfo->value.enumerated.name, imux->items[index].label);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_input_mux_info);
 
 int snd_hda_input_mux_put(struct hda_codec *codec,
 			  const struct hda_input_mux *imux,
@@ -2589,6 +2946,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 	*cur_val = idx;
 	return 1;
 }
+EXPORT_SYMBOL_HDA(snd_hda_input_mux_put);
 
 
 /*
@@ -2641,6 +2999,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_open);
 
 int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 				  struct hda_multi_out *mout,
@@ -2653,6 +3012,7 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare);
 
 /*
  * release the digital out
@@ -2665,6 +3025,7 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_close);
 
 /*
  * set up more restrictions for analog out
@@ -2704,6 +3065,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
 	return snd_pcm_hw_constraint_step(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 }
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_open);
 
 /*
  * set up the i/o for analog out
@@ -2762,6 +3124,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare);
 
 /*
  * clean up the setting for analog out
@@ -2788,6 +3151,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup);
 
 /*
  * Helper for automatic pin configuration
@@ -3073,11 +3437,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config);
 
 /* labels for input pins */
 const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = {
 	"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux"
 };
+EXPORT_SYMBOL_HDA(auto_pin_cfg_labels);
 
 
 #ifdef CONFIG_PM
@@ -3105,11 +3471,11 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
 	}
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_suspend);
 
 /**
  * snd_hda_resume - resume the codecs
  * @bus: the HDA bus
- * @state: resume state
  *
  * Returns 0 if successful.
  *
@@ -3126,16 +3492,79 @@ int snd_hda_resume(struct hda_bus *bus)
 	}
 	return 0;
 }
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-int snd_hda_codecs_inuse(struct hda_bus *bus)
-{
-	struct hda_codec *codec;
+EXPORT_SYMBOL_HDA(snd_hda_resume);
+#endif /* CONFIG_PM */
 
-	list_for_each_entry(codec, &bus->codec_list, list) {
-		if (snd_hda_codec_needs_resume(codec))
-			return 1;
+/*
+ * generic arrays
+ */
+
+/* get a new element from the given array
+ * if it exceeds the pre-allocated array size, re-allocate the array
+ */
+void *snd_array_new(struct snd_array *array)
+{
+	if (array->used >= array->alloced) {
+		int num = array->alloced + array->alloc_align;
+		void *nlist;
+		if (snd_BUG_ON(num >= 4096))
+			return NULL;
+		nlist = kcalloc(num + 1, array->elem_size, GFP_KERNEL);
+		if (!nlist)
+			return NULL;
+		if (array->list) {
+			memcpy(nlist, array->list,
+			       array->elem_size * array->alloced);
+			kfree(array->list);
+		}
+		array->list = nlist;
+		array->alloced = num;
 	}
-	return 0;
+	return snd_array_elem(array, array->used++);
 }
-#endif
-#endif
+EXPORT_SYMBOL_HDA(snd_array_new);
+
+/* free the given array elements */
+void snd_array_free(struct snd_array *array)
+{
+	kfree(array->list);
+	array->used = 0;
+	array->alloced = 0;
+	array->list = NULL;
+}
+EXPORT_SYMBOL_HDA(snd_array_free);
+
+/*
+ * used by hda_proc.c and hda_eld.c
+ */
+void snd_print_pcm_rates(int pcm, char *buf, int buflen)
+{
+	static unsigned int rates[] = {
+		8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
+		96000, 176400, 192000, 384000
+	};
+	int i, j;
+
+	for (i = 0, j = 0; i < ARRAY_SIZE(rates); i++)
+		if (pcm & (1 << i))
+			j += snprintf(buf + j, buflen - j,  " %d", rates[i]);
+
+	buf[j] = '\0'; /* necessary when j == 0 */
+}
+EXPORT_SYMBOL_HDA(snd_print_pcm_rates);
+
+void snd_print_pcm_bits(int pcm, char *buf, int buflen)
+{
+	static unsigned int bits[] = { 8, 16, 20, 24, 32 };
+	int i, j;
+
+	for (i = 0, j = 0; i < ARRAY_SIZE(bits); i++)
+		if (pcm & (AC_SUPPCM_BITS_8 << i))
+			j += snprintf(buf + j, buflen - j,  " %d", bits[i]);
+
+	buf[j] = '\0'; /* necessary when j == 0 */
+}
+EXPORT_SYMBOL_HDA(snd_print_pcm_bits);
+
+MODULE_DESCRIPTION("HDA codec core");
+MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 60468f562400..729fc7642d7f 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -519,6 +519,36 @@ enum {
 /* max. codec address */
 #define HDA_MAX_CODEC_ADDRESS	0x0f
 
+/*
+ * generic arrays
+ */
+struct snd_array {
+	unsigned int used;
+	unsigned int alloced;
+	unsigned int elem_size;
+	unsigned int alloc_align;
+	void *list;
+};
+
+void *snd_array_new(struct snd_array *array);
+void snd_array_free(struct snd_array *array);
+static inline void snd_array_init(struct snd_array *array, unsigned int size,
+				  unsigned int align)
+{
+	array->elem_size = size;
+	array->alloc_align = align;
+}
+
+static inline void *snd_array_elem(struct snd_array *array, unsigned int idx)
+{
+	return array->list + idx * array->elem_size;
+}
+
+static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
+{
+	return (unsigned long)(ptr - array->list) / array->elem_size;
+}
+
 /*
  * Structures
  */
@@ -536,15 +566,17 @@ typedef u16 hda_nid_t;
 /* bus operators */
 struct hda_bus_ops {
 	/* send a single command */
-	int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
-		       unsigned int verb, unsigned int parm);
+	int (*command)(struct hda_bus *bus, unsigned int cmd);
 	/* get a response from the last command */
-	unsigned int (*get_response)(struct hda_codec *codec);
+	unsigned int (*get_response)(struct hda_bus *bus);
 	/* free the private data */
 	void (*private_free)(struct hda_bus *);
+	/* attach a PCM stream */
+	int (*attach_pcm)(struct hda_bus *bus, struct hda_codec *codec,
+			  struct hda_pcm *pcm);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	/* notify power-up/down from codec to controller */
-	void (*pm_notify)(struct hda_codec *codec);
+	void (*pm_notify)(struct hda_bus *bus);
 #endif
 };
 
@@ -553,6 +585,7 @@ struct hda_bus_template {
 	void *private_data;
 	struct pci_dev *pci;
 	const char *modelname;
+	int *power_save;
 	struct hda_bus_ops ops;
 };
 
@@ -569,6 +602,7 @@ struct hda_bus {
 	void *private_data;
 	struct pci_dev *pci;
 	const char *modelname;
+	int *power_save;
 	struct hda_bus_ops ops;
 
 	/* codec linked list */
@@ -581,10 +615,12 @@ struct hda_bus {
 	/* unsolicited event queue */
 	struct hda_bus_unsolicited *unsol;
 
-	struct snd_info_entry *proc;
+	/* assigned PCMs */
+	DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES);
 
 	/* misc op flags */
 	unsigned int needs_damn_long_delay :1;
+	unsigned int shutdown :1;	/* being unloaded */
 };
 
 /*
@@ -604,6 +640,16 @@ struct hda_codec_preset {
 	int (*patch)(struct hda_codec *codec);
 };
 	
+struct hda_codec_preset_list {
+	const struct hda_codec_preset *preset;
+	struct module *owner;
+	struct list_head list;
+};
+
+/* initial hook */
+int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset);
+int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset);
+
 /* ops set by the preset patch */
 struct hda_codec_ops {
 	int (*build_controls)(struct hda_codec *codec);
@@ -635,10 +681,7 @@ struct hda_amp_info {
 
 struct hda_cache_rec {
 	u16 hash[64];			/* hash table for index */
-	unsigned int num_entries;	/* number of assigned entries */
-	unsigned int size;		/* allocated size */
-	unsigned int record_size;	/* record size (including header) */
-	void *buffer;			/* hash table entries */
+	struct snd_array buf;		/* record entries */
 };
 
 /* PCM callbacks */
@@ -680,7 +723,8 @@ struct hda_pcm {
 	char *name;
 	struct hda_pcm_stream stream[2];
 	unsigned int pcm_type;	/* HDA_PCM_TYPE_XXX */
-	int device;	/* assigned device number */
+	int device;		/* device number to assign */
+	struct snd_pcm *pcm;	/* assigned PCM instance */
 };
 
 /* codec information */
@@ -699,6 +743,9 @@ struct hda_codec {
 
 	/* detected preset */
 	const struct hda_codec_preset *preset;
+	struct module *owner;
+	const char *name;	/* codec name */
+	const char *modelname;	/* model name for preset */
 
 	/* set by patch */
 	struct hda_codec_ops patch_ops;
@@ -718,6 +765,8 @@ struct hda_codec {
 	hda_nid_t start_nid;
 	u32 *wcaps;
 
+	struct snd_array mixers;	/* list of assigned mixer elements */
+
 	struct hda_cache_rec amp_cache;	/* cache for amp access */
 	struct hda_cache_rec cmd_cache;	/* cache for other commands */
 
@@ -727,7 +776,11 @@ struct hda_codec {
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
 	hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
 
+#ifdef CONFIG_SND_HDA_HWDEP
 	struct snd_hwdep *hwdep;	/* assigned hwdep device */
+	struct snd_array init_verbs;	/* additional init verbs */
+	struct snd_array hints;		/* additional hints */
+#endif
 
 	/* misc flags */
 	unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each
@@ -740,6 +793,10 @@ struct hda_codec {
 	int power_count;	/* current (global) power refcount */
 	struct delayed_work power_work; /* delayed task for powerdown */
 #endif
+
+	/* codec-specific additional proc output */
+	void (*proc_widget_hook)(struct snd_info_buffer *buffer,
+				 struct hda_codec *codec, hda_nid_t nid);
 };
 
 /* direction */
@@ -754,7 +811,7 @@ enum {
 int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
 		    struct hda_bus **busp);
 int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
-		      struct hda_codec **codecp);
+		      int do_init, struct hda_codec **codecp);
 
 /*
  * low level functions
@@ -799,11 +856,13 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec);
  * Mixer
  */
 int snd_hda_build_controls(struct hda_bus *bus);
+int snd_hda_codec_build_controls(struct hda_codec *codec);
 
 /*
  * PCM
  */
 int snd_hda_build_pcms(struct hda_bus *bus);
+int snd_hda_codec_build_pcms(struct hda_codec *codec);
 void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 				u32 stream_tag,
 				int channel_id, int format);
@@ -812,8 +871,6 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 					unsigned int channels,
 					unsigned int format,
 					unsigned int maxbps);
-int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
-				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
 int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int format);
 
@@ -830,6 +887,13 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state);
 int snd_hda_resume(struct hda_bus *bus);
 #endif
 
+/*
+ * get widget information
+ */
+const char *snd_hda_get_jack_connectivity(u32 cfg);
+const char *snd_hda_get_jack_type(u32 cfg);
+const char *snd_hda_get_jack_location(u32 cfg);
+
 /*
  * power saving
  */
@@ -837,12 +901,25 @@ int snd_hda_resume(struct hda_bus *bus);
 void snd_hda_power_up(struct hda_codec *codec);
 void snd_hda_power_down(struct hda_codec *codec);
 #define snd_hda_codec_needs_resume(codec) codec->power_count
-int snd_hda_codecs_inuse(struct hda_bus *bus);
 #else
 static inline void snd_hda_power_up(struct hda_codec *codec) {}
 static inline void snd_hda_power_down(struct hda_codec *codec) {}
 #define snd_hda_codec_needs_resume(codec) 1
-#define snd_hda_codecs_inuse(bus) 1
+#endif
+
+/*
+ * Codec modularization
+ */
+
+/* Export symbols only for communication with codec drivers;
+ * When built in kernel, all HD-audio drivers are supposed to be statically
+ * linked to the kernel.  Thus, the symbols don't have to (or shouldn't) be
+ * exported unless it's built as a module.
+ */
+#ifdef MODULE
+#define EXPORT_SYMBOL_HDA(sym) EXPORT_SYMBOL_GPL(sym)
+#else
+#define EXPORT_SYMBOL_HDA(sym)
 #endif
 
 #endif /* __SOUND_HDA_CODEC_H */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
new file mode 100644
index 000000000000..fcad5ec31773
--- /dev/null
+++ b/sound/pci/hda/hda_eld.c
@@ -0,0 +1,590 @@
+/*
+ * Generic routines and proc interface for ELD(EDID Like Data) information
+ *
+ * Copyright(c) 2008 Intel Corporation.
+ *
+ * Authors:
+ * 		Wu Fengguang <wfg@linux.intel.com>
+ *
+ *  This driver is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 2 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This driver is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <sound/core.h>
+#include <asm/unaligned.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+enum eld_versions {
+	ELD_VER_CEA_861D	= 2,
+	ELD_VER_PARTIAL		= 31,
+};
+
+enum cea_edid_versions {
+	CEA_EDID_VER_NONE	= 0,
+	CEA_EDID_VER_CEA861	= 1,
+	CEA_EDID_VER_CEA861A	= 2,
+	CEA_EDID_VER_CEA861BCD	= 3,
+	CEA_EDID_VER_RESERVED	= 4,
+};
+
+static char *cea_speaker_allocation_names[] = {
+	/*  0 */ "FL/FR",
+	/*  1 */ "LFE",
+	/*  2 */ "FC",
+	/*  3 */ "RL/RR",
+	/*  4 */ "RC",
+	/*  5 */ "FLC/FRC",
+	/*  6 */ "RLC/RRC",
+	/*  7 */ "FLW/FRW",
+	/*  8 */ "FLH/FRH",
+	/*  9 */ "TC",
+	/* 10 */ "FCH",
+};
+
+static char *eld_connection_type_names[4] = {
+	"HDMI",
+	"DisplayPort",
+	"2-reserved",
+	"3-reserved"
+};
+
+enum cea_audio_coding_types {
+	AUDIO_CODING_TYPE_REF_STREAM_HEADER	=  0,
+	AUDIO_CODING_TYPE_LPCM			=  1,
+	AUDIO_CODING_TYPE_AC3			=  2,
+	AUDIO_CODING_TYPE_MPEG1			=  3,
+	AUDIO_CODING_TYPE_MP3			=  4,
+	AUDIO_CODING_TYPE_MPEG2			=  5,
+	AUDIO_CODING_TYPE_AACLC			=  6,
+	AUDIO_CODING_TYPE_DTS			=  7,
+	AUDIO_CODING_TYPE_ATRAC			=  8,
+	AUDIO_CODING_TYPE_SACD			=  9,
+	AUDIO_CODING_TYPE_EAC3			= 10,
+	AUDIO_CODING_TYPE_DTS_HD		= 11,
+	AUDIO_CODING_TYPE_MLP			= 12,
+	AUDIO_CODING_TYPE_DST			= 13,
+	AUDIO_CODING_TYPE_WMAPRO		= 14,
+	AUDIO_CODING_TYPE_REF_CXT		= 15,
+	/* also include valid xtypes below */
+	AUDIO_CODING_TYPE_HE_AAC		= 15,
+	AUDIO_CODING_TYPE_HE_AAC2		= 16,
+	AUDIO_CODING_TYPE_MPEG_SURROUND		= 17,
+};
+
+enum cea_audio_coding_xtypes {
+	AUDIO_CODING_XTYPE_HE_REF_CT		= 0,
+	AUDIO_CODING_XTYPE_HE_AAC		= 1,
+	AUDIO_CODING_XTYPE_HE_AAC2		= 2,
+	AUDIO_CODING_XTYPE_MPEG_SURROUND	= 3,
+	AUDIO_CODING_XTYPE_FIRST_RESERVED	= 4,
+};
+
+static char *cea_audio_coding_type_names[] = {
+	/*  0 */ "undefined",
+	/*  1 */ "LPCM",
+	/*  2 */ "AC-3",
+	/*  3 */ "MPEG1",
+	/*  4 */ "MP3",
+	/*  5 */ "MPEG2",
+	/*  6 */ "AAC-LC",
+	/*  7 */ "DTS",
+	/*  8 */ "ATRAC",
+	/*  9 */ "DSD (One Bit Audio)",
+	/* 10 */ "E-AC-3/DD+ (Dolby Digital Plus)",
+	/* 11 */ "DTS-HD",
+	/* 12 */ "MLP (Dolby TrueHD)",
+	/* 13 */ "DST",
+	/* 14 */ "WMAPro",
+	/* 15 */ "HE-AAC",
+	/* 16 */ "HE-AACv2",
+	/* 17 */ "MPEG Surround",
+};
+
+/*
+ * The following two lists are shared between
+ * 	- HDMI audio InfoFrame (source to sink)
+ * 	- CEA E-EDID Extension (sink to source)
+ */
+
+/*
+ * SS1:SS0 index => sample size
+ */
+static int cea_sample_sizes[4] = {
+	0,	 		/* 0: Refer to Stream Header */
+	AC_SUPPCM_BITS_16,	/* 1: 16 bits */
+	AC_SUPPCM_BITS_20,	/* 2: 20 bits */
+	AC_SUPPCM_BITS_24,	/* 3: 24 bits */
+};
+
+/*
+ * SF2:SF1:SF0 index => sampling frequency
+ */
+static int cea_sampling_frequencies[8] = {
+	0,			/* 0: Refer to Stream Header */
+	SNDRV_PCM_RATE_32000,	/* 1:  32000Hz */
+	SNDRV_PCM_RATE_44100,	/* 2:  44100Hz */
+	SNDRV_PCM_RATE_48000,	/* 3:  48000Hz */
+	SNDRV_PCM_RATE_88200,	/* 4:  88200Hz */
+	SNDRV_PCM_RATE_96000,	/* 5:  96000Hz */
+	SNDRV_PCM_RATE_176400,	/* 6: 176400Hz */
+	SNDRV_PCM_RATE_192000,	/* 7: 192000Hz */
+};
+
+static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+					int byte_index)
+{
+	unsigned int val;
+
+	val = snd_hda_codec_read(codec, nid, 0,
+					AC_VERB_GET_HDMI_ELDD, byte_index);
+
+#ifdef BE_PARANOID
+	printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
+#endif
+
+	if ((val & AC_ELDD_ELD_VALID) == 0) {
+		snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
+								byte_index);
+		val = 0;
+	}
+
+	return val & AC_ELDD_ELD_DATA;
+}
+
+#define GRAB_BITS(buf, byte, lowbit, bits) 		\
+({							\
+	BUILD_BUG_ON(lowbit > 7);			\
+	BUILD_BUG_ON(bits > 8);				\
+	BUILD_BUG_ON(bits <= 0);			\
+							\
+	(buf[byte] >> (lowbit)) & ((1 << (bits)) - 1);	\
+})
+
+static void hdmi_update_short_audio_desc(struct cea_sad *a,
+					 const unsigned char *buf)
+{
+	int i;
+	int val;
+
+	val = GRAB_BITS(buf, 1, 0, 7);
+	a->rates = 0;
+	for (i = 0; i < 7; i++)
+		if (val & (1 << i))
+			a->rates |= cea_sampling_frequencies[i + 1];
+
+	a->channels = GRAB_BITS(buf, 0, 0, 3);
+	a->channels++;
+
+	a->format = GRAB_BITS(buf, 0, 3, 4);
+	switch (a->format) {
+	case AUDIO_CODING_TYPE_REF_STREAM_HEADER:
+		snd_printd(KERN_INFO
+				"HDMI: audio coding type 0 not expected\n");
+		break;
+
+	case AUDIO_CODING_TYPE_LPCM:
+		val = GRAB_BITS(buf, 2, 0, 3);
+		a->sample_bits = 0;
+		for (i = 0; i < 3; i++)
+			if (val & (1 << i))
+				a->sample_bits |= cea_sample_sizes[i + 1];
+		break;
+
+	case AUDIO_CODING_TYPE_AC3:
+	case AUDIO_CODING_TYPE_MPEG1:
+	case AUDIO_CODING_TYPE_MP3:
+	case AUDIO_CODING_TYPE_MPEG2:
+	case AUDIO_CODING_TYPE_AACLC:
+	case AUDIO_CODING_TYPE_DTS:
+	case AUDIO_CODING_TYPE_ATRAC:
+		a->max_bitrate = GRAB_BITS(buf, 2, 0, 8);
+		a->max_bitrate *= 8000;
+		break;
+
+	case AUDIO_CODING_TYPE_SACD:
+		break;
+
+	case AUDIO_CODING_TYPE_EAC3:
+		break;
+
+	case AUDIO_CODING_TYPE_DTS_HD:
+		break;
+
+	case AUDIO_CODING_TYPE_MLP:
+		break;
+
+	case AUDIO_CODING_TYPE_DST:
+		break;
+
+	case AUDIO_CODING_TYPE_WMAPRO:
+		a->profile = GRAB_BITS(buf, 2, 0, 3);
+		break;
+
+	case AUDIO_CODING_TYPE_REF_CXT:
+		a->format = GRAB_BITS(buf, 2, 3, 5);
+		if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT ||
+		    a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) {
+			snd_printd(KERN_INFO
+				"HDMI: audio coding xtype %d not expected\n",
+				a->format);
+			a->format = 0;
+		} else
+			a->format += AUDIO_CODING_TYPE_HE_AAC -
+				     AUDIO_CODING_XTYPE_HE_AAC;
+		break;
+	}
+}
+
+/*
+ * Be careful, ELD buf could be totally rubbish!
+ */
+static int hdmi_update_eld(struct hdmi_eld *e,
+			   const unsigned char *buf, int size)
+{
+	int mnl;
+	int i;
+
+	e->eld_ver = GRAB_BITS(buf, 0, 3, 5);
+	if (e->eld_ver != ELD_VER_CEA_861D &&
+	    e->eld_ver != ELD_VER_PARTIAL) {
+		snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n",
+								e->eld_ver);
+		goto out_fail;
+	}
+
+	e->eld_size = size;
+	e->baseline_len = GRAB_BITS(buf, 2, 0, 8);
+	mnl		= GRAB_BITS(buf, 4, 0, 5);
+	e->cea_edid_ver	= GRAB_BITS(buf, 4, 5, 3);
+
+	e->support_hdcp	= GRAB_BITS(buf, 5, 0, 1);
+	e->support_ai	= GRAB_BITS(buf, 5, 1, 1);
+	e->conn_type	= GRAB_BITS(buf, 5, 2, 2);
+	e->sad_count	= GRAB_BITS(buf, 5, 4, 4);
+
+	e->aud_synch_delay = GRAB_BITS(buf, 6, 0, 8) * 2;
+	e->spk_alloc	= GRAB_BITS(buf, 7, 0, 7);
+
+	e->port_id	  = get_unaligned_le64(buf + 8);
+
+	/* not specified, but the spec's tendency is little endian */
+	e->manufacture_id = get_unaligned_le16(buf + 16);
+	e->product_id	  = get_unaligned_le16(buf + 18);
+
+	if (mnl > ELD_MAX_MNL) {
+		snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl);
+		goto out_fail;
+	} else if (ELD_FIXED_BYTES + mnl > size) {
+		snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
+		goto out_fail;
+	} else
+		strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl);
+
+	for (i = 0; i < e->sad_count; i++) {
+		if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
+			snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i);
+			goto out_fail;
+		}
+		hdmi_update_short_audio_desc(e->sad + i,
+					buf + ELD_FIXED_BYTES + mnl + 3 * i);
+	}
+
+	return 0;
+
+out_fail:
+	e->eld_ver = 0;
+	return -EINVAL;
+}
+
+static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0);
+}
+
+static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid)
+{
+	int eldv;
+	int present;
+
+	present = hdmi_present_sense(codec, nid);
+	eldv    = (present & AC_PINSENSE_ELDV);
+	present = (present & AC_PINSENSE_PRESENCE);
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+	printk(KERN_INFO "HDMI: sink_present = %d, eld_valid = %d\n",
+			!!present, !!eldv);
+#endif
+
+	return eldv && present;
+}
+
+int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE,
+						 AC_DIPSIZE_ELD_BUF);
+}
+
+int snd_hdmi_get_eld(struct hdmi_eld *eld,
+		     struct hda_codec *codec, hda_nid_t nid)
+{
+	int i;
+	int ret;
+	int size;
+	unsigned char *buf;
+
+	if (!hdmi_eld_valid(codec, nid))
+		return -ENOENT;
+
+	size = snd_hdmi_get_eld_size(codec, nid);
+	if (size == 0) {
+		/* wfg: workaround for ASUS P5E-VM HDMI board */
+		snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n");
+		size = 128;
+	}
+	if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) {
+		snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size);
+		return -ERANGE;
+	}
+
+	buf = kmalloc(size, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	for (i = 0; i < size; i++)
+		buf[i] = hdmi_get_eld_byte(codec, nid, i);
+
+	ret = hdmi_update_eld(eld, buf, size);
+
+	kfree(buf);
+	return ret;
+}
+
+static void hdmi_show_short_audio_desc(struct cea_sad *a)
+{
+	char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
+	char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits =";
+
+	if (!a->format)
+		return;
+
+	snd_print_pcm_rates(a->rates, buf, sizeof(buf));
+
+	if (a->format == AUDIO_CODING_TYPE_LPCM)
+		snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+	else if (a->max_bitrate)
+		snprintf(buf2, sizeof(buf2),
+				", max bitrate = %d", a->max_bitrate);
+	else
+		buf2[0] = '\0';
+
+	printk(KERN_INFO "HDMI: supports coding type %s:"
+			" channels = %d, rates =%s%s\n",
+			cea_audio_coding_type_names[a->format],
+			a->channels,
+			buf,
+			buf2);
+}
+
+void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen)
+{
+	int i, j;
+
+	for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) {
+		if (spk_alloc & (1 << i))
+			j += snprintf(buf + j, buflen - j,  " %s",
+					cea_speaker_allocation_names[i]);
+	}
+	buf[j] = '\0';	/* necessary when j == 0 */
+}
+
+void snd_hdmi_show_eld(struct hdmi_eld *e)
+{
+	int i;
+
+	printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
+			e->monitor_name,
+			eld_connection_type_names[e->conn_type]);
+
+	if (e->spk_alloc) {
+		char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
+		snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
+		printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
+	}
+
+	for (i = 0; i < e->sad_count; i++)
+		hdmi_show_short_audio_desc(e->sad + i);
+}
+
+#ifdef CONFIG_PROC_FS
+
+static void hdmi_print_sad_info(int i, struct cea_sad *a,
+				struct snd_info_buffer *buffer)
+{
+	char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
+
+	snd_iprintf(buffer, "sad%d_coding_type\t[0x%x] %s\n",
+			i, a->format, cea_audio_coding_type_names[a->format]);
+	snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels);
+
+	snd_print_pcm_rates(a->rates, buf, sizeof(buf));
+	snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf);
+
+	if (a->format == AUDIO_CODING_TYPE_LPCM) {
+		snd_print_pcm_bits(a->sample_bits, buf, sizeof(buf));
+		snd_iprintf(buffer, "sad%d_bits\t\t[0x%x]%s\n",
+							i, a->sample_bits, buf);
+	}
+
+	if (a->max_bitrate)
+		snd_iprintf(buffer, "sad%d_max_bitrate\t%d\n",
+							i, a->max_bitrate);
+
+	if (a->profile)
+		snd_iprintf(buffer, "sad%d_profile\t\t%d\n", i, a->profile);
+}
+
+static void hdmi_print_eld_info(struct snd_info_entry *entry,
+				struct snd_info_buffer *buffer)
+{
+	struct hdmi_eld *e = entry->private_data;
+	char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
+	int i;
+	static char *eld_versoin_names[32] = {
+		"reserved",
+		"reserved",
+		"CEA-861D or below",
+		[3 ... 30] = "reserved",
+		[31] = "partial"
+	};
+	static char *cea_edid_version_names[8] = {
+		"no CEA EDID Timing Extension block present",
+		"CEA-861",
+		"CEA-861-A",
+		"CEA-861-B, C or D",
+		[4 ... 7] = "reserved"
+	};
+
+	snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name);
+	snd_iprintf(buffer, "connection_type\t\t%s\n",
+				eld_connection_type_names[e->conn_type]);
+	snd_iprintf(buffer, "eld_version\t\t[0x%x] %s\n", e->eld_ver,
+					eld_versoin_names[e->eld_ver]);
+	snd_iprintf(buffer, "edid_version\t\t[0x%x] %s\n", e->cea_edid_ver,
+				cea_edid_version_names[e->cea_edid_ver]);
+	snd_iprintf(buffer, "manufacture_id\t\t0x%x\n", e->manufacture_id);
+	snd_iprintf(buffer, "product_id\t\t0x%x\n", e->product_id);
+	snd_iprintf(buffer, "port_id\t\t\t0x%llx\n", (long long)e->port_id);
+	snd_iprintf(buffer, "support_hdcp\t\t%d\n", e->support_hdcp);
+	snd_iprintf(buffer, "support_ai\t\t%d\n", e->support_ai);
+	snd_iprintf(buffer, "audio_sync_delay\t%d\n", e->aud_synch_delay);
+
+	snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
+	snd_iprintf(buffer, "speakers\t\t[0x%x]%s\n", e->spk_alloc, buf);
+
+	snd_iprintf(buffer, "sad_count\t\t%d\n", e->sad_count);
+
+	for (i = 0; i < e->sad_count; i++)
+		hdmi_print_sad_info(i, e->sad + i, buffer);
+}
+
+static void hdmi_write_eld_info(struct snd_info_entry *entry,
+				struct snd_info_buffer *buffer)
+{
+	struct hdmi_eld *e = entry->private_data;
+	char line[64];
+	char name[64];
+	char *sname;
+	long long val;
+	int n;
+
+	while (!snd_info_get_line(buffer, line, sizeof(line))) {
+		if (sscanf(line, "%s %llx", name, &val) != 2)
+			continue;
+		/*
+		 * We don't allow modification to these fields:
+		 * 	monitor_name manufacture_id product_id
+		 * 	eld_version edid_version
+		 */
+		if (!strcmp(name, "connection_type"))
+			e->conn_type = val;
+		else if (!strcmp(name, "port_id"))
+			e->port_id = val;
+		else if (!strcmp(name, "support_hdcp"))
+			e->support_hdcp = val;
+		else if (!strcmp(name, "support_ai"))
+			e->support_ai = val;
+		else if (!strcmp(name, "audio_sync_delay"))
+			e->aud_synch_delay = val;
+		else if (!strcmp(name, "speakers"))
+			e->spk_alloc = val;
+		else if (!strcmp(name, "sad_count"))
+			e->sad_count = val;
+		else if (!strncmp(name, "sad", 3)) {
+			sname = name + 4;
+			n = name[3] - '0';
+			if (name[4] >= '0' && name[4] <= '9') {
+				sname++;
+				n = 10 * n + name[4] - '0';
+			}
+			if (n < 0 || n > 31) /* double the CEA limit */
+				continue;
+			if (!strcmp(sname, "_coding_type"))
+				e->sad[n].format = val;
+			else if (!strcmp(sname, "_channels"))
+				e->sad[n].channels = val;
+			else if (!strcmp(sname, "_rates"))
+				e->sad[n].rates = val;
+			else if (!strcmp(sname, "_bits"))
+				e->sad[n].sample_bits = val;
+			else if (!strcmp(sname, "_max_bitrate"))
+				e->sad[n].max_bitrate = val;
+			else if (!strcmp(sname, "_profile"))
+				e->sad[n].profile = val;
+			if (n >= e->sad_count)
+				e->sad_count = n + 1;
+		}
+	}
+}
+
+
+int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld)
+{
+	char name[32];
+	struct snd_info_entry *entry;
+	int err;
+
+	snprintf(name, sizeof(name), "eld#%d", codec->addr);
+	err = snd_card_proc_new(codec->bus->card, name, &entry);
+	if (err < 0)
+		return err;
+
+	snd_info_set_text_ops(entry, eld, hdmi_print_eld_info);
+	entry->c.text.write = hdmi_write_eld_info;
+	entry->mode |= S_IWUSR;
+	eld->proc_entry = entry;
+
+	return 0;
+}
+
+void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
+{
+	if (!codec->bus->shutdown && eld->proc_entry) {
+		snd_device_free(codec->bus->card, eld->proc_entry);
+		eld->proc_entry = NULL;
+	}
+}
+
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 0ca30894f7c6..65745e96dc70 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -723,7 +723,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
 		if (is_loopback)
 			add_input_loopback(codec, node->nid, HDA_INPUT, index);
 		snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
-		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+		err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+		if (err < 0)
 			return err;
 		created = 1;
 	} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
@@ -732,7 +733,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
 		if (is_loopback)
 			add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
 		snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
-		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+		err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+		if (err < 0)
 			return err;
 		created = 1;
 	}
@@ -745,14 +747,16 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
 	    (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) {
 		knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT);
 		snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
-		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+		err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+		if (err < 0)
 			return err;
 		created = 1;
 	} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
 		   (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) {
 		knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT);
 		snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
-		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+		err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+		if (err < 0)
 			return err;
 		created = 1;
 	}
@@ -849,8 +853,8 @@ static int build_input_controls(struct hda_codec *codec)
 	}
 
 	/* create input MUX if multiple sources are available */
-	if ((err = snd_ctl_add(codec->bus->card,
-			       snd_ctl_new1(&cap_sel, codec))) < 0)
+	err = snd_hda_ctl_add(codec, snd_ctl_new1(&cap_sel, codec));
+	if (err < 0)
 		return err;
 
 	/* no volume control? */
@@ -867,8 +871,8 @@ static int build_input_controls(struct hda_codec *codec)
 			HDA_CODEC_VOLUME(name, adc_node->nid,
 					 spec->input_mux.items[i].index,
 					 HDA_INPUT);
-		if ((err = snd_ctl_add(codec->bus->card,
-				       snd_ctl_new1(&knew, codec))) < 0)
+		err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+		if (err < 0)
 			return err;
 	}
 
@@ -1097,3 +1101,4 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec)
 	snd_hda_generic_free(codec);
 	return err;
 }
+EXPORT_SYMBOL(snd_hda_parse_generic_codec);
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 6e18a422d993..300ab407cf42 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -23,10 +23,12 @@
 #include <linux/pci.h>
 #include <linux/compat.h>
 #include <linux/mutex.h>
+#include <linux/ctype.h>
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
+#include <sound/minors.h>
 
 /*
  * write/read an out-of-bound verb
@@ -95,7 +97,26 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
 	return 0;
 }
 
-int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
+static void clear_hwdep_elements(struct hda_codec *codec)
+{
+	char **head;
+	int i;
+
+	/* clear init verbs */
+	snd_array_free(&codec->init_verbs);
+	/* clear hints */
+	head = codec->hints.list;
+	for (i = 0; i < codec->hints.used; i++, head++)
+		kfree(*head);
+	snd_array_free(&codec->hints);
+}
+
+static void hwdep_free(struct snd_hwdep *hwdep)
+{
+	clear_hwdep_elements(hwdep->private_data);
+}
+
+int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 {
 	char hwname[16];
 	struct snd_hwdep *hwdep;
@@ -109,6 +130,7 @@ int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
 	sprintf(hwdep->name, "HDA Codec %d", codec->addr);
 	hwdep->iface = SNDRV_HWDEP_IFACE_HDA;
 	hwdep->private_data = codec;
+	hwdep->private_free = hwdep_free;
 	hwdep->exclusive = 1;
 
 	hwdep->ops.open = hda_hwdep_open;
@@ -117,5 +139,215 @@ int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
 	hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat;
 #endif
 
+	snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
+	snd_array_init(&codec->hints, sizeof(char *), 32);
+
 	return 0;
 }
+
+#ifdef CONFIG_SND_HDA_RECONFIG
+
+/*
+ * sysfs interface
+ */
+
+static int clear_codec(struct hda_codec *codec)
+{
+	snd_hda_codec_reset(codec);
+	clear_hwdep_elements(codec);
+	return 0;
+}
+
+static int reconfig_codec(struct hda_codec *codec)
+{
+	int err;
+
+	snd_printk(KERN_INFO "hda-codec: reconfiguring\n");
+	snd_hda_codec_reset(codec);
+	err = snd_hda_codec_configure(codec);
+	if (err < 0)
+		return err;
+	/* rebuild PCMs */
+	err = snd_hda_codec_build_pcms(codec);
+	if (err < 0)
+		return err;
+	/* rebuild mixers */
+	err = snd_hda_codec_build_controls(codec);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+/*
+ * allocate a string at most len chars, and remove the trailing EOL
+ */
+static char *kstrndup_noeol(const char *src, size_t len)
+{
+	char *s = kstrndup(src, len, GFP_KERNEL);
+	char *p;
+	if (!s)
+		return NULL;
+	p = strchr(s, '\n');
+	if (p)
+		*p = 0;
+	return s;
+}
+
+#define CODEC_INFO_SHOW(type)					\
+static ssize_t type##_show(struct device *dev,			\
+			   struct device_attribute *attr,	\
+			   char *buf)				\
+{								\
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
+	struct hda_codec *codec = hwdep->private_data;		\
+	return sprintf(buf, "0x%x\n", codec->type);		\
+}
+
+#define CODEC_INFO_STR_SHOW(type)				\
+static ssize_t type##_show(struct device *dev,			\
+			     struct device_attribute *attr,	\
+					char *buf)		\
+{								\
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
+	struct hda_codec *codec = hwdep->private_data;		\
+	return sprintf(buf, "%s\n",				\
+		       codec->type ? codec->type : "");		\
+}
+
+CODEC_INFO_SHOW(vendor_id);
+CODEC_INFO_SHOW(subsystem_id);
+CODEC_INFO_SHOW(revision_id);
+CODEC_INFO_SHOW(afg);
+CODEC_INFO_SHOW(mfg);
+CODEC_INFO_STR_SHOW(name);
+CODEC_INFO_STR_SHOW(modelname);
+
+#define CODEC_INFO_STORE(type)					\
+static ssize_t type##_store(struct device *dev,			\
+			    struct device_attribute *attr,	\
+			    const char *buf, size_t count)	\
+{								\
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
+	struct hda_codec *codec = hwdep->private_data;		\
+	char *after;						\
+	codec->type = simple_strtoul(buf, &after, 0);		\
+	return count;						\
+}
+
+#define CODEC_INFO_STR_STORE(type)				\
+static ssize_t type##_store(struct device *dev,			\
+			    struct device_attribute *attr,	\
+			    const char *buf, size_t count)	\
+{								\
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
+	struct hda_codec *codec = hwdep->private_data;		\
+	char *s = kstrndup_noeol(buf, 64);			\
+	if (!s)							\
+		return -ENOMEM;					\
+	kfree(codec->type);					\
+	codec->type = s;					\
+	return count;						\
+}
+
+CODEC_INFO_STORE(vendor_id);
+CODEC_INFO_STORE(subsystem_id);
+CODEC_INFO_STORE(revision_id);
+CODEC_INFO_STR_STORE(name);
+CODEC_INFO_STR_STORE(modelname);
+
+#define CODEC_ACTION_STORE(type)				\
+static ssize_t type##_store(struct device *dev,			\
+			    struct device_attribute *attr,	\
+			    const char *buf, size_t count)	\
+{								\
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
+	struct hda_codec *codec = hwdep->private_data;		\
+	int err = 0;						\
+	if (*buf)						\
+		err = type##_codec(codec);			\
+	return err < 0 ? err : count;				\
+}
+
+CODEC_ACTION_STORE(reconfig);
+CODEC_ACTION_STORE(clear);
+
+static ssize_t init_verbs_store(struct device *dev,
+				struct device_attribute *attr,
+				const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	char *p;
+	struct hda_verb verb, *v;
+
+	verb.nid = simple_strtoul(buf, &p, 0);
+	verb.verb = simple_strtoul(p, &p, 0);
+	verb.param = simple_strtoul(p, &p, 0);
+	if (!verb.nid || !verb.verb || !verb.param)
+		return -EINVAL;
+	v = snd_array_new(&codec->init_verbs);
+	if (!v)
+		return -ENOMEM;
+	*v = verb;
+	return count;
+}
+
+static ssize_t hints_store(struct device *dev,
+			   struct device_attribute *attr,
+			   const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	char *p;
+	char **hint;
+
+	if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n')
+		return count;
+	p = kstrndup_noeol(buf, 1024);
+	if (!p)
+		return -ENOMEM;
+	hint = snd_array_new(&codec->hints);
+	if (!hint) {
+		kfree(p);
+		return -ENOMEM;
+	}
+	*hint = p;
+	return count;
+}
+
+#define CODEC_ATTR_RW(type) \
+	__ATTR(type, 0644, type##_show, type##_store)
+#define CODEC_ATTR_RO(type) \
+	__ATTR_RO(type)
+#define CODEC_ATTR_WO(type) \
+	__ATTR(type, 0200, NULL, type##_store)
+
+static struct device_attribute codec_attrs[] = {
+	CODEC_ATTR_RW(vendor_id),
+	CODEC_ATTR_RW(subsystem_id),
+	CODEC_ATTR_RW(revision_id),
+	CODEC_ATTR_RO(afg),
+	CODEC_ATTR_RO(mfg),
+	CODEC_ATTR_RW(name),
+	CODEC_ATTR_RW(modelname),
+	CODEC_ATTR_WO(init_verbs),
+	CODEC_ATTR_WO(hints),
+	CODEC_ATTR_WO(reconfig),
+	CODEC_ATTR_WO(clear),
+};
+
+/*
+ * create sysfs files on hwdep directory
+ */
+int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
+{
+	struct snd_hwdep *hwdep = codec->hwdep;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(codec_attrs); i++)
+		snd_add_device_sysfs_file(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card,
+					  hwdep->device, &codec_attrs[i]);
+	return 0;
+}
+
+#endif /* CONFIG_SND_HDA_RECONFIG */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 35722ec920cb..f04de115ee11 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -58,6 +58,7 @@ static char *model[SNDRV_CARDS];
 static int position_fix[SNDRV_CARDS];
 static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
 static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
+static int probe_only[SNDRV_CARDS];
 static int single_cmd;
 static int enable_msi;
 
@@ -76,6 +77,8 @@ module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
 MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1).");
+module_param_array(probe_only, bool, NULL, 0444);
+MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization.");
 module_param(single_cmd, bool, 0444);
 MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
 		 "(for debugging only).");
@@ -83,7 +86,10 @@ module_param(enable_msi, int, 0444);
 MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-/* power_save option is defined in hda_codec.c */
+static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+		 "(in second, 0 = disable).");
 
 /* reset the HD-audio controller in power save mode.
  * this may give more power-saving, but will take longer time to
@@ -292,6 +298,8 @@ enum {
 /* Define VIA HD Audio Device ID*/
 #define VIA_HDAC_DEVICE_ID		0x3288
 
+/* HD Audio class code */
+#define PCI_CLASS_MULTIMEDIA_HD_AUDIO	0x0403
 
 /*
  */
@@ -392,6 +400,7 @@ struct azx {
 	unsigned int msi :1;
 	unsigned int irq_pending_warned :1;
 	unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */
+	unsigned int probing :1; /* codec probing phase */
 
 	/* for debugging */
 	unsigned int last_cmd;	/* last issued command (to sync) */
@@ -414,6 +423,7 @@ enum {
 	AZX_DRIVER_ULI,
 	AZX_DRIVER_NVIDIA,
 	AZX_DRIVER_TERA,
+	AZX_DRIVER_GENERIC,
 	AZX_NUM_DRIVERS, /* keep this as last entry */
 };
 
@@ -427,6 +437,7 @@ static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_ULI] = "HDA ULI M5461",
 	[AZX_DRIVER_NVIDIA] = "HDA NVidia",
 	[AZX_DRIVER_TERA] = "HDA Teradici", 
+	[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
 };
 
 /*
@@ -527,9 +538,9 @@ static void azx_free_cmd_io(struct azx *chip)
 }
 
 /* send a command */
-static int azx_corb_send_cmd(struct hda_codec *codec, u32 val)
+static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
 {
-	struct azx *chip = codec->bus->private_data;
+	struct azx *chip = bus->private_data;
 	unsigned int wp;
 
 	/* add command to corb */
@@ -577,9 +588,9 @@ static void azx_update_rirb(struct azx *chip)
 }
 
 /* receive a response */
-static unsigned int azx_rirb_get_response(struct hda_codec *codec)
+static unsigned int azx_rirb_get_response(struct hda_bus *bus)
 {
-	struct azx *chip = codec->bus->private_data;
+	struct azx *chip = bus->private_data;
 	unsigned long timeout;
 
  again:
@@ -596,7 +607,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
 		}
 		if (time_after(jiffies, timeout))
 			break;
-		if (codec->bus->needs_damn_long_delay)
+		if (bus->needs_damn_long_delay)
 			msleep(2); /* temporary workaround */
 		else {
 			udelay(10);
@@ -624,6 +635,14 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
 		goto again;
 	}
 
+	if (chip->probing) {
+		/* If this critical timeout happens during the codec probing
+		 * phase, this is likely an access to a non-existing codec
+		 * slot.  Better to return an error and reset the system.
+		 */
+		return -1;
+	}
+
 	snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
 		   "switching to single_cmd mode: last cmd=0x%08x\n",
 		   chip->last_cmd);
@@ -646,9 +665,9 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
  */
 
 /* send a command */
-static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
+static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
 {
-	struct azx *chip = codec->bus->private_data;
+	struct azx *chip = bus->private_data;
 	int timeout = 50;
 
 	while (timeout--) {
@@ -671,9 +690,9 @@ static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
 }
 
 /* receive a response */
-static unsigned int azx_single_get_response(struct hda_codec *codec)
+static unsigned int azx_single_get_response(struct hda_bus *bus)
 {
-	struct azx *chip = codec->bus->private_data;
+	struct azx *chip = bus->private_data;
 	int timeout = 50;
 
 	while (timeout--) {
@@ -696,38 +715,29 @@ static unsigned int azx_single_get_response(struct hda_codec *codec)
  */
 
 /* send a command */
-static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid,
-			int direct, unsigned int verb,
-			unsigned int para)
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
 {
-	struct azx *chip = codec->bus->private_data;
-	u32 val;
+	struct azx *chip = bus->private_data;
 
-	val = (u32)(codec->addr & 0x0f) << 28;
-	val |= (u32)direct << 27;
-	val |= (u32)nid << 20;
-	val |= verb << 8;
-	val |= para;
 	chip->last_cmd = val;
-
 	if (chip->single_cmd)
-		return azx_single_send_cmd(codec, val);
+		return azx_single_send_cmd(bus, val);
 	else
-		return azx_corb_send_cmd(codec, val);
+		return azx_corb_send_cmd(bus, val);
 }
 
 /* get a response */
-static unsigned int azx_get_response(struct hda_codec *codec)
+static unsigned int azx_get_response(struct hda_bus *bus)
 {
-	struct azx *chip = codec->bus->private_data;
+	struct azx *chip = bus->private_data;
 	if (chip->single_cmd)
-		return azx_single_get_response(codec);
+		return azx_single_get_response(bus);
 	else
-		return azx_rirb_get_response(codec);
+		return azx_rirb_get_response(bus);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static void azx_power_notify(struct hda_codec *codec);
+static void azx_power_notify(struct hda_bus *bus);
 #endif
 
 /* reset codec link */
@@ -1184,6 +1194,28 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 	return 0;
 }
 
+/*
+ * Probe the given codec address
+ */
+static int probe_codec(struct azx *chip, int addr)
+{
+	unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) |
+		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
+	unsigned int res;
+
+	chip->probing = 1;
+	azx_send_cmd(chip->bus, cmd);
+	res = azx_get_response(chip->bus);
+	chip->probing = 0;
+	if (res == -1)
+		return -EIO;
+	snd_printdd("hda_intel: codec #%d probed OK\n", addr);
+	return 0;
+}
+
+static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
+				 struct hda_pcm *cpcm);
+static void azx_stop_chip(struct azx *chip);
 
 /*
  * Codec initialization
@@ -1194,21 +1226,13 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
 	[AZX_DRIVER_TERA] = 1,
 };
 
-/* number of slots to probe as default
- * this can be different from azx_max_codecs[] -- e.g. some boards
- * report wrongly the non-existing 4th slot availability
- */
-static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = {
-	[AZX_DRIVER_ICH] = 3,
-	[AZX_DRIVER_ATI] = 3,
-};
-
 static int __devinit azx_codec_create(struct azx *chip, const char *model,
-				      unsigned int codec_probe_mask)
+				      unsigned int codec_probe_mask,
+				      int no_init)
 {
 	struct hda_bus_template bus_temp;
-	int c, codecs, audio_codecs, err;
-	int def_slots, max_slots;
+	int c, codecs, err;
+	int max_slots;
 
 	memset(&bus_temp, 0, sizeof(bus_temp));
 	bus_temp.private_data = chip;
@@ -1216,7 +1240,9 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 	bus_temp.pci = chip->pci;
 	bus_temp.ops.command = azx_send_cmd;
 	bus_temp.ops.get_response = azx_get_response;
+	bus_temp.ops.attach_pcm = azx_attach_pcm_stream;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
+	bus_temp.power_save = &power_save;
 	bus_temp.ops.pm_notify = azx_power_notify;
 #endif
 
@@ -1227,33 +1253,43 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 	if (chip->driver_type == AZX_DRIVER_NVIDIA)
 		chip->bus->needs_damn_long_delay = 1;
 
-	codecs = audio_codecs = 0;
+	codecs = 0;
 	max_slots = azx_max_codecs[chip->driver_type];
 	if (!max_slots)
 		max_slots = AZX_MAX_CODECS;
-	def_slots = azx_default_codecs[chip->driver_type];
-	if (!def_slots)
-		def_slots = max_slots;
-	for (c = 0; c < def_slots; c++) {
+
+	/* First try to probe all given codec slots */
+	for (c = 0; c < max_slots; c++) {
+		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+			if (probe_codec(chip, c) < 0) {
+				/* Some BIOSen give you wrong codec addresses
+				 * that don't exist
+				 */
+				snd_printk(KERN_WARNING
+					   "hda_intel: Codec #%d probe error; "
+					   "disabling it...\n", c);
+				chip->codec_mask &= ~(1 << c);
+				/* More badly, accessing to a non-existing
+				 * codec often screws up the controller chip,
+				 * and distrubs the further communications.
+				 * Thus if an error occurs during probing,
+				 * better to reset the controller chip to
+				 * get back to the sanity state.
+				 */
+				azx_stop_chip(chip);
+				azx_init_chip(chip);
+			}
+		}
+	}
+
+	/* Then create codec instances */
+	for (c = 0; c < max_slots; c++) {
 		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
 			struct hda_codec *codec;
-			err = snd_hda_codec_new(chip->bus, c, &codec);
+			err = snd_hda_codec_new(chip->bus, c, !no_init, &codec);
 			if (err < 0)
 				continue;
 			codecs++;
-			if (codec->afg)
-				audio_codecs++;
-		}
-	}
-	if (!audio_codecs) {
-		/* probe additional slots if no codec is found */
-		for (; c < max_slots; c++) {
-			if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
-				err = snd_hda_codec_new(chip->bus, c, NULL);
-				if (err < 0)
-					continue;
-				codecs++;
-			}
 		}
 	}
 	if (!codecs) {
@@ -1722,111 +1758,59 @@ static struct snd_pcm_ops azx_pcm_ops = {
 
 static void azx_pcm_free(struct snd_pcm *pcm)
 {
-	kfree(pcm->private_data);
+	struct azx_pcm *apcm = pcm->private_data;
+	if (apcm) {
+		apcm->chip->pcm[pcm->device] = NULL;
+		kfree(apcm);
+	}
 }
 
-static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
-				      struct hda_pcm *cpcm)
+static int
+azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
+		      struct hda_pcm *cpcm)
 {
-	int err;
+	struct azx *chip = bus->private_data;
 	struct snd_pcm *pcm;
 	struct azx_pcm *apcm;
+	int pcm_dev = cpcm->device;
+	int s, err;
 
-	/* if no substreams are defined for both playback and capture,
-	 * it's just a placeholder.  ignore it.
-	 */
-	if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
-		return 0;
-
-	if (snd_BUG_ON(!cpcm->name))
+	if (pcm_dev >= AZX_MAX_PCMS) {
+		snd_printk(KERN_ERR SFX "Invalid PCM device number %d\n",
+			   pcm_dev);
 		return -EINVAL;
-
-	err = snd_pcm_new(chip->card, cpcm->name, cpcm->device,
-			  cpcm->stream[0].substreams,
-			  cpcm->stream[1].substreams,
+	}
+	if (chip->pcm[pcm_dev]) {
+		snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev);
+		return -EBUSY;
+	}
+	err = snd_pcm_new(chip->card, cpcm->name, pcm_dev,
+			  cpcm->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams,
+			  cpcm->stream[SNDRV_PCM_STREAM_CAPTURE].substreams,
 			  &pcm);
 	if (err < 0)
 		return err;
 	strcpy(pcm->name, cpcm->name);
-	apcm = kmalloc(sizeof(*apcm), GFP_KERNEL);
+	apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
 	if (apcm == NULL)
 		return -ENOMEM;
 	apcm->chip = chip;
 	apcm->codec = codec;
-	apcm->hinfo[0] = &cpcm->stream[0];
-	apcm->hinfo[1] = &cpcm->stream[1];
 	pcm->private_data = apcm;
 	pcm->private_free = azx_pcm_free;
-	if (cpcm->stream[0].substreams)
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops);
-	if (cpcm->stream[1].substreams)
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops);
+	if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM)
+		pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
+	chip->pcm[pcm_dev] = pcm;
+	cpcm->pcm = pcm;
+	for (s = 0; s < 2; s++) {
+		apcm->hinfo[s] = &cpcm->stream[s];
+		if (cpcm->stream[s].substreams)
+			snd_pcm_set_ops(pcm, s, &azx_pcm_ops);
+	}
+	/* buffer pre-allocation */
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
 					      snd_dma_pci_data(chip->pci),
 					      1024 * 64, 32 * 1024 * 1024);
-	chip->pcm[cpcm->device] = pcm;
-	return 0;
-}
-
-static int __devinit azx_pcm_create(struct azx *chip)
-{
-	static const char *dev_name[HDA_PCM_NTYPES] = {
-		"Audio", "SPDIF", "HDMI", "Modem"
-	};
-	/* starting device index for each PCM type */
-	static int dev_idx[HDA_PCM_NTYPES] = {
-		[HDA_PCM_TYPE_AUDIO] = 0,
-		[HDA_PCM_TYPE_SPDIF] = 1,
-		[HDA_PCM_TYPE_HDMI] = 3,
-		[HDA_PCM_TYPE_MODEM] = 6
-	};
-	/* normal audio device indices; not linear to keep compatibility */
-	static int audio_idx[4] = { 0, 2, 4, 5 };
-	struct hda_codec *codec;
-	int c, err;
-	int num_devs[HDA_PCM_NTYPES];
-
-	err = snd_hda_build_pcms(chip->bus);
-	if (err < 0)
-		return err;
-
-	/* create audio PCMs */
-	memset(num_devs, 0, sizeof(num_devs));
-	list_for_each_entry(codec, &chip->bus->codec_list, list) {
-		for (c = 0; c < codec->num_pcms; c++) {
-			struct hda_pcm *cpcm = &codec->pcm_info[c];
-			int type = cpcm->pcm_type;
-			switch (type) {
-			case HDA_PCM_TYPE_AUDIO:
-				if (num_devs[type] >= ARRAY_SIZE(audio_idx)) {
-					snd_printk(KERN_WARNING
-						   "Too many audio devices\n");
-					continue;
-				}
-				cpcm->device = audio_idx[num_devs[type]];
-				break;
-			case HDA_PCM_TYPE_SPDIF:
-			case HDA_PCM_TYPE_HDMI:
-			case HDA_PCM_TYPE_MODEM:
-				if (num_devs[type]) {
-					snd_printk(KERN_WARNING
-						   "%s already defined\n",
-						   dev_name[type]);
-					continue;
-				}
-				cpcm->device = dev_idx[type];
-				break;
-			default:
-				snd_printk(KERN_WARNING
-					   "Invalid PCM type %d\n", type);
-				continue;
-			}
-			num_devs[type]++;
-			err = create_codec_pcm(chip, codec, cpcm);
-			if (err < 0)
-				return err;
-		}
-	}
 	return 0;
 }
 
@@ -1903,13 +1887,13 @@ static void azx_stop_chip(struct azx *chip)
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 /* power-up/down the controller */
-static void azx_power_notify(struct hda_codec *codec)
+static void azx_power_notify(struct hda_bus *bus)
 {
-	struct azx *chip = codec->bus->private_data;
+	struct azx *chip = bus->private_data;
 	struct hda_codec *c;
 	int power_on = 0;
 
-	list_for_each_entry(c, &codec->bus->codec_list, list) {
+	list_for_each_entry(c, &bus->codec_list, list) {
 		if (c->power_on) {
 			power_on = 1;
 			break;
@@ -1926,6 +1910,18 @@ static void azx_power_notify(struct hda_codec *codec)
 /*
  * power management
  */
+
+static int snd_hda_codecs_inuse(struct hda_bus *bus)
+{
+	struct hda_codec *codec;
+
+	list_for_each_entry(codec, &bus->codec_list, list) {
+		if (snd_hda_codec_needs_resume(codec))
+			return 1;
+	}
+	return 0;
+}
+
 static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
@@ -1951,13 +1947,16 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 	return 0;
 }
 
+static int azx_resume_early(struct pci_dev *pci)
+{
+	return pci_restore_state(pci);
+}
+
 static int azx_resume(struct pci_dev *pci)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct azx *chip = card->private_data;
 
-	pci_set_power_state(pci, PCI_D0);
-	pci_restore_state(pci);
 	if (pci_enable_device(pci) < 0) {
 		printk(KERN_ERR "hda-intel: pci_enable_device failed, "
 		       "disabling device\n");
@@ -2095,6 +2094,10 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1014, 0x05b7, "Thinkpad Z60", 0x01),
 	SND_PCI_QUIRK(0x17aa, 0x2010, "Thinkpad X/T/R60", 0x01),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X/T/R61", 0x01),
+	/* broken BIOS */
+	SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
+	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
+	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
 	{}
 };
 
@@ -2229,6 +2232,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 			chip->playback_streams = ATIHDMI_NUM_PLAYBACK;
 			chip->capture_streams = ATIHDMI_NUM_CAPTURE;
 			break;
+		case AZX_DRIVER_GENERIC:
 		default:
 			chip->playback_streams = ICH6_NUM_PLAYBACK;
 			chip->capture_streams = ICH6_NUM_CAPTURE;
@@ -2338,40 +2342,31 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	}
 
 	err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
-	if (err < 0) {
-		snd_card_free(card);
-		return err;
-	}
+	if (err < 0)
+		goto out_free;
 	card->private_data = chip;
 
 	/* create codec instances */
-	err = azx_codec_create(chip, model[dev], probe_mask[dev]);
-	if (err < 0) {
-		snd_card_free(card);
-		return err;
-	}
+	err = azx_codec_create(chip, model[dev], probe_mask[dev],
+			       probe_only[dev]);
+	if (err < 0)
+		goto out_free;
 
 	/* create PCM streams */
-	err = azx_pcm_create(chip);
-	if (err < 0) {
-		snd_card_free(card);
-		return err;
-	}
+	err = snd_hda_build_pcms(chip->bus);
+	if (err < 0)
+		goto out_free;
 
 	/* create mixer controls */
 	err = azx_mixer_create(chip);
-	if (err < 0) {
-		snd_card_free(card);
-		return err;
-	}
+	if (err < 0)
+		goto out_free;
 
 	snd_card_set_dev(card, &pci->dev);
 
 	err = snd_card_register(card);
-	if (err < 0) {
-		snd_card_free(card);
-		return err;
-	}
+	if (err < 0)
+		goto out_free;
 
 	pci_set_drvdata(pci, card);
 	chip->running = 1;
@@ -2380,6 +2375,9 @@ static int __devinit azx_probe(struct pci_dev *pci,
 
 	dev++;
 	return err;
+out_free:
+	snd_card_free(card);
+	return err;
 }
 
 static void __devexit azx_remove(struct pci_dev *pci)
@@ -2453,6 +2451,11 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
 	/* Teradici */
 	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
+	/* AMD Generic, PCI class code and Vendor ID for HD Audio */
+	{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
+	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
+	  .class_mask = 0xffffff,
+	  .driver_data = AZX_DRIVER_GENERIC },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
@@ -2465,6 +2468,7 @@ static struct pci_driver driver = {
 	.remove = __devexit_p(azx_remove),
 #ifdef CONFIG_PM
 	.suspend = azx_suspend,
+	.resume_early = azx_resume_early,
 	.resume = azx_resume,
 #endif
 };
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 7957fefda730..6f2fe0f9fdd8 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -96,6 +96,8 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 					    const char *name);
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 			unsigned int *tlv, const char **slaves);
+void snd_hda_codec_reset(struct hda_codec *codec);
+int snd_hda_codec_configure(struct hda_codec *codec);
 
 /* amp value bits */
 #define HDA_AMP_MUTE	0x80
@@ -282,6 +284,12 @@ int snd_hda_codec_proc_new(struct hda_codec *codec);
 static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
 #endif
 
+#define SND_PRINT_RATES_ADVISED_BUFSIZE	80
+void snd_print_pcm_rates(int pcm, char *buf, int buflen);
+
+#define SND_PRINT_BITS_ADVISED_BUFSIZE	16
+void snd_print_pcm_bits(int pcm, char *buf, int buflen);
+
 /*
  * Misc
  */
@@ -364,17 +372,17 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 /* amp values */
 #define AMP_IN_MUTE(idx)	(0x7080 | ((idx)<<8))
 #define AMP_IN_UNMUTE(idx)	(0x7000 | ((idx)<<8))
-#define AMP_OUT_MUTE	0xb080
-#define AMP_OUT_UNMUTE	0xb000
-#define AMP_OUT_ZERO	0xb000
+#define AMP_OUT_MUTE		0xb080
+#define AMP_OUT_UNMUTE		0xb000
+#define AMP_OUT_ZERO		0xb000
 /* pinctl values */
 #define PIN_IN			(AC_PINCTL_IN_EN)
-#define PIN_VREFHIZ	(AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ)
+#define PIN_VREFHIZ		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ)
 #define PIN_VREF50		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_50)
-#define PIN_VREFGRD	(AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD)
+#define PIN_VREFGRD		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD)
 #define PIN_VREF80		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_80)
-#define PIN_VREF100	(AC_PINCTL_IN_EN | AC_PINCTL_VREF_100)
-#define PIN_OUT		(AC_PINCTL_OUT_EN)
+#define PIN_VREF100		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_100)
+#define PIN_OUT			(AC_PINCTL_OUT_EN)
 #define PIN_HP			(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
 #define PIN_HP_AMP		(AC_PINCTL_HP_EN)
 
@@ -393,10 +401,26 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps);
 
+int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl);
+void snd_hda_ctls_clear(struct hda_codec *codec);
+
 /*
  * hwdep interface
  */
+#ifdef CONFIG_SND_HDA_HWDEP
 int snd_hda_create_hwdep(struct hda_codec *codec);
+#else
+static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; }
+#endif
+
+#ifdef CONFIG_SND_HDA_RECONFIG
+int snd_hda_hwdep_add_sysfs(struct hda_codec *codec);
+#else
+static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
+{
+	return 0;
+}
+#endif
 
 /*
  * power-management
@@ -430,4 +454,66 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 #define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
 #define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
 
+/*
+ * CEA Short Audio Descriptor data
+ */
+struct cea_sad {
+	int	channels;
+	int	format;		/* (format == 0) indicates invalid SAD */
+	int	rates;
+	int	sample_bits;	/* for LPCM */
+	int	max_bitrate;	/* for AC3...ATRAC */
+	int	profile;	/* for WMAPRO */
+};
+
+#define ELD_FIXED_BYTES	20
+#define ELD_MAX_MNL	16
+#define ELD_MAX_SAD	16
+
+/*
+ * ELD: EDID Like Data
+ */
+struct hdmi_eld {
+	int	eld_size;
+	int	baseline_len;
+	int	eld_ver;	/* (eld_ver == 0) indicates invalid ELD */
+	int	cea_edid_ver;
+	char	monitor_name[ELD_MAX_MNL + 1];
+	int	manufacture_id;
+	int	product_id;
+	u64	port_id;
+	int	support_hdcp;
+	int	support_ai;
+	int	conn_type;
+	int	aud_synch_delay;
+	int	spk_alloc;
+	int	sad_count;
+	struct cea_sad sad[ELD_MAX_SAD];
+#ifdef CONFIG_PROC_FS
+	struct snd_info_entry *proc_entry;
+#endif
+};
+
+int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
+int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
+void snd_hdmi_show_eld(struct hdmi_eld *eld);
+
+#ifdef CONFIG_PROC_FS
+int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld);
+void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld);
+#else
+static inline int snd_hda_eld_proc_new(struct hda_codec *codec,
+				       struct hdmi_eld *eld)
+{
+	return 0;
+}
+static inline void snd_hda_eld_proc_free(struct hda_codec *codec,
+					 struct hdmi_eld *eld)
+{
+}
+#endif
+
+#define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80
+void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen);
+
 #endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
deleted file mode 100644
index dfbcfa88da44..000000000000
--- a/sound/pci/hda/hda_patch.h
+++ /dev/null
@@ -1,22 +0,0 @@
-/*
- * HDA Patches - included by hda_codec.c
- */
-
-/* Realtek codecs */
-extern struct hda_codec_preset snd_hda_preset_realtek[];
-/* C-Media codecs */
-extern struct hda_codec_preset snd_hda_preset_cmedia[];
-/* Analog Devices codecs */
-extern struct hda_codec_preset snd_hda_preset_analog[];
-/* SigmaTel codecs */
-extern struct hda_codec_preset snd_hda_preset_sigmatel[];
-/* SiLabs 3054/3055 modem codecs */
-extern struct hda_codec_preset snd_hda_preset_si3054[];
-/* ATI HDMI codecs */
-extern struct hda_codec_preset snd_hda_preset_atihdmi[];
-/* Conexant audio codec */
-extern struct hda_codec_preset snd_hda_preset_conexant[];
-/* VIA codecs */
-extern struct hda_codec_preset snd_hda_preset_via[];
-/* NVIDIA HDMI codecs */
-extern struct hda_codec_preset snd_hda_preset_nvhdmi[];
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index c39af986bff1..7ca66d654148 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -91,31 +91,21 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
 
 static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm)
 {
-	static unsigned int rates[] = {
-		8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
-		96000, 176400, 192000, 384000
-	};
-	int i;
+	char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
 
 	pcm &= AC_SUPPCM_RATES;
 	snd_iprintf(buffer, "    rates [0x%x]:", pcm);
-	for (i = 0; i < ARRAY_SIZE(rates); i++) 
-		if (pcm & (1 << i))
-			snd_iprintf(buffer, " %d", rates[i]);
-	snd_iprintf(buffer, "\n");
+	snd_print_pcm_rates(pcm, buf, sizeof(buf));
+	snd_iprintf(buffer, "%s\n", buf);
 }
 
 static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm)
 {
-	static unsigned int bits[] = { 8, 16, 20, 24, 32 };
-	int i;
+	char buf[SND_PRINT_BITS_ADVISED_BUFSIZE];
 
-	pcm = (pcm >> 16) & 0xff;
-	snd_iprintf(buffer, "    bits [0x%x]:", pcm);
-	for (i = 0; i < ARRAY_SIZE(bits); i++)
-		if (pcm & (1 << i))
-			snd_iprintf(buffer, " %d", bits[i]);
-	snd_iprintf(buffer, "\n");
+	snd_iprintf(buffer, "    bits [0x%x]:", (pcm >> 16) & 0xff);
+	snd_print_pcm_bits(pcm, buf, sizeof(buf));
+	snd_iprintf(buffer, "%s\n", buf);
 }
 
 static void print_pcm_formats(struct snd_info_buffer *buffer,
@@ -145,32 +135,6 @@ static void print_pcm_caps(struct snd_info_buffer *buffer,
 	print_pcm_formats(buffer, stream);
 }
 
-static const char *get_jack_location(u32 cfg)
-{
-	static char *bases[7] = {
-		"N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom",
-	};
-	static unsigned char specials_idx[] = {
-		0x07, 0x08,
-		0x17, 0x18, 0x19,
-		0x37, 0x38
-	};
-	static char *specials[] = {
-		"Rear Panel", "Drive Bar",
-		"Riser", "HDMI", "ATAPI",
-		"Mobile-In", "Mobile-Out"
-	};
-	int i;
-	cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT;
-	if ((cfg & 0x0f) < 7)
-		return bases[cfg & 0x0f];
-	for (i = 0; i < ARRAY_SIZE(specials_idx); i++) {
-		if (cfg == specials_idx[i])
-			return specials[i];
-	}
-	return "UNKNOWN";
-}
-
 static const char *get_jack_connection(u32 cfg)
 {
 	static char *names[16] = {
@@ -206,13 +170,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
 			   int *supports_vref)
 {
 	static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" };
-	static char *jack_types[16] = {
-		"Line Out", "Speaker", "HP Out", "CD",
-		"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
-		"Line In", "Aux", "Mic", "Telephony",
-		"SPDIF In", "Digitial In", "Reserved", "Other"
-	};
-	static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" };
 	unsigned int caps, val;
 
 	caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
@@ -274,9 +231,9 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
 	caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
 	snd_iprintf(buffer, "  Pin Default 0x%08x: [%s] %s at %s %s\n", caps,
 		    jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT],
-		    jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT],
-		    jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3],
-		    get_jack_location(caps));
+		    snd_hda_get_jack_type(caps),
+		    snd_hda_get_jack_connectivity(caps),
+		    snd_hda_get_jack_location(caps));
 	snd_iprintf(buffer, "    Conn = %s, Color = %s\n",
 		    get_jack_connection(caps),
 		    get_jack_color(caps));
@@ -457,17 +414,6 @@ static void print_conn_list(struct snd_info_buffer *buffer,
 	}
 }
 
-static void print_realtek_coef(struct snd_info_buffer *buffer,
-			       struct hda_codec *codec, hda_nid_t nid)
-{
-	int coeff = snd_hda_codec_read(codec, nid, 0,
-				       AC_VERB_GET_PROC_COEF, 0);
-	snd_iprintf(buffer, "  Processing Coefficient: 0x%02x\n", coeff);
-	coeff = snd_hda_codec_read(codec, nid, 0,
-				   AC_VERB_GET_COEF_INDEX, 0);
-	snd_iprintf(buffer, "  Coefficient Index: 0x%02x\n", coeff);
-}
-
 static void print_gpio(struct snd_info_buffer *buffer,
 		       struct hda_codec *codec, hda_nid_t nid)
 {
@@ -500,12 +446,13 @@ static void print_gpio(struct snd_info_buffer *buffer,
 	for (i = 0; i < max; ++i)
 		snd_iprintf(buffer,
 			    "  IO[%d]: enable=%d, dir=%d, wake=%d, "
-			    "sticky=%d, data=%d\n", i,
+			    "sticky=%d, data=%d, unsol=%d\n", i,
 			    (enable & (1<<i)) ? 1 : 0,
 			    (direction & (1<<i)) ? 1 : 0,
 			    (wake & (1<<i)) ? 1 : 0,
 			    (sticky & (1<<i)) ? 1 : 0,
-			    (data & (1<<i)) ? 1 : 0);
+			    (data & (1<<i)) ? 1 : 0,
+			    (unsol & (1<<i)) ? 1 : 0);
 	/* FIXME: add GPO and GPI pin information */
 }
 
@@ -513,12 +460,11 @@ static void print_codec_info(struct snd_info_entry *entry,
 			     struct snd_info_buffer *buffer)
 {
 	struct hda_codec *codec = entry->private_data;
-	char buf[32];
 	hda_nid_t nid;
 	int i, nodes;
 
-	snd_hda_get_codec_name(codec, buf, sizeof(buf));
-	snd_iprintf(buffer, "Codec: %s\n", buf);
+	snd_iprintf(buffer, "Codec: %s\n",
+		    codec->name ? codec->name : "Not Set");
 	snd_iprintf(buffer, "Address: %d\n", codec->addr);
 	snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
 	snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
@@ -547,6 +493,8 @@ static void print_codec_info(struct snd_info_entry *entry,
 	}
 
 	print_gpio(buffer, codec, codec->afg);
+	if (codec->proc_widget_hook)
+		codec->proc_widget_hook(buffer, codec, codec->afg);
 
 	for (i = 0; i < nodes; i++, nid++) {
 		unsigned int wid_caps =
@@ -649,9 +597,8 @@ static void print_codec_info(struct snd_info_entry *entry,
 		if (wid_caps & AC_WCAP_PROC_WID)
 			print_proc_caps(buffer, codec, nid);
 
-		/* NID 0x20 == Realtek Define Registers */
-		if (codec->vendor_id == 0x10ec && nid == 0x20)
-			print_realtek_coef(buffer, codec, nid);
+		if (codec->proc_widget_hook)
+			codec->proc_widget_hook(buffer, codec, nid);
 	}
 	snd_hda_power_down(codec);
 }
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 686c77491dea..26247cfe749d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -27,7 +27,6 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 
 struct ad198x_spec {
 	struct snd_kcontrol_new *mixers[5];
@@ -67,8 +66,7 @@ struct ad198x_spec {
 
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
-	unsigned int num_kctl_alloc, num_kctl_used;
-	struct snd_kcontrol_new *kctl_alloc;
+	struct snd_array kctls;
 	struct hda_input_mux private_imux;
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
@@ -154,6 +152,8 @@ static const char *ad_slave_sws[] = {
 	NULL
 };
 
+static void ad198x_free_kctls(struct hda_codec *codec);
+
 static int ad198x_build_controls(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -202,6 +202,7 @@ static int ad198x_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	ad198x_free_kctls(codec); /* no longer needed */
 	return 0;
 }
 
@@ -375,16 +376,27 @@ static int ad198x_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+static void ad198x_free_kctls(struct hda_codec *codec)
+{
+	struct ad198x_spec *spec = codec->spec;
+
+	if (spec->kctls.list) {
+		struct snd_kcontrol_new *kctl = spec->kctls.list;
+		int i;
+		for (i = 0; i < spec->kctls.used; i++)
+			kfree(kctl[i].name);
+	}
+	snd_array_free(&spec->kctls);
+}
+
 static void ad198x_free(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
-	unsigned int i;
 
-	if (spec->kctl_alloc) {
-		for (i = 0; i < spec->num_kctl_used; i++)
-			kfree(spec->kctl_alloc[i].name);
-		kfree(spec->kctl_alloc);
-	}
+	if (!spec)
+		return;
+
+	ad198x_free_kctls(codec);
 	kfree(codec->spec);
 }
 
@@ -625,6 +637,36 @@ static struct hda_input_mux ad1986a_automic_capture_source = {
 };
 
 static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
+	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
+	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = ad198x_mux_enum_info,
+		.get = ad198x_mux_enum_get,
+		.put = ad198x_mux_enum_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "External Amplifier",
+		.info = ad198x_eapd_info,
+		.get = ad198x_eapd_get,
+		.put = ad198x_eapd_put,
+		.private_value = 0x1b | (1 << 8), /* port-D, inversed */
+	},
+	{ } /* end */
+};
+
+static struct snd_kcontrol_new ad1986a_samsung_mixers[] = {
 	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -917,6 +959,7 @@ enum {
 	AD1986A_LAPTOP_EAPD,
 	AD1986A_LAPTOP_AUTOMUTE,
 	AD1986A_ULTRA,
+	AD1986A_SAMSUNG,
 	AD1986A_MODELS
 };
 
@@ -927,6 +970,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = {
 	[AD1986A_LAPTOP_EAPD]	= "laptop-eapd",
 	[AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
 	[AD1986A_ULTRA]		= "ultra",
+	[AD1986A_SAMSUNG]	= "samsung",
 };
 
 static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
@@ -949,9 +993,9 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
 	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_LAPTOP_EAPD),
+	SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG),
+	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG),
+	SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG),
 	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
 	SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
 	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
@@ -1033,6 +1077,17 @@ static int patch_ad1986a(struct hda_codec *codec)
 		break;
 	case AD1986A_LAPTOP_EAPD:
 		spec->mixers[0] = ad1986a_laptop_eapd_mixers;
+		spec->num_init_verbs = 2;
+		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+		spec->multiout.max_channels = 2;
+		spec->multiout.num_dacs = 1;
+		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+		if (!is_jack_available(codec, 0x25))
+			spec->multiout.dig_out_nid = 0;
+		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
+		break;
+	case AD1986A_SAMSUNG:
+		spec->mixers[0] = ad1986a_samsung_mixers;
 		spec->num_init_verbs = 3;
 		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
 		spec->init_verbs[2] = ad1986a_automic_verbs;
@@ -2452,9 +2507,6 @@ static struct hda_amp_list ad1988_loopbacks[] = {
  * Automatic parse of I/O pins from the BIOS configuration
  */
 
-#define NUM_CONTROL_ALLOC	32
-#define NUM_VERB_ALLOC		32
-
 enum {
 	AD_CTL_WIDGET_VOL,
 	AD_CTL_WIDGET_MUTE,
@@ -2472,27 +2524,15 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name,
 {
 	struct snd_kcontrol_new *knew;
 
-	if (spec->num_kctl_used >= spec->num_kctl_alloc) {
-		int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
-
-		knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
-		if (! knew)
-			return -ENOMEM;
-		if (spec->kctl_alloc) {
-			memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
-			kfree(spec->kctl_alloc);
-		}
-		spec->kctl_alloc = knew;
-		spec->num_kctl_alloc = num;
-	}
-
-	knew = &spec->kctl_alloc[spec->num_kctl_used];
+	snd_array_init(&spec->kctls, sizeof(*knew), 32);
+	knew = snd_array_new(&spec->kctls);
+	if (!knew)
+		return -ENOMEM;
 	*knew = ad1988_control_templates[type];
 	knew->name = kstrdup(name, GFP_KERNEL);
 	if (! knew->name)
 		return -ENOMEM;
 	knew->private_value = val;
-	spec->num_kctl_used++;
 	return 0;
 }
 
@@ -2846,8 +2886,8 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = AD1988_SPDIF_IN;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->init_verbs[spec->num_init_verbs++] = ad1988_6stack_init_verbs;
 
@@ -3861,6 +3901,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
@@ -4267,7 +4308,7 @@ static int patch_ad1882(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_analog[] = {
+static struct hda_codec_preset snd_hda_preset_analog[] = {
 	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
 	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
 	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
@@ -4285,3 +4326,26 @@ struct hda_codec_preset snd_hda_preset_analog[] = {
 	{ .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:11d4*");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Analog Devices HD-audio codec");
+
+static struct hda_codec_preset_list analog_list = {
+	.preset = snd_hda_preset_analog,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_analog_init(void)
+{
+	return snd_hda_add_codec_preset(&analog_list);
+}
+
+static void __exit patch_analog_exit(void)
+{
+	snd_hda_delete_codec_preset(&analog_list);
+}
+
+module_init(patch_analog_init)
+module_exit(patch_analog_exit)
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index ba61575983fd..233e4778bba9 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -27,7 +27,6 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 
 struct atihdmi_spec {
 	struct hda_multi_out multiout;
@@ -187,13 +186,40 @@ static int patch_atihdmi(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_atihdmi[] = {
-	{ .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
-	{ .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
-	{ .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi },
-	{ .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi },
+static struct hda_codec_preset snd_hda_preset_atihdmi[] = {
+	{ .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi },
+	{ .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi },
+	{ .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi },
+	{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi },
 	{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi },
-	{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi },
 	{ .id = 0x17e80047, .name = "Chrontel HDMI",  .patch = patch_atihdmi },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:1002793c");
+MODULE_ALIAS("snd-hda-codec-id:10027919");
+MODULE_ALIAS("snd-hda-codec-id:1002791a");
+MODULE_ALIAS("snd-hda-codec-id:1002aa01");
+MODULE_ALIAS("snd-hda-codec-id:10951390");
+MODULE_ALIAS("snd-hda-codec-id:17e80047");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ATI HDMI HD-audio codec");
+
+static struct hda_codec_preset_list atihdmi_list = {
+	.preset = snd_hda_preset_atihdmi,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_atihdmi_init(void)
+{
+	return snd_hda_add_codec_preset(&atihdmi_list);
+}
+
+static void __exit patch_atihdmi_exit(void)
+{
+	snd_hda_delete_codec_preset(&atihdmi_list);
+}
+
+module_init(patch_atihdmi_init)
+module_exit(patch_atihdmi_exit)
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 6ef57fbfb6eb..f3ebe837f2d5 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -28,7 +28,6 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 #define NUM_PINS	11
 
 
@@ -736,8 +735,32 @@ static int patch_cmi9880(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_cmedia[] = {
+static struct hda_codec_preset snd_hda_preset_cmedia[] = {
 	{ .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
  	{ .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:13f69880");
+MODULE_ALIAS("snd-hda-codec-id:434d4980");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("C-Media HD-audio codec");
+
+static struct hda_codec_preset_list cmedia_list = {
+	.preset = snd_hda_preset_cmedia,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_cmedia_init(void)
+{
+	return snd_hda_add_codec_preset(&cmedia_list);
+}
+
+static void __exit patch_cmedia_exit(void)
+{
+	snd_hda_delete_codec_preset(&cmedia_list);
+}
+
+module_init(patch_cmedia_init)
+module_exit(patch_cmedia_exit)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7c1eb23f0cec..b20e1cede00b 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -27,7 +27,6 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 
 #define CXT_PIN_DIR_IN              0x00
 #define CXT_PIN_DIR_OUT             0x01
@@ -86,8 +85,6 @@ struct conexant_spec {
 
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
-	unsigned int num_kctl_alloc, num_kctl_used;
-	struct snd_kcontrol_new *kctl_alloc;
 	struct hda_input_mux private_imux;
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
@@ -344,15 +341,6 @@ static int conexant_init(struct hda_codec *codec)
 
 static void conexant_free(struct hda_codec *codec)
 {
-        struct conexant_spec *spec = codec->spec;
-        unsigned int i;
-
-        if (spec->kctl_alloc) {
-                for (i = 0; i < spec->num_kctl_used; i++)
-                        kfree(spec->kctl_alloc[i].name);
-                kfree(spec->kctl_alloc);
-        }
-
 	kfree(codec->spec);
 }
 
@@ -1782,7 +1770,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 /*
  */
 
-struct hda_codec_preset snd_hda_preset_conexant[] = {
+static struct hda_codec_preset snd_hda_preset_conexant[] = {
 	{ .id = 0x14f15045, .name = "CX20549 (Venice)",
 	  .patch = patch_cxt5045 },
 	{ .id = 0x14f15047, .name = "CX20551 (Waikiki)",
@@ -1791,3 +1779,28 @@ struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_cxt5051 },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:14f15045");
+MODULE_ALIAS("snd-hda-codec-id:14f15047");
+MODULE_ALIAS("snd-hda-codec-id:14f15051");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Conexant HD-audio codec");
+
+static struct hda_codec_preset_list conexant_list = {
+	.preset = snd_hda_preset_conexant,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_conexant_init(void)
+{
+	return snd_hda_add_codec_preset(&conexant_list);
+}
+
+static void __exit patch_conexant_exit(void)
+{
+	snd_hda_delete_codec_preset(&conexant_list);
+}
+
+module_init(patch_conexant_init)
+module_exit(patch_conexant_exit)
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
new file mode 100644
index 000000000000..3564f4e4b74c
--- /dev/null
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -0,0 +1,711 @@
+/*
+ *
+ *  patch_intelhdmi.c - Patch for Intel HDMI codecs
+ *
+ *  Copyright(c) 2008 Intel Corporation. All rights reserved.
+ *
+ *  Authors:
+ *  			Jiang Zhe <zhe.jiang@intel.com>
+ *  			Wu Fengguang <wfg@linux.intel.com>
+ *
+ *  Maintained by:
+ *  			Wu Fengguang <wfg@linux.intel.com>
+ *
+ *  This program is free software; you can redistribute it and/or modify it
+ *  under the terms of the GNU General Public License as published by the Free
+ *  Software Foundation; either version 2 of the License, or (at your option)
+ *  any later version.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
+ *  or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License
+ *  for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software Foundation,
+ *  Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+#define CVT_NID		0x02	/* audio converter */
+#define PIN_NID		0x03	/* HDMI output pin */
+
+#define INTEL_HDMI_EVENT_TAG		0x08
+
+struct intel_hdmi_spec {
+	struct hda_multi_out multiout;
+	struct hda_pcm pcm_rec;
+	struct hdmi_eld sink_eld;
+};
+
+static struct hda_verb pinout_enable_verb[] = {
+	{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{} /* terminator */
+};
+
+static struct hda_verb pinout_disable_verb[] = {
+	{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00},
+	{}
+};
+
+static struct hda_verb unsolicited_response_verb[] = {
+	{PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
+						  INTEL_HDMI_EVENT_TAG},
+	{}
+};
+
+static struct hda_verb def_chan_map[] = {
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66},
+	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77},
+	{}
+};
+
+
+struct hdmi_audio_infoframe {
+	u8 type; /* 0x84 */
+	u8 ver;  /* 0x01 */
+	u8 len;  /* 0x0a */
+
+	u8 checksum;	/* PB0 */
+	u8 CC02_CT47;	/* CC in bits 0:2, CT in 4:7 */
+	u8 SS01_SF24;
+	u8 CXT04;
+	u8 CA;
+	u8 LFEPBL01_LSV36_DM_INH7;
+	u8 reserved[5];	/* PB6 - PB10 */
+};
+
+/*
+ * CEA speaker placement:
+ *
+ *        FLH       FCH        FRH
+ *  FLW    FL  FLC   FC   FRC   FR   FRW
+ *
+ *                                  LFE
+ *                     TC
+ *
+ *          RL  RLC   RC   RRC   RR
+ *
+ * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to
+ * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC.
+ */
+enum cea_speaker_placement {
+	FL  = (1 <<  0),	/* Front Left           */
+	FC  = (1 <<  1),	/* Front Center         */
+	FR  = (1 <<  2),	/* Front Right          */
+	FLC = (1 <<  3),	/* Front Left Center    */
+	FRC = (1 <<  4),	/* Front Right Center   */
+	RL  = (1 <<  5),	/* Rear Left            */
+	RC  = (1 <<  6),	/* Rear Center          */
+	RR  = (1 <<  7),	/* Rear Right           */
+	RLC = (1 <<  8),	/* Rear Left Center     */
+	RRC = (1 <<  9),	/* Rear Right Center    */
+	LFE = (1 << 10),	/* Low Frequency Effect */
+	FLW = (1 << 11),	/* Front Left Wide      */
+	FRW = (1 << 12),	/* Front Right Wide     */
+	FLH = (1 << 13),	/* Front Left High      */
+	FCH = (1 << 14),	/* Front Center High    */
+	FRH = (1 << 15),	/* Front Right High     */
+	TC  = (1 << 16),	/* Top Center           */
+};
+
+/*
+ * ELD SA bits in the CEA Speaker Allocation data block
+ */
+static int eld_speaker_allocation_bits[] = {
+	[0] = FL | FR,
+	[1] = LFE,
+	[2] = FC,
+	[3] = RL | RR,
+	[4] = RC,
+	[5] = FLC | FRC,
+	[6] = RLC | RRC,
+	/* the following are not defined in ELD yet */
+	[7] = FLW | FRW,
+	[8] = FLH | FRH,
+	[9] = TC,
+	[10] = FCH,
+};
+
+struct cea_channel_speaker_allocation {
+	int ca_index;
+	int speakers[8];
+
+	/* derived values, just for convenience */
+	int channels;
+	int spk_mask;
+};
+
+/*
+ * This is an ordered list!
+ *
+ * The preceding ones have better chances to be selected by
+ * hdmi_setup_channel_allocation().
+ */
+static struct cea_channel_speaker_allocation channel_allocations[] = {
+/* 			  channel:   8     7    6    5    4     3    2    1  */
+{ .ca_index = 0x00,  .speakers = {   0,    0,   0,   0,   0,    0,  FR,  FL } },
+				 /* 2.1 */
+{ .ca_index = 0x01,  .speakers = {   0,    0,   0,   0,   0,  LFE,  FR,  FL } },
+				 /* Dolby Surround */
+{ .ca_index = 0x02,  .speakers = {   0,    0,   0,   0,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x03,  .speakers = {   0,    0,   0,   0,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x04,  .speakers = {   0,    0,   0,  RC,   0,    0,  FR,  FL } },
+{ .ca_index = 0x05,  .speakers = {   0,    0,   0,  RC,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x06,  .speakers = {   0,    0,   0,  RC,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x07,  .speakers = {   0,    0,   0,  RC,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x08,  .speakers = {   0,    0,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x09,  .speakers = {   0,    0,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x0a,  .speakers = {   0,    0,  RR,  RL,  FC,    0,  FR,  FL } },
+				 /* 5.1 */
+{ .ca_index = 0x0b,  .speakers = {   0,    0,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x0c,  .speakers = {   0,   RC,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x0d,  .speakers = {   0,   RC,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x0e,  .speakers = {   0,   RC,  RR,  RL,  FC,    0,  FR,  FL } },
+				 /* 6.1 */
+{ .ca_index = 0x0f,  .speakers = {   0,   RC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x10,  .speakers = { RRC,  RLC,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x11,  .speakers = { RRC,  RLC,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x12,  .speakers = { RRC,  RLC,  RR,  RL,  FC,    0,  FR,  FL } },
+				 /* 7.1 */
+{ .ca_index = 0x13,  .speakers = { RRC,  RLC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x14,  .speakers = { FRC,  FLC,   0,   0,   0,    0,  FR,  FL } },
+{ .ca_index = 0x15,  .speakers = { FRC,  FLC,   0,   0,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x16,  .speakers = { FRC,  FLC,   0,   0,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x17,  .speakers = { FRC,  FLC,   0,   0,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x18,  .speakers = { FRC,  FLC,   0,  RC,   0,    0,  FR,  FL } },
+{ .ca_index = 0x19,  .speakers = { FRC,  FLC,   0,  RC,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x1a,  .speakers = { FRC,  FLC,   0,  RC,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x1b,  .speakers = { FRC,  FLC,   0,  RC,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x1c,  .speakers = { FRC,  FLC,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x1d,  .speakers = { FRC,  FLC,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x1e,  .speakers = { FRC,  FLC,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x1f,  .speakers = { FRC,  FLC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x20,  .speakers = {   0,  FCH,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x21,  .speakers = {   0,  FCH,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x22,  .speakers = {  TC,    0,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x23,  .speakers = {  TC,    0,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x24,  .speakers = { FRH,  FLH,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x25,  .speakers = { FRH,  FLH,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x26,  .speakers = { FRW,  FLW,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x27,  .speakers = { FRW,  FLW,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x28,  .speakers = {  TC,   RC,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x29,  .speakers = {  TC,   RC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x2a,  .speakers = { FCH,   RC,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x2b,  .speakers = { FCH,   RC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x2c,  .speakers = {  TC,  FCH,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x2d,  .speakers = {  TC,  FCH,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x2e,  .speakers = { FRH,  FLH,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x2f,  .speakers = { FRH,  FLH,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x30,  .speakers = { FRW,  FLW,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x31,  .speakers = { FRW,  FLW,  RR,  RL,  FC,  LFE,  FR,  FL } },
+};
+
+/*
+ * HDMI routines
+ */
+
+#ifdef BE_PARANOID
+static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid,
+				int *packet_index, int *byte_index)
+{
+	int val;
+
+	val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_INDEX, 0);
+
+	*packet_index = val >> 5;
+	*byte_index = val & 0x1f;
+}
+#endif
+
+static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t nid,
+				int packet_index, int byte_index)
+{
+	int val;
+
+	val = (packet_index << 5) | (byte_index & 0x1f);
+
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val);
+}
+
+static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
+				unsigned char val)
+{
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val);
+}
+
+static void hdmi_enable_output(struct hda_codec *codec)
+{
+	/* Enable Audio InfoFrame Transmission */
+	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+						AC_DIPXMIT_BEST);
+	/* Unmute */
+	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
+		snd_hda_codec_write(codec, PIN_NID, 0,
+				AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+	/* Enable pin out */
+	snd_hda_sequence_write(codec, pinout_enable_verb);
+}
+
+static void hdmi_disable_output(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, pinout_disable_verb);
+	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
+		snd_hda_codec_write(codec, PIN_NID, 0,
+				AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	/*
+	 * FIXME: noises may arise when playing music after reloading the
+	 * kernel module, until the next X restart or monitor repower.
+	 */
+}
+
+static int hdmi_get_channel_count(struct hda_codec *codec)
+{
+	return 1 + snd_hda_codec_read(codec, CVT_NID, 0,
+					AC_VERB_GET_CVT_CHAN_COUNT, 0);
+}
+
+static void hdmi_set_channel_count(struct hda_codec *codec, int chs)
+{
+	snd_hda_codec_write(codec, CVT_NID, 0,
+					AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
+
+	if (chs != hdmi_get_channel_count(codec))
+		snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n",
+					chs, hdmi_get_channel_count(codec));
+}
+
+static void hdmi_debug_channel_mapping(struct hda_codec *codec)
+{
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+	int i;
+	int slot;
+
+	for (i = 0; i < 8; i++) {
+		slot = snd_hda_codec_read(codec, CVT_NID, 0,
+						AC_VERB_GET_HDMI_CHAN_SLOT, i);
+		printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n",
+						slot >> 4, slot & 0x7);
+	}
+#endif
+}
+
+static void hdmi_parse_eld(struct hda_codec *codec)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+	struct hdmi_eld *eld = &spec->sink_eld;
+
+	if (!snd_hdmi_get_eld(eld, codec, PIN_NID))
+		snd_hdmi_show_eld(eld);
+}
+
+
+/*
+ * Audio InfoFrame routines
+ */
+
+static void hdmi_debug_dip_size(struct hda_codec *codec)
+{
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+	int i;
+	int size;
+
+	size = snd_hdmi_get_eld_size(codec, PIN_NID);
+	printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size);
+
+	for (i = 0; i < 8; i++) {
+		size = snd_hda_codec_read(codec, PIN_NID, 0,
+						AC_VERB_GET_HDMI_DIP_SIZE, i);
+		printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size);
+	}
+#endif
+}
+
+static void hdmi_clear_dip_buffers(struct hda_codec *codec)
+{
+#ifdef BE_PARANOID
+	int i, j;
+	int size;
+	int pi, bi;
+	for (i = 0; i < 8; i++) {
+		size = snd_hda_codec_read(codec, PIN_NID, 0,
+						AC_VERB_GET_HDMI_DIP_SIZE, i);
+		if (size == 0)
+			continue;
+
+		hdmi_set_dip_index(codec, PIN_NID, i, 0x0);
+		for (j = 1; j < 1000; j++) {
+			hdmi_write_dip_byte(codec, PIN_NID, 0x0);
+			hdmi_get_dip_index(codec, PIN_NID, &pi, &bi);
+			if (pi != i)
+				snd_printd(KERN_INFO "dip index %d: %d != %d\n",
+						bi, pi, i);
+			if (bi == 0) /* byte index wrapped around */
+				break;
+		}
+		snd_printd(KERN_INFO
+			"HDMI: DIP GP[%d] buf reported size=%d, written=%d\n",
+			i, size, j);
+	}
+#endif
+}
+
+static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
+					struct hdmi_audio_infoframe *ai)
+{
+	u8 *params = (u8 *)ai;
+	int i;
+
+	hdmi_debug_dip_size(codec);
+	hdmi_clear_dip_buffers(codec); /* be paranoid */
+
+	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	for (i = 0; i < sizeof(ai); i++)
+		hdmi_write_dip_byte(codec, PIN_NID, params[i]);
+}
+
+/*
+ * Compute derived values in channel_allocations[].
+ */
+static void init_channel_allocations(void)
+{
+	int i, j;
+	struct cea_channel_speaker_allocation *p;
+
+	for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+		p = channel_allocations + i;
+		p->channels = 0;
+		p->spk_mask = 0;
+		for (j = 0; j < ARRAY_SIZE(p->speakers); j++)
+			if (p->speakers[j]) {
+				p->channels++;
+				p->spk_mask |= p->speakers[j];
+			}
+	}
+}
+
+/*
+ * The transformation takes two steps:
+ *
+ * 	eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask
+ * 	      spk_mask => (channel_allocations[])         => ai->CA
+ *
+ * TODO: it could select the wrong CA from multiple candidates.
+*/
+static int hdmi_setup_channel_allocation(struct hda_codec *codec,
+					 struct hdmi_audio_infoframe *ai)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+	struct hdmi_eld *eld = &spec->sink_eld;
+	int i;
+	int spk_mask = 0;
+	int channels = 1 + (ai->CC02_CT47 & 0x7);
+	char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
+
+	/*
+	 * CA defaults to 0 for basic stereo audio
+	 */
+	if (!eld->eld_ver)
+		return 0;
+	if (!eld->spk_alloc)
+		return 0;
+	if (channels <= 2)
+		return 0;
+
+	/*
+	 * expand ELD's speaker allocation mask
+	 *
+	 * ELD tells the speaker mask in a compact(paired) form,
+	 * expand ELD's notions to match the ones used by Audio InfoFrame.
+	 */
+	for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
+		if (eld->spk_alloc & (1 << i))
+			spk_mask |= eld_speaker_allocation_bits[i];
+	}
+
+	/* search for the first working match in the CA table */
+	for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+		if (channels == channel_allocations[i].channels &&
+		    (spk_mask & channel_allocations[i].spk_mask) ==
+				channel_allocations[i].spk_mask) {
+			ai->CA = channel_allocations[i].ca_index;
+			break;
+		}
+	}
+
+	snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf));
+	snd_printdd(KERN_INFO
+			"HDMI: select CA 0x%x for %d-channel allocation: %s\n",
+			ai->CA, channels, buf);
+
+	return ai->CA;
+}
+
+static void hdmi_setup_channel_mapping(struct hda_codec *codec,
+					struct hdmi_audio_infoframe *ai)
+{
+	if (!ai->CA)
+		return;
+
+	/*
+	 * TODO: adjust channel mapping if necessary
+	 * ALSA sequence is front/surr/clfe/side?
+	 */
+
+	snd_hda_sequence_write(codec, def_chan_map);
+	hdmi_debug_channel_mapping(codec);
+}
+
+
+static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
+{
+	struct hdmi_audio_infoframe ai = {
+		.type		= 0x84,
+		.ver		= 0x01,
+		.len		= 0x0a,
+		.CC02_CT47	= substream->runtime->channels - 1,
+	};
+
+	hdmi_setup_channel_allocation(codec, &ai);
+	hdmi_setup_channel_mapping(codec, &ai);
+
+	hdmi_fill_audio_infoframe(codec, &ai);
+}
+
+
+/*
+ * Unsolicited events
+ */
+
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+{
+	int pind = !!(res & AC_UNSOL_RES_PD);
+	int eldv = !!(res & AC_UNSOL_RES_ELDV);
+
+	printk(KERN_INFO
+		"HDMI hot plug event: Presence_Detect=%d ELD_Valid=%d\n",
+		pind, eldv);
+
+	if (pind && eldv) {
+		hdmi_parse_eld(codec);
+		/* TODO: do real things about ELD */
+	}
+}
+
+static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
+{
+	int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
+	int cp_state = !!(res & AC_UNSOL_RES_CP_STATE);
+	int cp_ready = !!(res & AC_UNSOL_RES_CP_READY);
+
+	printk(KERN_INFO
+		"HDMI content protection event: SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+		subtag,
+		cp_state,
+		cp_ready);
+
+	/* TODO */
+	if (cp_state)
+		;
+	if (cp_ready)
+		;
+}
+
+
+static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
+	int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
+
+	if (tag != INTEL_HDMI_EVENT_TAG) {
+		snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag);
+		return;
+	}
+
+	if (subtag == 0)
+		hdmi_intrinsic_event(codec, res);
+	else
+		hdmi_non_intrinsic_event(codec, res);
+}
+
+/*
+ * Callbacks
+ */
+
+static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 struct snd_pcm_substream *substream)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+
+	hdmi_disable_output(codec);
+
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+					   struct hda_codec *codec,
+					   unsigned int stream_tag,
+					   unsigned int format,
+					   struct snd_pcm_substream *substream)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+
+	snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+					     format, substream);
+
+	hdmi_set_channel_count(codec, substream->runtime->channels);
+
+	hdmi_setup_audio_infoframe(codec, substream);
+
+	hdmi_enable_output(codec);
+
+	return 0;
+}
+
+static struct hda_pcm_stream intel_hdmi_pcm_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = CVT_NID, /* NID to query formats and rates and setup streams */
+	.ops = {
+		.open    = intel_hdmi_playback_pcm_open,
+		.close   = intel_hdmi_playback_pcm_close,
+		.prepare = intel_hdmi_playback_pcm_prepare
+	},
+};
+
+static int intel_hdmi_build_pcms(struct hda_codec *codec)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+	struct hda_pcm *info = &spec->pcm_rec;
+
+	codec->num_pcms = 1;
+	codec->pcm_info = info;
+
+	info->name = "INTEL HDMI";
+	info->pcm_type = HDA_PCM_TYPE_HDMI;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback;
+
+	return 0;
+}
+
+static int intel_hdmi_build_controls(struct hda_codec *codec)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static int intel_hdmi_init(struct hda_codec *codec)
+{
+	/* disable audio output as early as possible */
+	hdmi_disable_output(codec);
+
+	snd_hda_sequence_write(codec, unsolicited_response_verb);
+
+	return 0;
+}
+
+static void intel_hdmi_free(struct hda_codec *codec)
+{
+	struct intel_hdmi_spec *spec = codec->spec;
+
+	snd_hda_eld_proc_free(codec, &spec->sink_eld);
+	kfree(spec);
+}
+
+static struct hda_codec_ops intel_hdmi_patch_ops = {
+	.init			= intel_hdmi_init,
+	.free			= intel_hdmi_free,
+	.build_pcms		= intel_hdmi_build_pcms,
+	.build_controls 	= intel_hdmi_build_controls,
+	.unsol_event		= intel_hdmi_unsol_event,
+};
+
+static int patch_intel_hdmi(struct hda_codec *codec)
+{
+	struct intel_hdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	spec->multiout.num_dacs = 0;	  /* no analog */
+	spec->multiout.max_channels = 8;
+	spec->multiout.dig_out_nid = CVT_NID;
+
+	codec->spec = spec;
+	codec->patch_ops = intel_hdmi_patch_ops;
+
+	snd_hda_eld_proc_new(codec, &spec->sink_eld);
+
+	init_channel_allocations();
+
+	return 0;
+}
+
+static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
+	{ .id = 0x808629fb, .name = "G45 DEVCL",  .patch = patch_intel_hdmi },
+	{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
+	{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
+	{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
+	{ .id = 0x10951392, .name = "SiI1392 HDMI",     .patch = patch_intel_hdmi },
+	{} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:808629fb");
+MODULE_ALIAS("snd-hda-codec-id:80862801");
+MODULE_ALIAS("snd-hda-codec-id:80862802");
+MODULE_ALIAS("snd-hda-codec-id:80862803");
+MODULE_ALIAS("snd-hda-codec-id:10951392");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Intel HDMI HD-audio codec");
+
+static struct hda_codec_preset_list intel_list = {
+	.preset = snd_hda_preset_intelhdmi,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_intelhdmi_init(void)
+{
+	return snd_hda_add_codec_preset(&intel_list);
+}
+
+static void __exit patch_intelhdmi_exit(void)
+{
+	snd_hda_delete_codec_preset(&intel_list);
+}
+
+module_init(patch_intelhdmi_init)
+module_exit(patch_intelhdmi_exit)
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 2eed2c8b98da..0270fda0bda5 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -158,8 +158,34 @@ static int patch_nvhdmi(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
-	{ .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi },
-	{ .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi },
+static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
+	{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi },
+	{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi },
+	{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:10de0002");
+MODULE_ALIAS("snd-hda-codec-id:10de0007");
+MODULE_ALIAS("snd-hda-codec-id:10de0067");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec");
+
+static struct hda_codec_preset_list nvhdmi_list = {
+	.preset = snd_hda_preset_nvhdmi,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_nvhdmi_init(void)
+{
+	return snd_hda_add_codec_preset(&nvhdmi_list);
+}
+
+static void __exit patch_nvhdmi_exit(void)
+{
+	snd_hda_delete_codec_preset(&nvhdmi_list);
+}
+
+module_init(patch_nvhdmi_init)
+module_exit(patch_nvhdmi_exit)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a378c0145125..0bd4e6bf354d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -30,7 +30,6 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 
 #define ALC880_FRONT_EVENT		0x01
 #define ALC880_DCVOL_EVENT		0x02
@@ -114,6 +113,7 @@ enum {
 	ALC268_3ST,
 	ALC268_TOSHIBA,
 	ALC268_ACER,
+	ALC268_ACER_DMIC,
 	ALC268_ACER_ASPIRE_ONE,
 	ALC268_DELL,
 	ALC268_ZEPTO,
@@ -130,6 +130,8 @@ enum {
 	ALC269_QUANTA_FL1,
 	ALC269_ASUS_EEEPC_P703,
 	ALC269_ASUS_EEEPC_P901,
+	ALC269_FUJITSU,
+	ALC269_LIFEBOOK,
 	ALC269_AUTO,
 	ALC269_MODEL_LAST /* last tag */
 };
@@ -152,6 +154,7 @@ enum {
 enum {
 	ALC660VD_3ST,
 	ALC660VD_3ST_DIG,
+	ALC660VD_ASUS_V1S,
 	ALC861VD_3ST,
 	ALC861VD_3ST_DIG,
 	ALC861VD_6ST_DIG,
@@ -212,6 +215,7 @@ enum {
 	ALC883_TARGA_2ch_DIG,
 	ALC883_ACER,
 	ALC883_ACER_ASPIRE,
+	ALC888_ACER_ASPIRE_4930G,
 	ALC883_MEDION,
 	ALC883_MEDION_MD2,
 	ALC883_LAPTOP_EAPD,
@@ -225,9 +229,11 @@ enum {
 	ALC883_MITAC,
 	ALC883_CLEVO_M720,
 	ALC883_FUJITSU_PI2515,
+	ALC888_FUJITSU_XA3530,
 	ALC883_3ST_6ch_INTEL,
 	ALC888_ASUS_M90V,
 	ALC888_ASUS_EEE1601,
+	ALC1200_ASUS_P5Q,
 	ALC883_AUTO,
 	ALC883_MODEL_LAST,
 };
@@ -239,6 +245,7 @@ struct alc_spec {
 	/* codec parameterization */
 	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
 	unsigned int num_mixers;
+	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
 
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL
@@ -268,6 +275,7 @@ struct alc_spec {
 	hda_nid_t *adc_nids;
 	hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
+	unsigned char is_mix_capture;	/* matrix-style capture (non-mux) */
 
 	/* capture source */
 	unsigned int num_mux_defs;
@@ -284,8 +292,7 @@ struct alc_spec {
 
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
-	unsigned int num_kctl_alloc, num_kctl_used;
-	struct snd_kcontrol_new *kctl_alloc;
+	struct snd_array kctls;
 	struct hda_input_mux private_imux;
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
@@ -323,6 +330,7 @@ struct alc_config_preset {
 	struct snd_kcontrol_new *mixers[5]; /* should be identical size
 					     * with spec
 					     */
+	struct snd_kcontrol_new *cap_mixer; /* capture mixer */
 	const struct hda_verb *init_verbs[5];
 	unsigned int num_dacs;
 	hda_nid_t *dac_nids;
@@ -375,14 +383,39 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
+	const struct hda_input_mux *imux;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
+	unsigned int mux_idx;
 	hda_nid_t nid = spec->capsrc_nids ?
 		spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
-	return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
-				     nid, &spec->cur_mux[adc_idx]);
-}
 
+	mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
+	imux = &spec->input_mux[mux_idx];
+
+	if (spec->is_mix_capture) {
+		/* Matrix-mixer style (e.g. ALC882) */
+		unsigned int *cur_val = &spec->cur_mux[adc_idx];
+		unsigned int i, idx;
+
+		idx = ucontrol->value.enumerated.item[0];
+		if (idx >= imux->num_items)
+			idx = imux->num_items - 1;
+		if (*cur_val == idx)
+			return 0;
+		for (i = 0; i < imux->num_items; i++) {
+			unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+						 imux->items[i].index,
+						 HDA_AMP_MUTE, v);
+		}
+		*cur_val = idx;
+		return 1;
+	} else {
+		/* MUX style (e.g. ALC880) */
+		return snd_hda_input_mux_put(codec, imux, ucontrol, nid,
+					     &spec->cur_mux[adc_idx]);
+	}
+}
 
 /*
  * channel mode setting
@@ -716,6 +749,43 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
 	  .private_value = nid | (mask<<16) }
 #endif   /* CONFIG_SND_DEBUG */
 
+/*
+ */
+static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix)
+{
+	if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers)))
+		return;
+	spec->mixers[spec->num_mixers++] = mix;
+}
+
+static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
+{
+	if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs)))
+		return;
+	spec->init_verbs[spec->num_init_verbs++] = verb;
+}
+
+#ifdef CONFIG_PROC_FS
+/*
+ * hook for proc
+ */
+static void print_realtek_coef(struct snd_info_buffer *buffer,
+			       struct hda_codec *codec, hda_nid_t nid)
+{
+	int coeff;
+
+	if (nid != 0x20)
+		return;
+	coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0);
+	snd_iprintf(buffer, "  Processing Coefficient: 0x%02x\n", coeff);
+	coeff = snd_hda_codec_read(codec, nid, 0,
+				   AC_VERB_GET_COEF_INDEX, 0);
+	snd_iprintf(buffer, "  Coefficient Index: 0x%02x\n", coeff);
+}
+#else
+#define print_realtek_coef	NULL
+#endif
+
 /*
  * set up from the preset table
  */
@@ -725,11 +795,11 @@ static void setup_preset(struct alc_spec *spec,
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
-		spec->mixers[spec->num_mixers++] = preset->mixers[i];
+		add_mixer(spec, preset->mixers[i]);
+	spec->cap_mixer = preset->cap_mixer;
 	for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
 	     i++)
-		spec->init_verbs[spec->num_init_verbs++] =
-			preset->init_verbs[i];
+		add_verb(spec, preset->init_verbs[i]);
 
 	spec->channel_mode = preset->channel_mode;
 	spec->num_channel_mode = preset->num_channel_mode;
@@ -1106,6 +1176,226 @@ static void alc_fix_pincfg(struct hda_codec *codec,
 	}
 }
 
+/*
+ * ALC888
+ */
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc888_4ST_ch2_intel_init[] = {
+/* Mic-in jack as mic in */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-in jack as Line in */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-Out as Front */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc888_4ST_ch4_intel_init[] = {
+/* Mic-in jack as mic in */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-in jack as Surround */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as Front */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc888_4ST_ch6_intel_init[] = {
+/* Mic-in jack as CLFE */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-in jack as Surround */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc888_4ST_ch8_intel_init[] = {
+/* Mic-in jack as CLFE */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-in jack as Surround */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as Side */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{ } /* end */
+};
+
+static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
+	{ 2, alc888_4ST_ch2_intel_init },
+	{ 4, alc888_4ST_ch4_intel_init },
+	{ 6, alc888_4ST_ch6_intel_init },
+	{ 8, alc888_4ST_ch8_intel_init },
+};
+
+/*
+ * ALC888 Fujitsu Siemens Amillo xa3530
+ */
+
+static struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
+/* Front Mic: set to PIN_IN (empty by default) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Connect Internal HP to Front */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Bass HP to Front */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Line-Out side jack (SPDIF) to Side */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+/* Connect Mic jack to CLFE */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect Line-in jack to Surround */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect HP out jack to Front */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Enable unsolicited event for HP jack and Line-out jack */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{}
+};
+
+static void alc888_fujitsu_xa3530_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned int bits;
+	/* Line out presence */
+	present = snd_hda_codec_read(codec, 0x17, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	/* HP out presence */
+	present = present || snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? HDA_AMP_MUTE : 0;
+	/* Toggle internal speakers muting */
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	/* Toggle internal bass muting */
+	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc888_fujitsu_xa3530_unsol_event(struct hda_codec *codec,
+		unsigned int res)
+{
+	if (res >> 26 == ALC880_HP_EVENT)
+		alc888_fujitsu_xa3530_automute(codec);
+}
+
+
+/*
+ * ALC888 Acer Aspire 4930G model
+ */
+
+static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
+/* Front Mic: set to PIN_IN (empty by default) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Connect Internal HP to front */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect HP out to front */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ }
+};
+
+static struct hda_input_mux alc888_2_capture_sources[2] = {
+	/* Front mic only available on one ADC */
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Front Mic", 0xb },
+		},
+	},
+	{
+		.num_items = 3,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+		},
+	}
+};
+
+static struct snd_kcontrol_new alc888_base_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+		HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+	{ } /* end */
+};
+
+static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned int bits;
+	present = snd_hda_codec_read(codec, 0x15, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec,
+		unsigned int res)
+{
+	if (res >> 26 == ALC880_HP_EVENT)
+		alc888_acer_aspire_4930g_automute(codec);
+}
+
 /*
  * ALC880 3-stack model
  *
@@ -1205,49 +1495,126 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
 };
 
 /* capture mixer elements */
-static struct snd_kcontrol_new alc880_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 3,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
+static int alc_cap_vol_info(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	int err;
 
-/* capture mixer elements (in case NID 0x07 not available) */
-static struct snd_kcontrol_new alc880_capture_alt_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
+	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
+						      HDA_INPUT);
+	err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
+	mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	return err;
+}
 
+static int alc_cap_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+			   unsigned int size, unsigned int __user *tlv)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	int err;
+
+	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
+						      HDA_INPUT);
+	err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
+	mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	return err;
+}
+
+typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol);
+
+static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol,
+				 getput_call_t func)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	int err;
+
+	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
+						      3, 0, HDA_INPUT);
+	err = func(kcontrol, ucontrol);
+	mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	return err;
+}
+
+static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
+			   struct snd_ctl_elem_value *ucontrol)
+{
+	return alc_cap_getput_caller(kcontrol, ucontrol,
+				     snd_hda_mixer_amp_volume_get);
+}
+
+static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
+			   struct snd_ctl_elem_value *ucontrol)
+{
+	return alc_cap_getput_caller(kcontrol, ucontrol,
+				     snd_hda_mixer_amp_volume_put);
+}
+
+/* capture mixer elements */
+#define alc_cap_sw_info		snd_ctl_boolean_stereo_info
+
+static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	return alc_cap_getput_caller(kcontrol, ucontrol,
+				     snd_hda_mixer_amp_switch_get);
+}
+
+static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	return alc_cap_getput_caller(kcontrol, ucontrol,
+				     snd_hda_mixer_amp_switch_put);
+}
+
+#define DEFINE_CAPMIX(num) \
+static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
+	{ \
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+		.name = "Capture Switch", \
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+		.count = num, \
+		.info = alc_cap_sw_info, \
+		.get = alc_cap_sw_get, \
+		.put = alc_cap_sw_put, \
+	}, \
+	{ \
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+		.name = "Capture Volume", \
+		.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+			   SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+			   SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK), \
+		.count = num, \
+		.info = alc_cap_vol_info, \
+		.get = alc_cap_vol_get, \
+		.put = alc_cap_vol_put, \
+		.tlv = { .c = alc_cap_vol_tlv }, \
+	}, \
+	{ \
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+		/* .name = "Capture Source", */ \
+		.name = "Input Source", \
+		.count = num, \
+		.info = alc_mux_enum_info, \
+		.get = alc_mux_enum_get, \
+		.put = alc_mux_enum_put, \
+	}, \
+	{ } /* end */ \
+}
+
+/* up to three ADCs */
+DEFINE_CAPMIX(1);
+DEFINE_CAPMIX(2);
+DEFINE_CAPMIX(3);
 
 
 /*
@@ -1533,18 +1900,6 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -1619,6 +1974,7 @@ static const char *alc_slave_vols[] = {
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
+	"PCM Playback Volume",
 	NULL,
 };
 
@@ -1638,6 +1994,9 @@ static const char *alc_slave_sws[] = {
 /*
  * build control elements
  */
+
+static void alc_free_kctls(struct hda_codec *codec);
+
 static int alc_build_controls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1649,7 +2008,11 @@ static int alc_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-
+	if (spec->cap_mixer) {
+		err = snd_hda_add_new_ctls(codec, spec->cap_mixer);
+		if (err < 0)
+			return err;
+	}
 	if (spec->multiout.dig_out_nid) {
 		err = snd_hda_create_spdif_out_ctls(codec,
 						    spec->multiout.dig_out_nid);
@@ -1684,6 +2047,7 @@ static int alc_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	alc_free_kctls(codec); /* no longer needed */
 	return 0;
 }
 
@@ -2774,19 +3138,27 @@ static int alc_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+static void alc_free_kctls(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (spec->kctls.list) {
+		struct snd_kcontrol_new *kctl = spec->kctls.list;
+		int i;
+		for (i = 0; i < spec->kctls.used; i++)
+			kfree(kctl[i].name);
+	}
+	snd_array_free(&spec->kctls);
+}
+
 static void alc_free(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int i;
 
 	if (!spec)
 		return;
 
-	if (spec->kctl_alloc) {
-		for (i = 0; i < spec->num_kctl_used; i++)
-			kfree(spec->kctl_alloc[i].name);
-		kfree(spec->kctl_alloc);
-	}
+	alc_free_kctls(codec);
 	kfree(spec);
 	codec->spec = NULL; /* to be sure */
 }
@@ -3268,6 +3640,8 @@ static struct alc_config_preset alc880_presets[] = {
 				alc880_gpio2_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
 		.dac_nids = alc880_dac_nids,
+		.adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */
+		.num_adc_nids = 1, /* single ADC */
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
 		.channel_mode = alc880_2_jack_modes,
@@ -3532,9 +3906,6 @@ static struct alc_config_preset alc880_presets[] = {
  * Automatic parse of I/O pins from the BIOS configuration
  */
 
-#define NUM_CONTROL_ALLOC	32
-#define NUM_VERB_ALLOC		32
-
 enum {
 	ALC_CTL_WIDGET_VOL,
 	ALC_CTL_WIDGET_MUTE,
@@ -3552,29 +3923,15 @@ static int add_control(struct alc_spec *spec, int type, const char *name,
 {
 	struct snd_kcontrol_new *knew;
 
-	if (spec->num_kctl_used >= spec->num_kctl_alloc) {
-		int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
-
-		/* array + terminator */
-		knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL);
-		if (!knew)
-			return -ENOMEM;
-		if (spec->kctl_alloc) {
-			memcpy(knew, spec->kctl_alloc,
-			       sizeof(*knew) * spec->num_kctl_alloc);
-			kfree(spec->kctl_alloc);
-		}
-		spec->kctl_alloc = knew;
-		spec->num_kctl_alloc = num;
-	}
-
-	knew = &spec->kctl_alloc[spec->num_kctl_used];
+	snd_array_init(&spec->kctls, sizeof(*knew), 32);
+	knew = snd_array_new(&spec->kctls);
+	if (!knew)
+		return -ENOMEM;
 	*knew = alc880_control_templates[type];
 	knew->name = kstrdup(name, GFP_KERNEL);
 	if (!knew->name)
 		return -ENOMEM;
 	knew->private_value = val;
-	spec->num_kctl_used++;
 	return 0;
 }
 
@@ -3898,10 +4255,10 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC880_DIGIN_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
-	spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs;
+	add_verb(spec, alc880_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
@@ -3925,6 +4282,17 @@ static void alc880_auto_init(struct hda_codec *codec)
  * OK, here we have finally the patch for ALC880
  */
 
+static void set_capture_mixer(struct alc_spec *spec)
+{
+	static struct snd_kcontrol_new *caps[3] = {
+		alc_capture_mixer1,
+		alc_capture_mixer2,
+		alc_capture_mixer3,
+	};
+	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3)
+		spec->cap_mixer = caps[spec->num_adc_nids - 1];
+}
+
 static int patch_alc880(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
@@ -3980,16 +4348,12 @@ static int patch_alc880(struct hda_codec *codec)
 		if (wcap != AC_WID_AUD_IN) {
 			spec->adc_nids = alc880_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
-			spec->mixers[spec->num_mixers] =
-				alc880_capture_alt_mixer;
-			spec->num_mixers++;
 		} else {
 			spec->adc_nids = alc880_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
-			spec->mixers[spec->num_mixers] = alc880_capture_mixer;
-			spec->num_mixers++;
 		}
 	}
+	set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -4000,6 +4364,7 @@ static int patch_alc880(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc880_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -4024,11 +4389,6 @@ static hda_nid_t alc260_adc_nids_alt[1] = {
 	0x05,
 };
 
-static hda_nid_t alc260_hp_adc_nids[2] = {
-	/* ADC1, 0 */
-	0x05, 0x04
-};
-
 /* NIDs used when simultaneous access to both ADCs makes sense.  Note that
  * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
  */
@@ -4157,13 +4517,13 @@ static void alc260_hp_master_update(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 	unsigned int val = spec->master_sw ? PIN_HP : 0;
 	/* change HP and line-out pins */
-	snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+	snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			    val);
-	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+	snd_hda_codec_write(codec, line, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			    val);
 	/* mono (speaker) depending on the HP jack sense */
 	val = (val && !spec->jack_present) ? PIN_OUT : 0;
-	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+	snd_hda_codec_write(codec, mono, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			    val);
 }
 
@@ -4242,7 +4602,7 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
 		.info = snd_ctl_boolean_mono_info,
 		.get = alc260_hp_master_sw_get,
 		.put = alc260_hp_master_sw_put,
-		.private_value = (0x10 << 16) | (0x15 << 8) | 0x11
+		.private_value = (0x15 << 16) | (0x10 << 8) | 0x11
 	},
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
@@ -4295,7 +4655,7 @@ static void alc260_hp_3013_automute(struct hda_codec *codec)
 	present = snd_hda_codec_read(codec, 0x15, 0,
 				     AC_VERB_GET_PIN_SENSE, 0);
 	spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
-	alc260_hp_master_update(codec, 0x10, 0x15, 0x11);
+	alc260_hp_master_update(codec, 0x15, 0x10, 0x11);
 }
 
 static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
@@ -4427,45 +4787,6 @@ static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
 	{ } /* end */
 };
 
-/* capture mixer elements */
-static struct snd_kcontrol_new alc260_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x05, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static struct snd_kcontrol_new alc260_capture_alt_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 /*
  * initialization verbs
  */
@@ -5282,7 +5603,6 @@ static struct hda_verb alc260_volume_init_verbs[] = {
 static int alc260_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int wcap;
 	int err;
 	static hda_nid_t alc260_ignore[] = { 0x17, 0 };
 
@@ -5293,7 +5613,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	if (!spec->kctl_alloc)
+	if (!spec->kctls.list)
 		return 0; /* can't find valid BIOS pin config */
 	err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg);
 	if (err < 0)
@@ -5303,28 +5623,14 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
-	spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs;
+	add_verb(spec, alc260_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
-	/* check whether NID 0x04 is valid */
-	wcap = get_wcaps(codec, 0x04);
-	wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
-	if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
-		spec->adc_nids = alc260_adc_nids_alt;
-		spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
-		spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer;
-	} else {
-		spec->adc_nids = alc260_adc_nids;
-		spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
-		spec->mixers[spec->num_mixers] = alc260_capture_mixer;
-	}
-	spec->num_mixers++;
-
 	store_pin_configs(codec);
 	return 1;
 }
@@ -5394,12 +5700,11 @@ static struct alc_config_preset alc260_presets[] = {
 	[ALC260_BASIC] = {
 		.mixers = { alc260_base_output_mixer,
 			    alc260_input_mixer,
-			    alc260_pc_beep_mixer,
-			    alc260_capture_mixer },
+			    alc260_pc_beep_mixer },
 		.init_verbs = { alc260_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
 		.adc_nids = alc260_adc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
@@ -5407,14 +5712,13 @@ static struct alc_config_preset alc260_presets[] = {
 	},
 	[ALC260_HP] = {
 		.mixers = { alc260_hp_output_mixer,
-			    alc260_input_mixer,
-			    alc260_capture_alt_mixer },
+			    alc260_input_mixer },
 		.init_verbs = { alc260_init_verbs,
 				alc260_hp_unsol_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
-		.adc_nids = alc260_hp_adc_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+		.adc_nids = alc260_adc_nids_alt,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
@@ -5423,14 +5727,13 @@ static struct alc_config_preset alc260_presets[] = {
 	},
 	[ALC260_HP_DC7600] = {
 		.mixers = { alc260_hp_dc7600_mixer,
-			    alc260_input_mixer,
-			    alc260_capture_alt_mixer },
+			    alc260_input_mixer },
 		.init_verbs = { alc260_init_verbs,
 				alc260_hp_dc7600_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
-		.adc_nids = alc260_hp_adc_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+		.adc_nids = alc260_adc_nids_alt,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
@@ -5439,14 +5742,13 @@ static struct alc_config_preset alc260_presets[] = {
 	},
 	[ALC260_HP_3013] = {
 		.mixers = { alc260_hp_3013_mixer,
-			    alc260_input_mixer,
-			    alc260_capture_alt_mixer },
+			    alc260_input_mixer },
 		.init_verbs = { alc260_hp_3013_init_verbs,
 				alc260_hp_3013_unsol_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
-		.adc_nids = alc260_hp_adc_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+		.adc_nids = alc260_adc_nids_alt,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
@@ -5454,8 +5756,7 @@ static struct alc_config_preset alc260_presets[] = {
 		.init_hook = alc260_hp_3013_automute,
 	},
 	[ALC260_FUJITSU_S702X] = {
-		.mixers = { alc260_fujitsu_mixer,
-			    alc260_capture_mixer },
+		.mixers = { alc260_fujitsu_mixer },
 		.init_verbs = { alc260_fujitsu_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
@@ -5467,8 +5768,7 @@ static struct alc_config_preset alc260_presets[] = {
 		.input_mux = alc260_fujitsu_capture_sources,
 	},
 	[ALC260_ACER] = {
-		.mixers = { alc260_acer_mixer,
-			    alc260_capture_mixer },
+		.mixers = { alc260_acer_mixer },
 		.init_verbs = { alc260_acer_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
@@ -5480,8 +5780,7 @@ static struct alc_config_preset alc260_presets[] = {
 		.input_mux = alc260_acer_capture_sources,
 	},
 	[ALC260_WILL] = {
-		.mixers = { alc260_will_mixer,
-			    alc260_capture_mixer },
+		.mixers = { alc260_will_mixer },
 		.init_verbs = { alc260_init_verbs, alc260_will_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
@@ -5493,8 +5792,7 @@ static struct alc_config_preset alc260_presets[] = {
 		.input_mux = &alc260_capture_source,
 	},
 	[ALC260_REPLACER_672V] = {
-		.mixers = { alc260_replacer_672v_mixer,
-			    alc260_capture_mixer },
+		.mixers = { alc260_replacer_672v_mixer },
 		.init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
@@ -5509,8 +5807,7 @@ static struct alc_config_preset alc260_presets[] = {
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC260_TEST] = {
-		.mixers = { alc260_test_mixer,
-			    alc260_capture_mixer },
+		.mixers = { alc260_test_mixer },
 		.init_verbs = { alc260_test_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
 		.dac_nids = alc260_test_dac_nids,
@@ -5569,6 +5866,21 @@ static int patch_alc260(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc260_pcm_digital_playback;
 	spec->stream_digital_capture = &alc260_pcm_digital_capture;
 
+	if (!spec->adc_nids && spec->input_mux) {
+		/* check whether NID 0x04 is valid */
+		unsigned int wcap = get_wcaps(codec, 0x04);
+		wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		/* get type */
+		if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
+			spec->adc_nids = alc260_adc_nids_alt;
+			spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
+		} else {
+			spec->adc_nids = alc260_adc_nids;
+			spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
+		}
+	}
+	set_capture_mixer(spec);
+
 	spec->vmaster_nid = 0x08;
 
 	codec->patch_ops = alc_patch_ops;
@@ -5578,6 +5890,7 @@ static int patch_alc260(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc260_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -5625,36 +5938,6 @@ static struct hda_input_mux alc882_capture_source = {
 		{ "CD", 0x4 },
 	},
 };
-#define alc882_mux_enum_info alc_mux_enum_info
-#define alc882_mux_enum_get alc_mux_enum_get
-
-static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol,
-			       struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct alc_spec *spec = codec->spec;
-	const struct hda_input_mux *imux = spec->input_mux;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	hda_nid_t nid = spec->capsrc_nids ?
-		spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
-	unsigned int *cur_val = &spec->cur_mux[adc_idx];
-	unsigned int i, idx;
-
-	idx = ucontrol->value.enumerated.item[0];
-	if (idx >= imux->num_items)
-		idx = imux->num_items - 1;
-	if (*cur_val == idx)
-		return 0;
-	for (i = 0; i < imux->num_items; i++) {
-		unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
-		snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
-					 imux->items[i].index,
-					 HDA_AMP_MUTE, v);
-	}
-	*cur_val = idx;
-	return 1;
-}
-
 /*
  * 2ch mode
  */
@@ -6337,49 +6620,6 @@ static struct hda_verb alc882_auto_init_verbs[] = {
 	{ }
 };
 
-/* capture mixer elements */
-static struct snd_kcontrol_new alc882_capture_alt_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc882_mux_enum_info,
-		.get = alc882_mux_enum_get,
-		.put = alc882_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static struct snd_kcontrol_new alc882_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 3,
-		.info = alc882_mux_enum_info,
-		.get = alc882_mux_enum_get,
-		.put = alc882_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc882_loopbacks	alc880_loopbacks
 #endif
@@ -6508,8 +6748,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc885_imac24_init_hook,
 	},
 	[ALC882_TARGA] = {
-		.mixers = { alc882_targa_mixer, alc882_chmode_mixer,
-			    alc882_capture_mixer },
+		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
 		.init_verbs = { alc882_init_verbs, alc882_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
@@ -6525,8 +6764,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc882_targa_automute,
 	},
 	[ALC882_ASUS_A7J] = {
-		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer,
-			    alc882_capture_mixer },
+		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
 		.init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs},
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
@@ -6831,6 +7069,7 @@ static int patch_alc882(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc882_pcm_digital_playback;
 	spec->stream_digital_capture = &alc882_pcm_digital_capture;
 
+	spec->is_mix_capture = 1; /* matrix-style capture */
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
@@ -6840,17 +7079,13 @@ static int patch_alc882(struct hda_codec *codec)
 			spec->adc_nids = alc882_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
 			spec->capsrc_nids = alc882_capsrc_nids_alt;
-			spec->mixers[spec->num_mixers] =
-				alc882_capture_alt_mixer;
-			spec->num_mixers++;
 		} else {
 			spec->adc_nids = alc882_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
 			spec->capsrc_nids = alc882_capsrc_nids;
-			spec->mixers[spec->num_mixers] = alc882_capture_mixer;
-			spec->num_mixers++;
 		}
 	}
+	set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -6861,6 +7096,7 @@ static int patch_alc882(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc882_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -6879,6 +7115,8 @@ static int patch_alc882(struct hda_codec *codec)
 #define ALC883_DIGOUT_NID	0x06
 #define ALC883_DIGIN_NID	0x0a
 
+#define ALC1200_DIGOUT_NID	0x10
+
 static hda_nid_t alc883_dac_nids[4] = {
 	/* front, rear, clfe, rear_surr */
 	0x02, 0x03, 0x04, 0x05
@@ -6889,8 +7127,20 @@ static hda_nid_t alc883_adc_nids[2] = {
 	0x08, 0x09,
 };
 
+static hda_nid_t alc883_adc_nids_alt[1] = {
+	/* ADC1 */
+	0x08,
+};
+
+static hda_nid_t alc883_adc_nids_rev[2] = {
+	/* ADC2-1 */
+	0x09, 0x08
+};
+
 static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
 
+static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
+
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
 
@@ -6957,11 +7207,6 @@ static struct hda_input_mux alc883_asus_eee1601_capture_source = {
 	},
 };
 
-#define alc883_mux_enum_info alc_mux_enum_info
-#define alc883_mux_enum_get alc_mux_enum_get
-/* ALC883 has the ALC882-type input selection */
-#define alc883_mux_enum_put alc882_mux_enum_put
-
 /*
  * 2ch mode
  */
@@ -7115,19 +7360,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7145,19 +7377,6 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7172,19 +7391,6 @@ static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
 	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7199,19 +7405,6 @@ static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
 	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7231,19 +7424,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7269,17 +7449,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7306,19 +7475,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7344,18 +7500,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7376,19 +7520,6 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7404,19 +7535,6 @@ static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7429,17 +7547,6 @@ static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7453,19 +7560,6 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7479,19 +7573,6 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7504,19 +7585,6 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7544,19 +7612,6 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -7587,6 +7642,10 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
 	HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol),
 	HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch),
 	{
@@ -7594,9 +7653,9 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
 		/* .name = "Capture Source", */
 		.name = "Input Source",
 		.count = 1,
-		.info = alc883_mux_enum_info,
-		.get = alc883_mux_enum_get,
-		.put = alc883_mux_enum_put,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
 	},
 	{ } /* end */
 };
@@ -8251,27 +8310,6 @@ static struct hda_verb alc883_auto_init_verbs[] = {
 	{ }
 };
 
-/* capture mixer elements */
-static struct snd_kcontrol_new alc883_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc882_mux_enum_info,
-		.get = alc882_mux_enum_get,
-		.put = alc882_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 static struct hda_verb alc888_asus_m90v_verbs[] = {
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -8394,6 +8432,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 	[ALC883_TARGA_2ch_DIG]	= "targa-2ch-dig",
 	[ALC883_ACER]		= "acer",
 	[ALC883_ACER_ASPIRE]	= "acer-aspire",
+	[ALC888_ACER_ASPIRE_4930G]	= "acer-aspire-4930g",
 	[ALC883_MEDION]		= "medion",
 	[ALC883_MEDION_MD2]	= "medion-md2",
 	[ALC883_LAPTOP_EAPD]	= "laptop-eapd",
@@ -8407,7 +8446,9 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 	[ALC883_MITAC]		= "mitac",
 	[ALC883_CLEVO_M720]	= "clevo-m720",
 	[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
+	[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
 	[ALC883_3ST_6ch_INTEL]	= "3stack-6ch-intel",
+	[ALC1200_ASUS_P5Q]	= "asus-p5q",
 	[ALC883_AUTO]		= "auto",
 };
 
@@ -8418,6 +8459,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G",
+		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
@@ -8426,6 +8469,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
@@ -8452,6 +8496,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
@@ -8463,6 +8508,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
 	SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515),
+	SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530",
+		ALC888_FUJITSU_XA3530),
 	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
 	SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
@@ -8553,6 +8600,8 @@ static struct alc_config_preset alc883_presets[] = {
 		.init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
+		.adc_nids = alc883_adc_nids_alt,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
 		.dig_out_nid = ALC883_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
@@ -8586,6 +8635,26 @@ static struct alc_config_preset alc883_presets[] = {
 		.unsol_event = alc883_acer_aspire_unsol_event,
 		.init_hook = alc883_acer_aspire_automute,
 	},
+	[ALC888_ACER_ASPIRE_4930G] = {
+		.mixers = { alc888_base_mixer,
+				alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc888_acer_aspire_4930g_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.num_mux_defs =
+			ARRAY_SIZE(alc888_2_capture_sources),
+		.input_mux = alc888_2_capture_sources,
+		.unsol_event = alc888_acer_aspire_4930g_unsol_event,
+		.init_hook = alc888_acer_aspire_4930g_automute,
+	},
 	[ALC883_MEDION] = {
 		.mixers = { alc883_fivestack_mixer,
 			    alc883_chmode_mixer },
@@ -8593,6 +8662,8 @@ static struct alc_config_preset alc883_presets[] = {
 				alc883_medion_eapd_verbs },
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
+		.adc_nids = alc883_adc_nids_alt,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
 		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
 		.channel_mode = alc883_sixstack_modes,
 		.input_mux = &alc883_capture_source,
@@ -8635,6 +8706,8 @@ static struct alc_config_preset alc883_presets[] = {
 		.init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
+		.adc_nids = alc883_adc_nids_alt,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_lenovo_101e_capture_source,
@@ -8725,14 +8798,30 @@ static struct alc_config_preset alc883_presets[] = {
 		.unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event,
 		.init_hook = alc883_2ch_fujitsu_pi2515_automute,
 	},
+	[ALC888_FUJITSU_XA3530] = {
+		.mixers = { alc888_base_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs,
+			alc888_fujitsu_xa3530_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes),
+		.channel_mode = alc888_4ST_8ch_intel_modes,
+		.num_mux_defs =
+			ARRAY_SIZE(alc888_2_capture_sources),
+		.input_mux = alc888_2_capture_sources,
+		.unsol_event = alc888_fujitsu_xa3530_unsol_event,
+		.init_hook = alc888_fujitsu_xa3530_automute,
+	},
 	[ALC888_LENOVO_SKY] = {
 		.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
-		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
-		.adc_nids = alc883_adc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
 		.channel_mode = alc883_sixstack_modes,
 		.need_dac_fix = 1,
@@ -8756,6 +8845,7 @@ static struct alc_config_preset alc883_presets[] = {
 	},
 	[ALC888_ASUS_EEE1601] = {
 		.mixers = { alc883_asus_eee1601_mixer },
+		.cap_mixer = alc883_asus_eee1601_cap_mixer,
 		.init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs },
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
@@ -8768,6 +8858,17 @@ static struct alc_config_preset alc883_presets[] = {
 		.unsol_event = alc883_eee1601_unsol_event,
 		.init_hook = alc883_eee1601_inithook,
 	},
+	[ALC1200_ASUS_P5Q] = {
+		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC1200_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+		.channel_mode = alc883_sixstack_modes,
+		.input_mux = &alc883_capture_source,
+	},
 };
 
 
@@ -8862,8 +8963,6 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
 
 	/* hack - override the init verbs */
 	spec->init_verbs[0] = alc883_auto_init_verbs;
-	spec->mixers[spec->num_mixers] = alc883_capture_mixer;
-	spec->num_mixers++;
 
 	return 1; /* config found */
 }
@@ -8946,9 +9045,15 @@ static int patch_alc883(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc883_pcm_digital_playback;
 	spec->stream_digital_capture = &alc883_pcm_digital_capture;
 
-	spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
-	spec->adc_nids = alc883_adc_nids;
-	spec->capsrc_nids = alc883_capsrc_nids;
+	if (!spec->num_adc_nids) {
+		spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+		spec->adc_nids = alc883_adc_nids;
+	}
+	if (!spec->capsrc_nids)
+		spec->capsrc_nids = alc883_capsrc_nids;
+	spec->is_mix_capture = 1; /* matrix-style capture */
+	if (!spec->cap_mixer)
+		set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -8960,6 +9065,7 @@ static int patch_alc883(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc883_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -9439,20 +9545,6 @@ static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -9969,7 +10061,7 @@ static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol,
 	struct alc_spec *spec = codec->spec;
 	int ret;
 
-	ret = alc882_mux_enum_put(kcontrol, ucontrol);
+	ret = alc_mux_enum_put(kcontrol, ucontrol);
 	if (!ret)
 		return 0;
 	/* reprogram the HP pin as mic or HP according to the input source */
@@ -9986,8 +10078,8 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Capture Source",
-		.info = alc882_mux_enum_info,
-		.get = alc882_mux_enum_get,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
 		.put = alc262_ultra_mux_enum_put,
 	},
 	{ } /* end */
@@ -10380,10 +10472,10 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC262_DIGIN_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
-	spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs;
+	add_verb(spec, alc262_volume_init_verbs);
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
@@ -10466,6 +10558,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN",
+		      ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
 		      ALC262_TOSHIBA_RX1),
 	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
@@ -10624,7 +10718,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.init_hook = alc262_hippo_automute,
 	},
 	[ALC262_ULTRA] = {
-		.mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer },
+		.mixers = { alc262_ultra_mixer },
+		.cap_mixer = alc262_ultra_capture_mixer,
 		.init_verbs = { alc262_ultra_verbs },
 		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
 		.dac_nids = alc262_dac_nids,
@@ -10750,6 +10845,7 @@ static int patch_alc262(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc262_pcm_digital_playback;
 	spec->stream_digital_capture = &alc262_pcm_digital_capture;
 
+	spec->is_mix_capture = 1;
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
@@ -10760,17 +10856,14 @@ static int patch_alc262(struct hda_codec *codec)
 			spec->adc_nids = alc262_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt);
 			spec->capsrc_nids = alc262_capsrc_nids_alt;
-			spec->mixers[spec->num_mixers] =
-				alc262_capture_alt_mixer;
-			spec->num_mixers++;
 		} else {
 			spec->adc_nids = alc262_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids);
 			spec->capsrc_nids = alc262_capsrc_nids;
-			spec->mixers[spec->num_mixers] = alc262_capture_mixer;
-			spec->num_mixers++;
 		}
 	}
+	if (!spec->cap_mixer)
+		set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -10781,6 +10874,7 @@ static int patch_alc262(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc262_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -10942,6 +11036,22 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = {
 	{ }
 };
 
+static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
+	/* output mixer control */
+	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_amp_switch_get,
+		.put = alc268_acer_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+	},
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
+	{ }
+};
+
 static struct hda_verb alc268_acer_aspire_one_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -11218,10 +11328,6 @@ static struct hda_verb alc268_volume_init_verbs[] = {
 	{ }
 };
 
-#define alc268_mux_enum_info alc_mux_enum_info
-#define alc268_mux_enum_get alc_mux_enum_get
-#define alc268_mux_enum_put alc_mux_enum_put
-
 static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
@@ -11233,9 +11339,9 @@ static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
 		/* .name = "Capture Source", */
 		.name = "Input Source",
 		.count = 1,
-		.info = alc268_mux_enum_info,
-		.get = alc268_mux_enum_get,
-		.put = alc268_mux_enum_put,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
 	},
 	{ } /* end */
 };
@@ -11253,9 +11359,9 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = {
 		/* .name = "Capture Source", */
 		.name = "Input Source",
 		.count = 2,
-		.info = alc268_mux_enum_info,
-		.get = alc268_mux_enum_get,
-		.put = alc268_mux_enum_put,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
 	},
 	{ } /* end */
 };
@@ -11271,6 +11377,15 @@ static struct hda_input_mux alc268_capture_source = {
 };
 
 static struct hda_input_mux alc268_acer_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+static struct hda_input_mux alc268_acer_dmic_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -11512,13 +11627,13 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
 	if (spec->autocfg.speaker_pins[0] != 0x1d)
-		spec->mixers[spec->num_mixers++] = alc268_beep_mixer;
+		add_mixer(spec, alc268_beep_mixer);
 
-	spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs;
+	add_verb(spec, alc268_volume_init_verbs);
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
@@ -11554,6 +11669,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = {
 	[ALC268_3ST]		= "3stack",
 	[ALC268_TOSHIBA]	= "toshiba",
 	[ALC268_ACER]		= "acer",
+	[ALC268_ACER_DMIC]	= "acer-dmic",
 	[ALC268_ACER_ASPIRE_ONE]	= "acer-aspire",
 	[ALC268_DELL]		= "dell",
 	[ALC268_ZEPTO]		= "zepto",
@@ -11649,6 +11765,23 @@ static struct alc_config_preset alc268_presets[] = {
 		.unsol_event = alc268_acer_unsol_event,
 		.init_hook = alc268_acer_init_hook,
 	},
+	[ALC268_ACER_DMIC] = {
+		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_acer_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x02,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.input_mux = &alc268_acer_dmic_capture_source,
+		.unsol_event = alc268_acer_unsol_event,
+		.init_hook = alc268_acer_init_hook,
+	},
 	[ALC268_ACER_ASPIRE_ONE] = {
 		.mixers = { alc268_acer_aspire_one_mixer,
 				alc268_capture_alt_mixer },
@@ -11787,15 +11920,11 @@ static int patch_alc268(struct hda_codec *codec)
 		if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc268_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
-			spec->mixers[spec->num_mixers] =
-					alc268_capture_alt_mixer;
-			spec->num_mixers++;
+			add_mixer(spec, alc268_capture_alt_mixer);
 		} else {
 			spec->adc_nids = alc268_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
-			spec->mixers[spec->num_mixers] =
-				alc268_capture_mixer;
-			spec->num_mixers++;
+			add_mixer(spec, alc268_capture_mixer);
 		}
 		spec->capsrc_nids = alc268_capsrc_nids;
 		/* set default input source */
@@ -11811,6 +11940,8 @@ static int patch_alc268(struct hda_codec *codec)
 	if (board_config == ALC268_AUTO)
 		spec->init_hook = alc268_auto_init;
 
+	codec->proc_widget_hook = print_realtek_coef;
+
 	return 0;
 }
 
@@ -11893,6 +12024,31 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
 	{ }
 };
 
+static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
+	/* output mixer control */
+	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_amp_switch_get,
+		.put = alc268_acer_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+	},
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	{ }
+};
+
 /* bind volumes of both NID 0x0c and 0x0d */
 static struct hda_bind_ctls alc269_epc_bind_vol = {
 	.ops = &snd_hda_bind_vol,
@@ -11910,29 +12066,19 @@ static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
 	{ } /* end */
 };
 
-/* capture mixer elements */
-static struct snd_kcontrol_new alc269_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 /* capture mixer elements */
 static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+/* FSC amilo */
+static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol),
 	{ } /* end */
 };
 
@@ -11953,6 +12099,20 @@ static struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc269_lifebook_verbs[] = {
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{ }
+};
+
 /* toggle speaker-output according to the hp-jack state */
 static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
 {
@@ -11978,6 +12138,37 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
 			AC_VERB_SET_PROC_COEF, 0x480);
 }
 
+/* toggle speaker-output according to the hp-jacks state */
+static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	/* Check laptop headphone socket */
+	present = snd_hda_codec_read(codec, 0x15, 0,
+			AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+	/* Check port replicator headphone socket */
+	present |= snd_hda_codec_read(codec, 0x1a, 0,
+			AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+	bits = present ? AMP_IN_MUTE(0) : 0;
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+			AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+			AMP_IN_MUTE(0), bits);
+
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_COEF_INDEX, 0x0c);
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_PROC_COEF, 0x680);
+
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_COEF_INDEX, 0x0c);
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_PROC_COEF, 0x480);
+}
+
 static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
 {
 	unsigned int present;
@@ -11988,6 +12179,29 @@ static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
 			    AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1);
 }
 
+static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
+{
+	unsigned int present_laptop;
+	unsigned int present_dock;
+
+	present_laptop = snd_hda_codec_read(codec, 0x18, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+	present_dock = snd_hda_codec_read(codec, 0x1b, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+	/* Laptop mic port overrides dock mic port, design decision */
+	if (present_dock)
+		snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL, 0x3);
+	if (present_laptop)
+		snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL, 0x0);
+	if (!present_dock && !present_laptop)
+		snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL, 0x1);
+}
+
 static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
 				    unsigned int res)
 {
@@ -11997,12 +12211,27 @@ static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
 		alc269_quanta_fl1_mic_automute(codec);
 }
 
+static void alc269_lifebook_unsol_event(struct hda_codec *codec,
+					unsigned int res)
+{
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc269_lifebook_speaker_automute(codec);
+	if ((res >> 26) == ALC880_MIC_EVENT)
+		alc269_lifebook_mic_autoswitch(codec);
+}
+
 static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
 {
 	alc269_quanta_fl1_speaker_automute(codec);
 	alc269_quanta_fl1_mic_automute(codec);
 }
 
+static void alc269_lifebook_init_hook(struct hda_codec *codec)
+{
+	alc269_lifebook_speaker_automute(codec);
+	alc269_lifebook_mic_autoswitch(codec);
+}
+
 static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
@@ -12303,17 +12532,17 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = ALC269_DIGOUT_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
 	/* create a beep mixer control if the pin 0x1d isn't assigned */
 	for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++)
 		if (spec->autocfg.input_pins[i] == 0x1d)
 			break;
 	if (i >= ARRAY_SIZE(spec->autocfg.input_pins))
-		spec->mixers[spec->num_mixers++] = alc269_beep_mixer;
+		add_mixer(spec, alc269_beep_mixer);
 
-	spec->init_verbs[spec->num_init_verbs++] = alc269_init_verbs;
+	add_verb(spec, alc269_init_verbs);
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 	/* set default input source */
@@ -12325,8 +12554,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	spec->mixers[spec->num_mixers] = alc269_capture_mixer;
-	spec->num_mixers++;
+	if (!spec->cap_mixer)
+		set_capture_mixer(spec);
 
 	store_pin_configs(codec);
 	return 1;
@@ -12355,7 +12584,9 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
 	[ALC269_BASIC]			= "basic",
 	[ALC269_QUANTA_FL1]		= "quanta",
 	[ALC269_ASUS_EEEPC_P703]	= "eeepc-p703",
-	[ALC269_ASUS_EEEPC_P901]	= "eeepc-p901"
+	[ALC269_ASUS_EEEPC_P901]	= "eeepc-p901",
+	[ALC269_FUJITSU]		= "fujitsu",
+	[ALC269_LIFEBOOK]		= "lifebook"
 };
 
 static struct snd_pci_quirk alc269_cfg_tbl[] = {
@@ -12366,12 +12597,14 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
 		      ALC269_ASUS_EEEPC_P901),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
 		      ALC269_ASUS_EEEPC_P901),
+	SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
+	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
 	{}
 };
 
 static struct alc_config_preset alc269_presets[] = {
 	[ALC269_BASIC] = {
-		.mixers = { alc269_base_mixer, alc269_capture_mixer },
+		.mixers = { alc269_base_mixer },
 		.init_verbs = { alc269_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
 		.dac_nids = alc269_dac_nids,
@@ -12393,7 +12626,8 @@ static struct alc_config_preset alc269_presets[] = {
 		.init_hook = alc269_quanta_fl1_init_hook,
 	},
 	[ALC269_ASUS_EEEPC_P703] = {
-		.mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer },
+		.mixers = { alc269_eeepc_mixer },
+		.cap_mixer = alc269_epc_capture_mixer,
 		.init_verbs = { alc269_init_verbs,
 				alc269_eeepc_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
@@ -12406,7 +12640,8 @@ static struct alc_config_preset alc269_presets[] = {
 		.init_hook = alc269_eeepc_amic_inithook,
 	},
 	[ALC269_ASUS_EEEPC_P901] = {
-		.mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer},
+		.mixers = { alc269_eeepc_mixer },
+		.cap_mixer = alc269_epc_capture_mixer,
 		.init_verbs = { alc269_init_verbs,
 				alc269_eeepc_dmic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
@@ -12418,6 +12653,32 @@ static struct alc_config_preset alc269_presets[] = {
 		.unsol_event = alc269_eeepc_dmic_unsol_event,
 		.init_hook = alc269_eeepc_dmic_inithook,
 	},
+	[ALC269_FUJITSU] = {
+		.mixers = { alc269_fujitsu_mixer, alc269_beep_mixer },
+		.cap_mixer = alc269_epc_capture_mixer,
+		.init_verbs = { alc269_init_verbs,
+				alc269_eeepc_dmic_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.input_mux = &alc269_eeepc_dmic_capture_source,
+		.unsol_event = alc269_eeepc_dmic_unsol_event,
+		.init_hook = alc269_eeepc_dmic_inithook,
+	},
+	[ALC269_LIFEBOOK] = {
+		.mixers = { alc269_lifebook_mixer },
+		.init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.input_mux = &alc269_capture_source,
+		.unsol_event = alc269_lifebook_unsol_event,
+		.init_hook = alc269_lifebook_init_hook,
+	},
 };
 
 static int patch_alc269(struct hda_codec *codec)
@@ -12472,6 +12733,8 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->adc_nids = alc269_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
 	spec->capsrc_nids = alc269_capsrc_nids;
+	if (!spec->cap_mixer)
+		set_capture_mixer(spec);
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC269_AUTO)
@@ -12480,6 +12743,7 @@ static int patch_alc269(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc269_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -12612,17 +12876,6 @@ static struct snd_kcontrol_new alc861_base_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
 
-        /* Capture mixer control */
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -12646,17 +12899,6 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
 
-	/* Capture mixer control */
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -12674,18 +12916,6 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
 
-        /*Capture mixer control */
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-
 	{ } /* end */
 };
 
@@ -12709,17 +12939,6 @@ static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
 
-	/* Capture mixer control */
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -12751,17 +12970,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
 
-	/* Capture mixer control */
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -13293,25 +13501,6 @@ static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
 	return 0;
 }
 
-static struct snd_kcontrol_new alc861_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid,
 					      int pin_type, int dac_idx)
@@ -13402,18 +13591,17 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
-	spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs;
+	add_verb(spec, alc861_auto_init_verbs);
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
 	spec->adc_nids = alc861_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
-	spec->mixers[spec->num_mixers] = alc861_capture_mixer;
-	spec->num_mixers++;
+	set_capture_mixer(spec);
 
 	store_pin_configs(codec);
 	return 1;
@@ -13644,6 +13832,7 @@ static int patch_alc861(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc861_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -13709,11 +13898,6 @@ static struct hda_input_mux alc861vd_hp_capture_source = {
 	},
 };
 
-#define alc861vd_mux_enum_info alc_mux_enum_info
-#define alc861vd_mux_enum_get alc_mux_enum_get
-/* ALC861VD has the ALC882-type input selection (but has only one ADC) */
-#define alc861vd_mux_enum_put alc882_mux_enum_put
-
 /*
  * 2ch mode
  */
@@ -13759,25 +13943,6 @@ static struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861vd_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc861vd_mux_enum_info,
-		.get = alc861vd_mux_enum_get,
-		.put = alc861vd_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
  *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
  */
@@ -14169,6 +14334,7 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re
 static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
 	[ALC660VD_3ST]		= "3stack-660",
 	[ALC660VD_3ST_DIG]	= "3stack-660-digout",
+	[ALC660VD_ASUS_V1S]	= "asus-v1s",
 	[ALC861VD_3ST]		= "3stack",
 	[ALC861VD_3ST_DIG]	= "3stack-digout",
 	[ALC861VD_6ST_DIG]	= "6stack-digout",
@@ -14183,7 +14349,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
 	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
-	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC861VD_LENOVO),
+	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
 	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
@@ -14290,6 +14456,21 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.unsol_event = alc861vd_dallas_unsol_event,
 		.init_hook = alc861vd_dallas_automute,
 	},
+	[ALC660VD_ASUS_V1S] = {
+		.mixers = { alc861vd_lenovo_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				alc861vd_3stack_init_verbs,
+				alc861vd_eapd_verbs,
+				alc861vd_lenovo_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+		.dac_nids = alc660vd_dac_nids,
+		.dig_out_nid = ALC861VD_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+		.unsol_event = alc861vd_lenovo_unsol_event,
+		.init_hook = alc861vd_lenovo_automute,
+	},
 };
 
 /*
@@ -14514,11 +14695,10 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
-	spec->init_verbs[spec->num_init_verbs++]
-		= alc861vd_volume_init_verbs;
+	add_verb(spec, alc861vd_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
@@ -14585,7 +14765,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 		spec->stream_name_analog = "ALC660-VD Analog";
 		spec->stream_name_digital = "ALC660-VD Digital";
 		/* always turn on EAPD */
-		spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs;
+		add_verb(spec, alc660vd_eapd_verbs);
 	} else {
 		spec->stream_name_analog = "ALC861VD Analog";
 		spec->stream_name_digital = "ALC861VD Digital";
@@ -14600,9 +14780,9 @@ static int patch_alc861vd(struct hda_codec *codec)
 	spec->adc_nids = alc861vd_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
 	spec->capsrc_nids = alc861vd_capsrc_nids;
+	spec->is_mix_capture = 1;
 
-	spec->mixers[spec->num_mixers] = alc861vd_capture_mixer;
-	spec->num_mixers++;
+	set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x02;
 
@@ -14614,6 +14794,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc861vd_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -14689,10 +14870,6 @@ static struct hda_input_mux alc663_m51va_capture_source = {
 	},
 };
 
-#define alc662_mux_enum_info alc_mux_enum_info
-#define alc662_mux_enum_get alc_mux_enum_get
-#define alc662_mux_enum_put alc882_mux_enum_put
-
 /*
  * 2ch mode
  */
@@ -15278,25 +15455,6 @@ static struct hda_verb alc662_ecs_init_verbs[] = {
 	{}
 };
 
-/* capture mixer elements */
-static struct snd_kcontrol_new alc662_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc662_mux_enum_info,
-		.get = alc662_mux_enum_get,
-		.put = alc662_mux_enum_put,
-	},
-	{ } /* end */
-};
-
 static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
@@ -15868,7 +16026,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 
 static struct alc_config_preset alc662_presets[] = {
 	[ALC662_3ST_2ch_DIG] = {
-		.mixers = { alc662_3ST_2ch_mixer, alc662_capture_mixer },
+		.mixers = { alc662_3ST_2ch_mixer },
 		.init_verbs = { alc662_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15879,8 +16037,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.input_mux = &alc662_capture_source,
 	},
 	[ALC662_3ST_6ch_DIG] = {
-		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer,
-			    alc662_capture_mixer },
+		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
 		.init_verbs = { alc662_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15892,8 +16049,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.input_mux = &alc662_capture_source,
 	},
 	[ALC662_3ST_6ch] = {
-		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer,
-			    alc662_capture_mixer },
+		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
 		.init_verbs = { alc662_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15903,8 +16059,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.input_mux = &alc662_capture_source,
 	},
 	[ALC662_5ST_DIG] = {
-		.mixers = { alc662_base_mixer, alc662_chmode_mixer,
-			    alc662_capture_mixer },
+		.mixers = { alc662_base_mixer, alc662_chmode_mixer },
 		.init_verbs = { alc662_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15915,7 +16070,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.input_mux = &alc662_capture_source,
 	},
 	[ALC662_LENOVO_101E] = {
-		.mixers = { alc662_lenovo_101e_mixer, alc662_capture_mixer },
+		.mixers = { alc662_lenovo_101e_mixer },
 		.init_verbs = { alc662_init_verbs, alc662_sue_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15926,7 +16081,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc662_lenovo_101e_all_automute,
 	},
 	[ALC662_ASUS_EEEPC_P701] = {
-		.mixers = { alc662_eeepc_p701_mixer, alc662_capture_mixer },
+		.mixers = { alc662_eeepc_p701_mixer },
 		.init_verbs = { alc662_init_verbs,
 				alc662_eeepc_sue_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -15938,7 +16093,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc662_eeepc_inithook,
 	},
 	[ALC662_ASUS_EEEPC_EP20] = {
-		.mixers = { alc662_eeepc_ep20_mixer, alc662_capture_mixer,
+		.mixers = { alc662_eeepc_ep20_mixer,
 			    alc662_chmode_mixer },
 		.init_verbs = { alc662_init_verbs,
 				alc662_eeepc_ep20_sue_init_verbs },
@@ -15951,7 +16106,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc662_eeepc_ep20_inithook,
 	},
 	[ALC662_ECS] = {
-		.mixers = { alc662_ecs_mixer, alc662_capture_mixer },
+		.mixers = { alc662_ecs_mixer },
 		.init_verbs = { alc662_init_verbs,
 				alc662_ecs_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -15963,7 +16118,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc662_eeepc_inithook,
 	},
 	[ALC663_ASUS_M51VA] = {
-		.mixers = { alc663_m51va_mixer, alc662_capture_mixer},
+		.mixers = { alc663_m51va_mixer },
 		.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15975,7 +16130,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_m51va_inithook,
 	},
 	[ALC663_ASUS_G71V] = {
-		.mixers = { alc663_g71v_mixer, alc662_capture_mixer},
+		.mixers = { alc663_g71v_mixer },
 		.init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15987,7 +16142,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_g71v_inithook,
 	},
 	[ALC663_ASUS_H13] = {
-		.mixers = { alc663_m51va_mixer, alc662_capture_mixer},
+		.mixers = { alc663_m51va_mixer },
 		.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -15998,7 +16153,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_m51va_inithook,
 	},
 	[ALC663_ASUS_G50V] = {
-		.mixers = { alc663_g50v_mixer, alc662_capture_mixer},
+		.mixers = { alc663_g50v_mixer },
 		.init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
@@ -16010,7 +16165,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_g50v_inithook,
 	},
 	[ALC663_ASUS_MODE1] = {
-		.mixers = { alc663_m51va_mixer, alc662_auto_capture_mixer },
+		.mixers = { alc663_m51va_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
 				alc663_21jd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -16024,7 +16180,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_mode1_inithook,
 	},
 	[ALC662_ASUS_MODE2] = {
-		.mixers = { alc662_1bjd_mixer, alc662_auto_capture_mixer },
+		.mixers = { alc662_1bjd_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
 				alc662_1bjd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -16037,7 +16194,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc662_mode2_inithook,
 	},
 	[ALC663_ASUS_MODE3] = {
-		.mixers = { alc663_two_hp_m1_mixer, alc662_auto_capture_mixer },
+		.mixers = { alc663_two_hp_m1_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
 				alc663_two_hp_amic_m1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -16051,8 +16209,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_mode3_inithook,
 	},
 	[ALC663_ASUS_MODE4] = {
-		.mixers = { alc663_asus_21jd_clfe_mixer,
-				alc662_auto_capture_mixer},
+		.mixers = { alc663_asus_21jd_clfe_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
 				alc663_21jd_amic_init_verbs},
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -16066,8 +16224,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_mode4_inithook,
 	},
 	[ALC663_ASUS_MODE5] = {
-		.mixers = { alc663_asus_15jd_clfe_mixer,
-				alc662_auto_capture_mixer },
+		.mixers = { alc663_asus_15jd_clfe_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
 				alc663_15jd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -16081,7 +16239,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.init_hook = alc663_mode5_inithook,
 	},
 	[ALC663_ASUS_MODE6] = {
-		.mixers = { alc663_two_hp_m2_mixer, alc662_auto_capture_mixer },
+		.mixers = { alc663_two_hp_m2_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
 				alc663_two_hp_amic_m2_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
@@ -16342,24 +16501,20 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
-	spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs;
+	add_verb(spec, alc662_auto_init_verbs);
 	if (codec->vendor_id == 0x10ec0663)
-		spec->init_verbs[spec->num_init_verbs++] =
-			alc663_auto_init_verbs;
+		add_verb(spec, alc663_auto_init_verbs);
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
 		return err;
 
-	spec->mixers[spec->num_mixers] = alc662_capture_mixer;
-	spec->num_mixers++;
-
 	store_pin_configs(codec);
 	return 1;
 }
@@ -16435,6 +16590,10 @@ static int patch_alc662(struct hda_codec *codec)
 	spec->adc_nids = alc662_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
 	spec->capsrc_nids = alc662_capsrc_nids;
+	spec->is_mix_capture = 1;
+
+	if (!spec->cap_mixer)
+		set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x02;
 
@@ -16445,6 +16604,7 @@ static int patch_alc662(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc662_loopbacks;
 #endif
+	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
@@ -16452,7 +16612,7 @@ static int patch_alc662(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_realtek[] = {
+static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
 	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
 	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
@@ -16484,3 +16644,26 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:10ec*");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Realtek HD-audio codec");
+
+static struct hda_codec_preset_list realtek_list = {
+	.preset = snd_hda_preset_realtek,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_realtek_init(void)
+{
+	return snd_hda_add_codec_preset(&realtek_list);
+}
+
+static void __exit patch_realtek_exit(void)
+{
+	snd_hda_delete_codec_preset(&realtek_list);
+}
+
+module_init(patch_realtek_init)
+module_exit(patch_realtek_exit)
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 9332b63e406c..43b436c5d01b 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -28,7 +28,6 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 
 /* si3054 verbs */
 #define SI3054_VERB_READ_NODE  0x900
@@ -283,7 +282,7 @@ static int patch_si3054(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_si3054[] = {
+static struct hda_codec_preset snd_hda_preset_si3054[] = {
  	{ .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
  	{ .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
  	{ .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
@@ -301,3 +300,35 @@ struct hda_codec_preset snd_hda_preset_si3054[] = {
 	{}
 };
 
+MODULE_ALIAS("snd-hda-codec-id:163c3055");
+MODULE_ALIAS("snd-hda-codec-id:163c3155");
+MODULE_ALIAS("snd-hda-codec-id:11c13026");
+MODULE_ALIAS("snd-hda-codec-id:11c13055");
+MODULE_ALIAS("snd-hda-codec-id:11c13155");
+MODULE_ALIAS("snd-hda-codec-id:10573055");
+MODULE_ALIAS("snd-hda-codec-id:10573057");
+MODULE_ALIAS("snd-hda-codec-id:10573155");
+MODULE_ALIAS("snd-hda-codec-id:11063288");
+MODULE_ALIAS("snd-hda-codec-id:15433155");
+MODULE_ALIAS("snd-hda-codec-id:18540018");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Si3054 HD-audio modem codec");
+
+static struct hda_codec_preset_list si3054_list = {
+	.preset = snd_hda_preset_si3054,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_si3054_init(void)
+{
+	return snd_hda_add_codec_preset(&si3054_list);
+}
+
+static void __exit patch_si3054_exit(void)
+{
+	snd_hda_delete_codec_preset(&si3054_list);
+}
+
+module_init(patch_si3054_init)
+module_exit(patch_si3054_exit)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b77f330d2650..35b83dc6e19e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -30,17 +30,17 @@
 #include <linux/pci.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
+#include <sound/jack.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 #include "hda_beep.h"
 
-#define NUM_CONTROL_ALLOC	32
-
-#define STAC_VREF_EVENT		0x00
-#define STAC_INSERT_EVENT	0x10
-#define STAC_PWR_EVENT		0x20
-#define STAC_HP_EVENT		0x30
+enum {
+	STAC_VREF_EVENT	= 1,
+	STAC_INSERT_EVENT,
+	STAC_PWR_EVENT,
+	STAC_HP_EVENT,
+};
 
 enum {
 	STAC_REF,
@@ -137,6 +137,19 @@ enum {
 	STAC_927X_MODELS
 };
 
+struct sigmatel_event {
+	hda_nid_t nid;
+	unsigned char type;
+	unsigned char tag;
+	int data;
+};
+
+struct sigmatel_jack {
+	hda_nid_t nid;
+	int type;
+	struct snd_jack *jack;
+};
+
 struct sigmatel_spec {
 	struct snd_kcontrol_new *mixers[4];
 	unsigned int num_mixers;
@@ -144,8 +157,6 @@ struct sigmatel_spec {
 	int board_config;
 	unsigned int eapd_switch: 1;
 	unsigned int surr_switch: 1;
-	unsigned int line_switch: 1;
-	unsigned int mic_switch: 1;
 	unsigned int alt_switch: 1;
 	unsigned int hp_detect: 1;
 	unsigned int spdif_mute: 1;
@@ -170,12 +181,20 @@ struct sigmatel_spec {
 	hda_nid_t *pwr_nids;
 	hda_nid_t *dac_list;
 
+	/* jack detection */
+	struct snd_array jacks;
+
+	/* events */
+	struct snd_array events;
+
 	/* playback */
 	struct hda_input_mux *mono_mux;
 	struct hda_input_mux *amp_mux;
 	unsigned int cur_mmux;
 	struct hda_multi_out multiout;
 	hda_nid_t dac_nids[5];
+	hda_nid_t hp_dacs[5];
+	hda_nid_t speaker_dacs[5];
 
 	/* capture */
 	hda_nid_t *adc_nids;
@@ -199,7 +218,6 @@ struct sigmatel_spec {
 	hda_nid_t *pin_nids;
 	unsigned int num_pins;
 	unsigned int *pin_configs;
-	unsigned int *bios_pin_configs;
 
 	/* codec specific stuff */
 	struct hda_verb *init;
@@ -220,15 +238,16 @@ struct sigmatel_spec {
 	/* i/o switches */
 	unsigned int io_switch[2];
 	unsigned int clfe_swap;
-	unsigned int hp_switch; /* NID of HP as line-out */
+	hda_nid_t line_switch;	/* shared line-in for input and output */
+	hda_nid_t mic_switch;	/* shared mic-in for input and output */
+	hda_nid_t hp_switch; /* NID of HP as line-out */
 	unsigned int aloopback;
 
 	struct hda_pcm pcm_rec[2];	/* PCM information */
 
 	/* dynamic controls and input_mux */
 	struct auto_pin_cfg autocfg;
-	unsigned int num_kctl_alloc, num_kctl_used;
-	struct snd_kcontrol_new *kctl_alloc;
+	struct snd_array kctls;
 	struct hda_input_mux private_dimux;
 	struct hda_input_mux private_imux;
 	struct hda_input_mux private_smux;
@@ -272,9 +291,6 @@ static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
 };
 
 #define STAC92HD73_DAC_COUNT 5
-static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = {
-	0x15, 0x16, 0x17, 0x18, 0x19,
-};
 
 static hda_nid_t stac92hd73xx_mux_nids[4] = {
 	0x28, 0x29, 0x2a, 0x2b,
@@ -293,11 +309,7 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
 	0x11, 0x12, 0
 };
 
-#define STAC92HD81_DAC_COUNT 2
 #define STAC92HD83_DAC_COUNT 3
-static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = {
-	0x13, 0x14, 0x22,
-};
 
 static hda_nid_t stac92hd83xxx_dmux_nids[2] = {
 	0x17, 0x18,
@@ -339,10 +351,6 @@ static hda_nid_t stac92hd71bxx_smux_nids[2] = {
 	0x24, 0x25,
 };
 
-static hda_nid_t stac92hd71bxx_dac_nids[1] = {
-	0x10, /*0x11, */
-};
-
 #define STAC92HD71BXX_NUM_DMICS	2
 static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = {
 	0x18, 0x19, 0
@@ -574,12 +582,12 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 		else
 			nid = codec->slave_dig_outs[smux_idx - 1];
 		if (spec->cur_smux[smux_idx] == smux->num_items - 1)
-			val = AMP_OUT_MUTE;
+			val = HDA_AMP_MUTE;
 		else
-			val = AMP_OUT_UNMUTE;
+			val = 0;
 		/* un/mute SPDIF out */
-		snd_hda_codec_write_cache(codec, nid, 0,
-			AC_VERB_SET_AMP_GAIN_MUTE, val);
+		snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, val);
 	}
 	return 0;
 }
@@ -744,10 +752,6 @@ static struct hda_verb stac9200_eapd_init[] = {
 static struct hda_verb stac92hd73xx_6ch_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup audio connections */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x11, AC_VERB_SET_CONNECT_SEL, 0x02},
 	/* setup adcs to point to mixer */
 	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
 	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
@@ -766,10 +770,6 @@ static struct hda_verb dell_eq_core_init[] = {
 	/* set master volume to max value without distortion
 	 * and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
-	/* setup audio connections */
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02},
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01},
 	/* setup adcs to point to mixer */
 	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
 	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
@@ -783,10 +783,6 @@ static struct hda_verb dell_eq_core_init[] = {
 
 static struct hda_verb dell_m6_core_init[] = {
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup audio connections */
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02},
 	/* setup adcs to point to mixer */
 	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
 	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
@@ -801,13 +797,6 @@ static struct hda_verb dell_m6_core_init[] = {
 static struct hda_verb stac92hd73xx_8ch_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup audio connections */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x11, AC_VERB_SET_CONNECT_SEL, 0x02},
-	/* connect hp ports to dac3 */
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03},
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03},
 	/* setup adcs to point to mixer */
 	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
 	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
@@ -825,15 +814,8 @@ static struct hda_verb stac92hd73xx_8ch_core_init[] = {
 static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup audio connections */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 },
 	/* dac3 is connected to import3 mux */
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f},
-	/* connect hp ports to dac4 */
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04},
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04},
 	/* setup adcs to point to mixer */
 	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
 	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
@@ -865,8 +847,6 @@ static struct hda_verb stac92hd83xxx_core_init[] = {
 static struct hda_verb stac92hd71bxx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* connect headphone jack to dac1 */
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
 	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -886,8 +866,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* connect headphone jack to dac1 */
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
 	/* unmute right and left channels for nodes 0x0a, 0xd */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1087,21 +1065,21 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT),
 
-	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT),
 
 	/*
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT),
 	*/
 	{ } /* end */
 };
@@ -1240,9 +1218,14 @@ static const char *slave_sws[] = {
 	NULL
 };
 
+static void stac92xx_free_kctls(struct hda_codec *codec);
+static int stac92xx_add_jack(struct hda_codec *codec, hda_nid_t nid, int type);
+
 static int stac92xx_build_controls(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid;
 	int err;
 	int i;
 
@@ -1257,7 +1240,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 	}
 	if (spec->num_dmuxes > 0) {
 		stac_dmux_mixer.count = spec->num_dmuxes;
-		err = snd_ctl_add(codec->bus->card,
+		err = snd_hda_ctl_add(codec,
 				  snd_ctl_new1(&stac_dmux_mixer, codec));
 		if (err < 0)
 			return err;
@@ -1273,7 +1256,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			spec->spdif_mute = 1;
 		}
 		stac_smux_mixer.count = spec->num_smuxes;
-		err = snd_ctl_add(codec->bus->card,
+		err = snd_hda_ctl_add(codec,
 				  snd_ctl_new1(&stac_smux_mixer, codec));
 		if (err < 0)
 			return err;
@@ -1312,6 +1295,37 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	stac92xx_free_kctls(codec); /* no longer needed */
+
+	/* create jack input elements */
+	if (spec->hp_detect) {
+		for (i = 0; i < cfg->hp_outs; i++) {
+			int type = SND_JACK_HEADPHONE;
+			nid = cfg->hp_pins[i];
+			/* jack detection */
+			if (cfg->hp_outs == i)
+				type |= SND_JACK_LINEOUT;
+			err = stac92xx_add_jack(codec, nid, type);
+			if (err < 0)
+				return err;
+		}
+	}
+	for (i = 0; i < cfg->line_outs; i++) {
+		err = stac92xx_add_jack(codec, cfg->line_out_pins[i],
+					SND_JACK_LINEOUT);
+		if (err < 0)
+			return err;
+	}
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		nid = cfg->input_pins[i];
+		if (nid) {
+			err = stac92xx_add_jack(codec, nid,
+						SND_JACK_MICROPHONE);
+			if (err < 0)
+				return err;
+		}
+	}
+
 	return 0;	
 }
 
@@ -1720,6 +1734,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP dv5", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
 		      "HP dv7", STAC_HP_M4),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
+		      "HP dv7", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
 				"unknown HP", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
@@ -2203,12 +2219,11 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
 	int i;
 	struct sigmatel_spec *spec = codec->spec;
 	
-	if (! spec->bios_pin_configs) {
-		spec->bios_pin_configs = kcalloc(spec->num_pins,
-		                                 sizeof(*spec->bios_pin_configs), GFP_KERNEL);
-		if (! spec->bios_pin_configs)
-			return -ENOMEM;
-	}
+	kfree(spec->pin_configs);
+	spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs),
+				    GFP_KERNEL);
+	if (!spec->pin_configs)
+		return -ENOMEM;
 	
 	for (i = 0; i < spec->num_pins; i++) {
 		hda_nid_t nid = spec->pin_nids[i];
@@ -2218,7 +2233,7 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
 			AC_VERB_GET_CONFIG_DEFAULT, 0x00);	
 		snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n",
 					nid, pin_cfg);
-		spec->bios_pin_configs[i] = pin_cfg;
+		spec->pin_configs[i] = pin_cfg;
 	}
 	
 	return 0;
@@ -2260,6 +2275,39 @@ static void stac92xx_set_config_regs(struct hda_codec *codec)
 					spec->pin_configs[i]);
 }
 
+static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (!pins)
+		return stac92xx_save_bios_config_regs(codec);
+
+	kfree(spec->pin_configs);
+	spec->pin_configs = kmemdup(pins,
+				    spec->num_pins * sizeof(*pins),
+				    GFP_KERNEL);
+	if (!spec->pin_configs)
+		return -ENOMEM;
+
+	stac92xx_set_config_regs(codec);
+	return 0;
+}
+
+static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid,
+				   unsigned int cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_pins; i++) {
+		if (spec->pin_nids[i] == nid) {
+			spec->pin_configs[i] = cfg;
+			stac92xx_set_config_reg(codec, nid, cfg);
+			break;
+		}
+	}
+}
+
 /*
  * Analog playback callbacks
  */
@@ -2337,7 +2385,7 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
 
 	if (spec->powerdown_adcs) {
 		msleep(40);
-		snd_hda_codec_write_cache(codec, nid, 0,
+		snd_hda_codec_write(codec, nid, 0,
 			AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
 	}
 	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
@@ -2353,7 +2401,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 
 	snd_hda_codec_cleanup_stream(codec, nid);
 	if (spec->powerdown_adcs)
-		snd_hda_codec_write_cache(codec, nid, 0,
+		snd_hda_codec_write(codec, nid, 0,
 			AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 	return 0;
 }
@@ -2485,6 +2533,9 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid,
+				   unsigned char type);
+
 static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
 			struct snd_ctl_elem_value *ucontrol)
 {
@@ -2497,7 +2548,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
 	/* check to be sure that the ports are upto date with
 	 * switch changes
 	 */
-	codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+	stac_issue_unsol_event(codec, nid, STAC_HP_EVENT);
 
 	return 1;
 }
@@ -2537,7 +2588,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	 * appropriately according to the pin direction
 	 */
 	if (spec->hp_detect)
-		codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+		stac_issue_unsol_event(codec, nid, STAC_HP_EVENT);
 
         return 1;
 }
@@ -2632,28 +2683,16 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
 {
 	struct snd_kcontrol_new *knew;
 
-	if (spec->num_kctl_used >= spec->num_kctl_alloc) {
-		int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
-
-		knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
-		if (! knew)
-			return -ENOMEM;
-		if (spec->kctl_alloc) {
-			memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
-			kfree(spec->kctl_alloc);
-		}
-		spec->kctl_alloc = knew;
-		spec->num_kctl_alloc = num;
-	}
-
-	knew = &spec->kctl_alloc[spec->num_kctl_used];
+	snd_array_init(&spec->kctls, sizeof(*knew), 32);
+	knew = snd_array_new(&spec->kctls);
+	if (!knew)
+		return -ENOMEM;
 	*knew = *ktemp;
 	knew->index = idx;
 	knew->name = kstrdup(name, GFP_KERNEL);
 	if (!knew->name)
 		return -ENOMEM;
 	knew->private_value = val;
-	spec->num_kctl_used++;
 	return 0;
 }
 
@@ -2674,69 +2713,52 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
 	return stac92xx_add_control_idx(spec, type, 0, name, val);
 }
 
-/* flag inputs as additional dynamic lineouts */
-static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg)
+/* check whether the line-input can be used as line-out */
+static hda_nid_t check_line_out_switch(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	unsigned int wcaps, wtype;
-	int i, num_dacs = 0;
-	
-	/* use the wcaps cache to count all DACs available for line-outs */
-	for (i = 0; i < codec->num_nodes; i++) {
-		wcaps = codec->wcaps[i];
-		wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
-
-		if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL))
-			num_dacs++;
-	}
-
-	snd_printdd("%s: total dac count=%d\n", __func__, num_dacs);
-	
-	switch (cfg->line_outs) {
-	case 3:
-		/* add line-in as side */
-		if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 3) {
-			cfg->line_out_pins[cfg->line_outs] =
-				cfg->input_pins[AUTO_PIN_LINE];
-			spec->line_switch = 1;
-			cfg->line_outs++;
-		}
-		break;
-	case 2:
-		/* add line-in as clfe and mic as side */
-		if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 2) {
-			cfg->line_out_pins[cfg->line_outs] =
-				cfg->input_pins[AUTO_PIN_LINE];
-			spec->line_switch = 1;
-			cfg->line_outs++;
-		}
-		if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 3) {
-			cfg->line_out_pins[cfg->line_outs] =
-				cfg->input_pins[AUTO_PIN_MIC];
-			spec->mic_switch = 1;
-			cfg->line_outs++;
-		}
-		break;
-	case 1:
-		/* add line-in as surr and mic as clfe */
-		if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 1) {
-			cfg->line_out_pins[cfg->line_outs] =
-				cfg->input_pins[AUTO_PIN_LINE];
-			spec->line_switch = 1;
-			cfg->line_outs++;
-		}
-		if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 2) {
-			cfg->line_out_pins[cfg->line_outs] =
-				cfg->input_pins[AUTO_PIN_MIC];
-			spec->mic_switch = 1;
-			cfg->line_outs++;
-		}
-		break;
-	}
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid;
+	unsigned int pincap;
 
+	if (cfg->line_out_type != AUTO_PIN_LINE_OUT)
+		return 0;
+	nid = cfg->input_pins[AUTO_PIN_LINE];
+	pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	if (pincap & AC_PINCAP_OUT)
+		return nid;
 	return 0;
 }
 
+/* check whether the mic-input can be used as line-out */
+static hda_nid_t check_mic_out_switch(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	unsigned int def_conf, pincap;
+	unsigned int mic_pin;
+
+	if (cfg->line_out_type != AUTO_PIN_LINE_OUT)
+		return 0;
+	mic_pin = AUTO_PIN_MIC;
+	for (;;) {
+		hda_nid_t nid = cfg->input_pins[mic_pin];
+		def_conf = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		/* some laptops have an internal analog microphone
+		 * which can't be used as a output */
+		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
+			pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+			if (pincap & AC_PINCAP_OUT)
+				return nid;
+		}
+		if (mic_pin == AUTO_PIN_MIC)
+			mic_pin = AUTO_PIN_FRONT_MIC;
+		else
+			break;
+	}
+	return 0;
+}
 
 static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
 {
@@ -2750,6 +2772,52 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
 	return 0;
 }
 
+static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+	int i;
+	if (is_in_dac_nids(spec, nid))
+		return 1;
+	for (i = 0; i < spec->autocfg.hp_outs; i++)
+		if (spec->hp_dacs[i] == nid)
+			return 1;
+	for (i = 0; i < spec->autocfg.speaker_outs; i++)
+		if (spec->speaker_dacs[i] == nid)
+			return 1;
+	return 0;
+}
+
+static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int j, conn_len;
+	hda_nid_t conn[HDA_MAX_CONNECTIONS];
+	unsigned int wcaps, wtype;
+
+	conn_len = snd_hda_get_connections(codec, nid, conn,
+					   HDA_MAX_CONNECTIONS);
+	for (j = 0; j < conn_len; j++) {
+		wcaps = snd_hda_param_read(codec, conn[j],
+					   AC_PAR_AUDIO_WIDGET_CAP);
+		wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		/* we check only analog outputs */
+		if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL))
+			continue;
+		/* if this route has a free DAC, assign it */
+		if (!check_all_dac_nids(spec, conn[j])) {
+			if (conn_len > 1) {
+				/* select this DAC in the pin's input mux */
+				snd_hda_codec_write_cache(codec, nid, 0,
+						  AC_VERB_SET_CONNECT_SEL, j);
+			}
+			return conn[j];
+		}
+	}
+	return 0;
+}
+
+static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid);
+static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid);
+
 /*
  * Fill in the dac_nids table from the parsed pin configuration
  * This function only works when every pin in line_out_pins[]
@@ -2757,31 +2825,17 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
  * codecs are not connected directly to a DAC, such as the 9200
  * and 9202/925x. For those, dac_nids[] must be hard-coded.
  */
-static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
-				       struct auto_pin_cfg *cfg)
+static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int i, j, conn_len = 0; 
-	hda_nid_t nid, conn[HDA_MAX_CONNECTIONS];
-	unsigned int wcaps, wtype;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+	hda_nid_t nid, dac;
 	
 	for (i = 0; i < cfg->line_outs; i++) {
 		nid = cfg->line_out_pins[i];
-		conn_len = snd_hda_get_connections(codec, nid, conn,
-						   HDA_MAX_CONNECTIONS);
-		for (j = 0; j < conn_len; j++) {
-			wcaps = snd_hda_param_read(codec, conn[j],
-						   AC_PAR_AUDIO_WIDGET_CAP);
-			wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
-			if (wtype != AC_WID_AUD_OUT ||
-			    (wcaps & AC_WCAP_DIGITAL))
-				continue;
-			/* conn[j] is a DAC routed to this line-out */
-			if (!is_in_dac_nids(spec, conn[j]))
-				break;
-		}
-
-		if (j == conn_len) {
+		dac = get_unassigned_dac(codec, nid);
+		if (!dac) {
 			if (spec->multiout.num_dacs > 0) {
 				/* we have already working output pins,
 				 * so let's drop the broken ones again
@@ -2795,24 +2849,64 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
 				   __func__, nid);
 			return -ENODEV;
 		}
+		add_spec_dacs(spec, dac);
+	}
 
-		spec->multiout.dac_nids[i] = conn[j];
-		spec->multiout.num_dacs++;
-		if (conn_len > 1) {
-			/* select this DAC in the pin's input mux */
-			snd_hda_codec_write_cache(codec, nid, 0,
-						  AC_VERB_SET_CONNECT_SEL, j);
-
+	/* add line-in as output */
+	nid = check_line_out_switch(codec);
+	if (nid) {
+		dac = get_unassigned_dac(codec, nid);
+		if (dac) {
+			snd_printdd("STAC: Add line-in 0x%x as output %d\n",
+				    nid, cfg->line_outs);
+			cfg->line_out_pins[cfg->line_outs] = nid;
+			cfg->line_outs++;
+			spec->line_switch = nid;
+			add_spec_dacs(spec, dac);
+		}
+	}
+	/* add mic as output */
+	nid = check_mic_out_switch(codec);
+	if (nid) {
+		dac = get_unassigned_dac(codec, nid);
+		if (dac) {
+			snd_printdd("STAC: Add mic-in 0x%x as output %d\n",
+				    nid, cfg->line_outs);
+			cfg->line_out_pins[cfg->line_outs] = nid;
+			cfg->line_outs++;
+			spec->mic_switch = nid;
+			add_spec_dacs(spec, dac);
 		}
 	}
 
-	snd_printd("dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+	for (i = 0; i < cfg->hp_outs; i++) {
+		nid = cfg->hp_pins[i];
+		dac = get_unassigned_dac(codec, nid);
+		if (dac) {
+			if (!spec->multiout.hp_nid)
+				spec->multiout.hp_nid = dac;
+			else
+				add_spec_extra_dacs(spec, dac);
+		}
+		spec->hp_dacs[i] = dac;
+	}
+
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		nid = cfg->speaker_pins[i];
+		dac = get_unassigned_dac(codec, nid);
+		if (dac)
+			add_spec_extra_dacs(spec, dac);
+		spec->speaker_dacs[i] = dac;
+	}
+
+	snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
 		   spec->multiout.num_dacs,
 		   spec->multiout.dac_nids[0],
 		   spec->multiout.dac_nids[1],
 		   spec->multiout.dac_nids[2],
 		   spec->multiout.dac_nids[3],
 		   spec->multiout.dac_nids[4]);
+
 	return 0;
 }
 
@@ -2837,9 +2931,7 @@ static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_
 
 static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 {
-	if (!spec->multiout.hp_nid)
-		spec->multiout.hp_nid = nid;
-	else if (spec->multiout.num_dacs > 4) {
+	if (spec->multiout.num_dacs > 4) {
 		printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid);
 		return 1;
 	} else {
@@ -2849,35 +2941,47 @@ static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 	return 0;
 }
 
-static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
+static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 {
-	if (is_in_dac_nids(spec, nid))
-		return 1;
+	int i;
+	for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) {
+		if (!spec->multiout.extra_out_nid[i]) {
+			spec->multiout.extra_out_nid[i] = nid;
+			return 0;
+		}
+	}
+	printk(KERN_WARNING "stac92xx: No space for extra DAC 0x%x\n", nid);
+	return 1;
+}
+
+static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+	int i;
+
+	if (spec->autocfg.line_outs != 1)
+		return 0;
 	if (spec->multiout.hp_nid == nid)
-		return 1;
-	return 0;
+		return 0;
+	for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++)
+		if (spec->multiout.extra_out_nid[i] == nid)
+			return 0;
+	return 1;
 }
 
 /* add playback controls from the parsed DAC table */
 static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 					       const struct auto_pin_cfg *cfg)
 {
+	struct sigmatel_spec *spec = codec->spec;
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
 	hda_nid_t nid = 0;
 	int i, err;
+	unsigned int wid_caps;
 
-	struct sigmatel_spec *spec = codec->spec;
-	unsigned int wid_caps, pincap;
-
-
-	for (i = 0; i < cfg->line_outs && i < spec->multiout.num_dacs; i++) {
-		if (!spec->multiout.dac_nids[i])
-			continue;
-
+	for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) {
 		nid = spec->multiout.dac_nids[i];
-
 		if (i == 2) {
 			/* Center/LFE */
 			err = create_controls(spec, "Center", nid, 1);
@@ -2899,16 +3003,24 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 			}
 
 		} else {
-			err = create_controls(spec, chname[i], nid, 3);
+			const char *name = chname[i];
+			/* if it's a single DAC, assign a better name */
+			if (!i && is_unique_dac(spec, nid)) {
+				switch (cfg->line_out_type) {
+				case AUTO_PIN_HP_OUT:
+					name = "Headphone";
+					break;
+				case AUTO_PIN_SPEAKER_OUT:
+					name = "Speaker";
+					break;
+				}
+			}
+			err = create_controls(spec, name, nid, 3);
 			if (err < 0)
 				return err;
 		}
 	}
 
-	if ((spec->multiout.num_dacs - cfg->line_outs) > 0 &&
-	    cfg->hp_outs == 1 && !spec->multiout.hp_nid)
-		spec->multiout.hp_nid = nid;
-
 	if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) {
 		err = stac92xx_add_control(spec,
 			STAC_CTL_WIDGET_HP_SWITCH,
@@ -2919,45 +3031,19 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 	}
 
 	if (spec->line_switch) {
-		nid = cfg->input_pins[AUTO_PIN_LINE];
-		pincap = snd_hda_param_read(codec, nid,
-						AC_PAR_PIN_CAP);
-		if (pincap & AC_PINCAP_OUT) {
-			err = stac92xx_add_control(spec,
-				STAC_CTL_WIDGET_IO_SWITCH,
-				"Line In as Output Switch", nid << 8);
-			if (err < 0)
-				return err;
-		}
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH,
+					   "Line In as Output Switch",
+					   spec->line_switch << 8);
+		if (err < 0)
+			return err;
 	}
 
 	if (spec->mic_switch) {
-		unsigned int def_conf;
-		unsigned int mic_pin = AUTO_PIN_MIC;
-again:
-		nid = cfg->input_pins[mic_pin];
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-						AC_VERB_GET_CONFIG_DEFAULT, 0);
-		/* some laptops have an internal analog microphone
-		 * which can't be used as a output */
-		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
-			pincap = snd_hda_param_read(codec, nid,
-							AC_PAR_PIN_CAP);
-			if (pincap & AC_PINCAP_OUT) {
-				err = stac92xx_add_control(spec,
-					STAC_CTL_WIDGET_IO_SWITCH,
-					"Mic as Output Switch", (nid << 8) | 1);
-				nid = snd_hda_codec_read(codec, nid, 0,
-					 AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
-				if (!check_in_dac_nids(spec, nid))
-					add_spec_dacs(spec, nid);
-				if (err < 0)
-					return err;
-			}
-		} else if (mic_pin == AUTO_PIN_MIC) {
-			mic_pin = AUTO_PIN_FRONT_MIC;
-			goto again;
-		}
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH,
+					   "Mic as Output Switch",
+					   (spec->mic_switch << 8) | 1);
+		if (err < 0)
+			return err;
 	}
 
 	return 0;
@@ -2969,55 +3055,39 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 {
 	struct sigmatel_spec *spec = codec->spec;
 	hda_nid_t nid;
-	int i, old_num_dacs, err;
+	int i, err, nums;
 
-	old_num_dacs = spec->multiout.num_dacs;
+	nums = 0;
 	for (i = 0; i < cfg->hp_outs; i++) {
+		static const char *pfxs[] = {
+			"Headphone", "Headphone2", "Headphone3",
+		};
 		unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]);
 		if (wid_caps & AC_WCAP_UNSOL_CAP)
 			spec->hp_detect = 1;
-		nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0,
-					 AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
-		if (check_in_dac_nids(spec, nid))
-			nid = 0;
-		if (! nid)
+		if (nums >= ARRAY_SIZE(pfxs))
 			continue;
-		add_spec_dacs(spec, nid);
+		nid = spec->hp_dacs[i];
+		if (!nid)
+			continue;
+		err = create_controls(spec, pfxs[nums++], nid, 3);
+		if (err < 0)
+			return err;
 	}
+	nums = 0;
 	for (i = 0; i < cfg->speaker_outs; i++) {
-		nid = snd_hda_codec_read(codec, cfg->speaker_pins[i], 0,
-					 AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
-		if (check_in_dac_nids(spec, nid))
-			nid = 0;
-		if (! nid)
-			continue;
-		add_spec_dacs(spec, nid);
-	}
-	for (i = 0; i < cfg->line_outs; i++) {
-		nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0,
-					AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
-		if (check_in_dac_nids(spec, nid))
-			nid = 0;
-		if (! nid)
-			continue;
-		add_spec_dacs(spec, nid);
-	}
-	for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) {
 		static const char *pfxs[] = {
 			"Speaker", "External Speaker", "Speaker2",
 		};
-		err = create_controls(spec, pfxs[i - old_num_dacs],
-				      spec->multiout.dac_nids[i], 3);
+		if (nums >= ARRAY_SIZE(pfxs))
+			continue;
+		nid = spec->speaker_dacs[i];
+		if (!nid)
+			continue;
+		err = create_controls(spec, pfxs[nums++], nid, 3);
 		if (err < 0)
 			return err;
 	}
-	if (spec->multiout.hp_nid) {
-		err = create_controls(spec, "Headphone",
-				      spec->multiout.hp_nid, 3);
-		if (err < 0)
-			return err;
-	}
-
 	return 0;
 }
 
@@ -3355,7 +3425,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 {
 	struct sigmatel_spec *spec = codec->spec;
 	int err;
-	int hp_speaker_swap = 0;
 
 	if ((err = snd_hda_parse_pin_def_config(codec,
 						&spec->autocfg,
@@ -3373,13 +3442,15 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		 * speaker_outs so that the following routines can handle
 		 * HP pins as primary outputs.
 		 */
+		snd_printdd("stac92xx: Enabling multi-HPs workaround\n");
 		memcpy(spec->autocfg.speaker_pins, spec->autocfg.line_out_pins,
 		       sizeof(spec->autocfg.line_out_pins));
 		spec->autocfg.speaker_outs = spec->autocfg.line_outs;
 		memcpy(spec->autocfg.line_out_pins, spec->autocfg.hp_pins,
 		       sizeof(spec->autocfg.hp_pins));
 		spec->autocfg.line_outs = spec->autocfg.hp_outs;
-		hp_speaker_swap = 1;
+		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
+		spec->autocfg.hp_outs = 0;
 	}
 	if (spec->autocfg.mono_out_pin) {
 		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
@@ -3431,11 +3502,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 					 AC_PINCTL_OUT_EN);
 	}
 
-	if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0)
-		return err;
-	if (spec->multiout.num_dacs == 0)
-		if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
+	if (!spec->multiout.num_dacs) {
+		err = stac92xx_auto_fill_dac_nids(codec);
+		if (err < 0)
 			return err;
+	}
 
 	err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg);
 
@@ -3473,19 +3544,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	}
 #endif
 
-	if (hp_speaker_swap == 1) {
-		/* Restore the hp_outs and line_outs */
-		memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins,
-		       sizeof(spec->autocfg.line_out_pins));
-		spec->autocfg.hp_outs = spec->autocfg.line_outs;
-		memcpy(spec->autocfg.line_out_pins, spec->autocfg.speaker_pins,
-		       sizeof(spec->autocfg.speaker_pins));
-		spec->autocfg.line_outs = spec->autocfg.speaker_outs;
-		memset(spec->autocfg.speaker_pins, 0,
-		       sizeof(spec->autocfg.speaker_pins));
-		spec->autocfg.speaker_outs = 0;
-	}
-
 	err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg);
 
 	if (err < 0)
@@ -3530,11 +3588,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (dig_in && spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = dig_in;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->input_mux = &spec->private_imux;
-	spec->dinput_mux = &spec->private_dimux;
+	if (!spec->dinput_mux)
+		spec->dinput_mux = &spec->private_dimux;
 	spec->sinput_mux = &spec->private_smux;
 	spec->mono_mux = &spec->private_mono_mux;
 	spec->amp_mux = &spec->private_amp_mux;
@@ -3638,8 +3697,8 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = 0x04;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->input_mux = &spec->private_imux;
 	spec->dinput_mux = &spec->private_dimux;
@@ -3683,13 +3742,101 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask,
 			   AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */
 }
 
-static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
-			      unsigned int event)
+static int stac92xx_add_jack(struct hda_codec *codec,
+		hda_nid_t nid, int type)
 {
-	if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)
-		snd_hda_codec_write_cache(codec, nid, 0,
-					  AC_VERB_SET_UNSOLICITED_ENABLE,
-					  (AC_USRSP_EN | event));
+#ifdef CONFIG_SND_JACK
+	struct sigmatel_spec *spec = codec->spec;
+	struct sigmatel_jack *jack;
+	int def_conf = snd_hda_codec_read(codec, nid,
+			0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+	int connectivity = get_defcfg_connect(def_conf);
+	char name[32];
+
+	if (connectivity && connectivity != AC_JACK_PORT_FIXED)
+		return 0;
+
+	snd_array_init(&spec->jacks, sizeof(*jack), 32);
+	jack = snd_array_new(&spec->jacks);
+	if (!jack)
+		return -ENOMEM;
+	jack->nid = nid;
+	jack->type = type;
+
+	sprintf(name, "%s at %s %s Jack",
+		snd_hda_get_jack_type(def_conf),
+		snd_hda_get_jack_connectivity(def_conf),
+		snd_hda_get_jack_location(def_conf));
+
+	return snd_jack_new(codec->bus->card, name, type, &jack->jack);
+#else
+	return 0;
+#endif
+}
+
+static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid,
+			  unsigned char type, int data)
+{
+	struct sigmatel_event *event;
+
+	snd_array_init(&spec->events, sizeof(*event), 32);
+	event = snd_array_new(&spec->events);
+	if (!event)
+		return -ENOMEM;
+	event->nid = nid;
+	event->type = type;
+	event->tag = spec->events.used;
+	event->data = data;
+
+	return event->tag;
+}
+
+static struct sigmatel_event *stac_get_event(struct hda_codec *codec,
+					     hda_nid_t nid, unsigned char type)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct sigmatel_event *event = spec->events.list;
+	int i;
+
+	for (i = 0; i < spec->events.used; i++, event++) {
+		if (event->nid == nid && event->type == type)
+			return event;
+	}
+	return NULL;
+}
+
+static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec,
+						      unsigned char tag)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct sigmatel_event *event = spec->events.list;
+	int i;
+
+	for (i = 0; i < spec->events.used; i++, event++) {
+		if (event->tag == tag)
+			return event;
+	}
+	return NULL;
+}
+
+static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
+			      unsigned int type)
+{
+	struct sigmatel_event *event;
+	int tag;
+
+	if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
+		return;
+	event = stac_get_event(codec, nid, type);
+	if (event)
+		tag = event->tag;
+	else
+		tag = stac_add_event(codec->spec, nid, type, 0);
+	if (tag < 0)
+		return;
+	snd_hda_codec_write_cache(codec, nid, 0,
+				  AC_VERB_SET_UNSOLICITED_ENABLE,
+				  AC_USRSP_EN | tag);
 }
 
 static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
@@ -3709,9 +3856,8 @@ static void stac92xx_power_down(struct hda_codec *codec)
 	/* power down inactive DACs */
 	hda_nid_t *dac;
 	for (dac = spec->dac_list; *dac; dac++)
-		if (!is_in_dac_nids(spec, *dac) &&
-			spec->multiout.hp_nid != *dac)
-			snd_hda_codec_write_cache(codec, *dac, 0,
+		if (!check_all_dac_nids(spec, *dac))
+			snd_hda_codec_write(codec, *dac, 0,
 					AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 }
 
@@ -3730,7 +3876,7 @@ static int stac92xx_init(struct hda_codec *codec)
 	/* power down adcs initially */
 	if (spec->powerdown_adcs)
 		for (i = 0; i < spec->num_adcs; i++)
-			snd_hda_codec_write_cache(codec,
+			snd_hda_codec_write(codec,
 				spec->adc_nids[i], 0,
 				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 
@@ -3746,37 +3892,51 @@ static int stac92xx_init(struct hda_codec *codec)
 	/* set up pins */
 	if (spec->hp_detect) {
 		/* Enable unsolicited responses on the HP widget */
-		for (i = 0; i < cfg->hp_outs; i++)
-			enable_pin_detect(codec, cfg->hp_pins[i],
-					  STAC_HP_EVENT);
+		for (i = 0; i < cfg->hp_outs; i++) {
+			hda_nid_t nid = cfg->hp_pins[i];
+			enable_pin_detect(codec, nid, STAC_HP_EVENT);
+		}
 		/* force to enable the first line-out; the others are set up
 		 * in unsol_event
 		 */
 		stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
-					 AC_PINCTL_OUT_EN);
-		stac92xx_auto_init_hp_out(codec);
+				AC_PINCTL_OUT_EN);
 		/* fake event to set up pins */
-		codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0],
+				       STAC_HP_EVENT);
 	} else {
 		stac92xx_auto_init_multi_out(codec);
 		stac92xx_auto_init_hp_out(codec);
+		for (i = 0; i < cfg->hp_outs; i++)
+			stac_toggle_power_map(codec, cfg->hp_pins[i], 1);
 	}
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = cfg->input_pins[i];
 		if (nid) {
-			unsigned int pinctl;
+			unsigned int pinctl, conf;
 			if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) {
 				/* for mic pins, force to initialize */
 				pinctl = stac92xx_get_vref(codec, nid);
+				pinctl |= AC_PINCTL_IN_EN;
+				stac92xx_auto_set_pinctl(codec, nid, pinctl);
 			} else {
 				pinctl = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 				/* if PINCTL already set then skip */
-				if (pinctl & AC_PINCTL_IN_EN)
-					continue;
+				if (!(pinctl & AC_PINCTL_IN_EN)) {
+					pinctl |= AC_PINCTL_IN_EN;
+					stac92xx_auto_set_pinctl(codec, nid,
+								 pinctl);
+				}
+			}
+			conf = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+			if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) {
+				enable_pin_detect(codec, nid,
+						  STAC_INSERT_EVENT);
+				stac_issue_unsol_event(codec, nid,
+						       STAC_INSERT_EVENT);
 			}
-			pinctl |= AC_PINCTL_IN_EN;
-			stac92xx_auto_set_pinctl(codec, nid, pinctl);
 		}
 	}
 	for (i = 0; i < spec->num_dmics; i++)
@@ -3791,9 +3951,14 @@ static int stac92xx_init(struct hda_codec *codec)
 	for (i = 0; i < spec->num_pwrs; i++)  {
 		hda_nid_t nid = spec->pwr_nids[i];
 		int pinctl, def_conf;
-		int event = STAC_PWR_EVENT;
 
-		if (is_nid_hp_pin(cfg, nid) && spec->hp_detect)
+		/* power on when no jack detection is available */
+		if (!spec->hp_detect) {
+			stac_toggle_power_map(codec, nid, 1);
+			continue;
+		}
+
+		if (is_nid_hp_pin(cfg, nid))
 			continue; /* already has an unsol event */
 
 		pinctl = snd_hda_codec_read(codec, nid, 0,
@@ -3802,8 +3967,10 @@ static int stac92xx_init(struct hda_codec *codec)
 		 * any attempts on powering down a input port cause the
 		 * referenced VREF to act quirky.
 		 */
-		if (pinctl & AC_PINCTL_IN_EN)
+		if (pinctl & AC_PINCTL_IN_EN) {
+			stac_toggle_power_map(codec, nid, 1);
 			continue;
+		}
 		def_conf = snd_hda_codec_read(codec, nid, 0,
 					      AC_VERB_GET_CONFIG_DEFAULT, 0);
 		def_conf = get_defcfg_connect(def_conf);
@@ -3814,30 +3981,54 @@ static int stac92xx_init(struct hda_codec *codec)
 				stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
-		enable_pin_detect(codec, spec->pwr_nids[i], event | i);
-		codec->patch_ops.unsol_event(codec, (event | i) << 26);
+		if (!stac_get_event(codec, nid, STAC_INSERT_EVENT)) {
+			enable_pin_detect(codec, nid, STAC_PWR_EVENT);
+			stac_issue_unsol_event(codec, nid, STAC_PWR_EVENT);
+		}
 	}
 	if (spec->dac_list)
 		stac92xx_power_down(codec);
 	return 0;
 }
 
+static void stac92xx_free_jacks(struct hda_codec *codec)
+{
+#ifdef CONFIG_SND_JACK
+	/* free jack instances manually when clearing/reconfiguring */
+	struct sigmatel_spec *spec = codec->spec;
+	if (!codec->bus->shutdown && spec->jacks.list) {
+		struct sigmatel_jack *jacks = spec->jacks.list;
+		int i;
+		for (i = 0; i < spec->jacks.used; i++)
+			snd_device_free(codec->bus->card, &jacks[i].jack);
+	}
+	snd_array_free(&spec->jacks);
+#endif
+}
+
+static void stac92xx_free_kctls(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (spec->kctls.list) {
+		struct snd_kcontrol_new *kctl = spec->kctls.list;
+		int i;
+		for (i = 0; i < spec->kctls.used; i++)
+			kfree(kctl[i].name);
+	}
+	snd_array_free(&spec->kctls);
+}
+
 static void stac92xx_free(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int i;
 
 	if (! spec)
 		return;
 
-	if (spec->kctl_alloc) {
-		for (i = 0; i < spec->num_kctl_used; i++)
-			kfree(spec->kctl_alloc[i].name);
-		kfree(spec->kctl_alloc);
-	}
-
-	if (spec->bios_pin_configs)
-		kfree(spec->bios_pin_configs);
+	kfree(spec->pin_configs);
+	stac92xx_free_jacks(codec);
+	snd_array_free(&spec->events);
 
 	kfree(spec);
 	snd_hda_detach_beep_device(codec);
@@ -3856,11 +4047,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
 		 * "xxx as Output" mixer switch
 		 */
 		struct sigmatel_spec *spec = codec->spec;
-		struct auto_pin_cfg *cfg = &spec->autocfg;
-		if ((nid == cfg->input_pins[AUTO_PIN_LINE] &&
-		     spec->line_switch) ||
-		    (nid == cfg->input_pins[AUTO_PIN_MIC] &&
-		     spec->mic_switch))
+		if (nid == spec->line_switch || nid == spec->mic_switch)
 			return;
 	}
 
@@ -3884,20 +4071,13 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
 			pin_ctl & ~flag);
 }
 
-static int get_hp_pin_presence(struct hda_codec *codec, hda_nid_t nid)
+static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
 {
 	if (!nid)
 		return 0;
 	if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00)
-	    & (1 << 31)) {
-		unsigned int pinctl;
-		pinctl = snd_hda_codec_read(codec, nid, 0,
-					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		if (pinctl & AC_PINCTL_IN_EN)
-			return 0; /* mic- or line-input */
-		else
-			return 1; /* HP-output */
-	}
+	    & (1 << 31))
+		return 1;
 	return 0;
 }
 
@@ -3909,11 +4089,9 @@ static int no_hp_sensing(struct sigmatel_spec *spec, int i)
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 
 	/* ignore sensing of shared line and mic jacks */
-	if (spec->line_switch &&
-	    cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_LINE])
+	if (cfg->hp_pins[i] == spec->line_switch)
 		return 1;
-	if (spec->mic_switch &&
-	    cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_MIC])
+	if (cfg->hp_pins[i] == spec->mic_switch)
 		return 1;
 	/* ignore if the pin is set as line-out */
 	if (cfg->hp_pins[i] == spec->hp_switch)
@@ -3921,7 +4099,7 @@ static int no_hp_sensing(struct sigmatel_spec *spec, int i)
 	return 0;
 }
 
-static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
+static void stac92xx_hp_detect(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
@@ -3937,7 +4115,14 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
 			break;
 		if (no_hp_sensing(spec, i))
 			continue;
-		presence = get_hp_pin_presence(codec, cfg->hp_pins[i]);
+		presence = get_pin_presence(codec, cfg->hp_pins[i]);
+		if (presence) {
+			unsigned int pinctl;
+			pinctl = snd_hda_codec_read(codec, cfg->hp_pins[i], 0,
+					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+			if (pinctl & AC_PINCTL_IN_EN)
+				presence = 0; /* mic- or line-input */
+		}
 	}
 
 	if (presence) {
@@ -4014,50 +4199,145 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
 
 static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 {
-	stac_toggle_power_map(codec, nid, get_hp_pin_presence(codec, nid));
+	stac_toggle_power_map(codec, nid, get_pin_presence(codec, nid));
+}
+
+static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct sigmatel_jack *jacks = spec->jacks.list;
+
+	if (jacks) {
+		int i;
+		for (i = 0; i < spec->jacks.used; i++) {
+			if (jacks->nid == nid) {
+				unsigned int pin_ctl =
+					snd_hda_codec_read(codec, nid,
+					0, AC_VERB_GET_PIN_WIDGET_CONTROL,
+					 0x00);
+				int type = jacks->type;
+				if (type == (SND_JACK_LINEOUT
+						| SND_JACK_HEADPHONE))
+					type = (pin_ctl & AC_PINCTL_HP_EN)
+					? SND_JACK_HEADPHONE : SND_JACK_LINEOUT;
+				snd_jack_report(jacks->jack,
+					get_pin_presence(codec, nid)
+					? type : 0);
+			}
+			jacks++;
+		}
+	}
+}
+
+static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid,
+				   unsigned char type)
+{
+	struct sigmatel_event *event = stac_get_event(codec, nid, type);
+	if (!event)
+		return;
+	codec->patch_ops.unsol_event(codec, (unsigned)event->tag << 26);
 }
 
 static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int idx = res >> 26 & 0x0f;
+	struct sigmatel_event *event;
+	int tag, data;
 
-	switch ((res >> 26) & 0x70) {
+	tag = (res >> 26) & 0x7f;
+	event = stac_get_event_from_tag(codec, tag);
+	if (!event)
+		return;
+
+	switch (event->type) {
 	case STAC_HP_EVENT:
-		stac92xx_hp_detect(codec, res);
+		stac92xx_hp_detect(codec);
 		/* fallthru */
+	case STAC_INSERT_EVENT:
 	case STAC_PWR_EVENT:
 		if (spec->num_pwrs > 0)
-			stac92xx_pin_sense(codec, idx);
+			stac92xx_pin_sense(codec, event->nid);
+		stac92xx_report_jack(codec, event->nid);
 		break;
-	case STAC_VREF_EVENT: {
-		int data = snd_hda_codec_read(codec, codec->afg, 0,
-			AC_VERB_GET_GPIO_DATA, 0);
+	case STAC_VREF_EVENT:
+		data = snd_hda_codec_read(codec, codec->afg, 0,
+					  AC_VERB_GET_GPIO_DATA, 0);
 		/* toggle VREF state based on GPIOx status */
 		snd_hda_codec_write(codec, codec->afg, 0, 0x7e0,
-			!!(data & (1 << idx)));
+				    !!(data & (1 << event->data)));
 		break;
-		}
 	}
 }
 
+#ifdef CONFIG_PROC_FS
+static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
+			       struct hda_codec *codec, hda_nid_t nid)
+{
+	if (nid == codec->afg)
+		snd_iprintf(buffer, "Power-Map: 0x%02x\n", 
+			    snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0));
+}
+
+static void analog_loop_proc_hook(struct snd_info_buffer *buffer,
+				  struct hda_codec *codec,
+				  unsigned int verb)
+{
+	snd_iprintf(buffer, "Analog Loopback: 0x%02x\n",
+		    snd_hda_codec_read(codec, codec->afg, 0, verb, 0));
+}
+
+/* stac92hd71bxx, stac92hd73xx */
+static void stac92hd7x_proc_hook(struct snd_info_buffer *buffer,
+				 struct hda_codec *codec, hda_nid_t nid)
+{
+	stac92hd_proc_hook(buffer, codec, nid);
+	if (nid == codec->afg)
+		analog_loop_proc_hook(buffer, codec, 0xfa0);
+}
+
+static void stac9205_proc_hook(struct snd_info_buffer *buffer,
+			       struct hda_codec *codec, hda_nid_t nid)
+{
+	if (nid == codec->afg)
+		analog_loop_proc_hook(buffer, codec, 0xfe0);
+}
+
+static void stac927x_proc_hook(struct snd_info_buffer *buffer,
+			       struct hda_codec *codec, hda_nid_t nid)
+{
+	if (nid == codec->afg)
+		analog_loop_proc_hook(buffer, codec, 0xfeb);
+}
+#else
+#define stac92hd_proc_hook	NULL
+#define stac92hd7x_proc_hook	NULL
+#define stac9205_proc_hook	NULL
+#define stac927x_proc_hook	NULL
+#endif
+
 #ifdef SND_HDA_NEEDS_RESUME
 static int stac92xx_resume(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 
 	stac92xx_set_config_regs(codec);
-	snd_hda_sequence_write(codec, spec->init);
-	stac_gpio_set(codec, spec->gpio_mask,
-		spec->gpio_dir, spec->gpio_data);
+	stac92xx_init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
-	/* power down inactive DACs */
-	if (spec->dac_list)
-		stac92xx_power_down(codec);
-	/* invoke unsolicited event to reset the HP state */
+	/* fake event to set up pins again to override cached values */
 	if (spec->hp_detect)
-		codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0],
+				       STAC_HP_EVENT);
+	return 0;
+}
+
+static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	if (spec->eapd_mask)
+		stac_gpio_set(codec, spec->gpio_mask,
+				spec->gpio_dir, spec->gpio_data &
+				~spec->eapd_mask);
 	return 0;
 }
 #endif
@@ -4069,6 +4349,7 @@ static struct hda_codec_ops stac92xx_patch_ops = {
 	.free = stac92xx_free,
 	.unsol_event = stac92xx_unsol_event,
 #ifdef SND_HDA_NEEDS_RESUME
+	.suspend = stac92xx_suspend,
 	.resume = stac92xx_resume,
 #endif
 };
@@ -4091,14 +4372,12 @@ static int patch_stac9200(struct hda_codec *codec)
 	if (spec->board_config < 0) {
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else {
-		spec->pin_configs = stac9200_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+					 stac9200_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	spec->multiout.max_channels = 2;
@@ -4154,14 +4433,12 @@ static int patch_stac925x(struct hda_codec *codec)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," 
 				      "using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else if (stac925x_brd_tbl[spec->board_config] != NULL){
-		spec->pin_configs = stac925x_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+					 stac925x_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	spec->multiout.max_channels = 2;
@@ -4225,6 +4502,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 	struct sigmatel_spec *spec;
 	hda_nid_t conn[STAC92HD73_DAC_COUNT + 2];
 	int err = 0;
+	int num_dacs;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4243,43 +4521,37 @@ again:
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD73XX, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else {
-		spec->pin_configs = stac92hd73xx_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+				stac92hd73xx_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
-	spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a,
+	num_dacs = snd_hda_get_connections(codec, 0x0a,
 			conn, STAC92HD73_DAC_COUNT + 2) - 1;
 
-	if (spec->multiout.num_dacs < 0) {
+	if (num_dacs < 3 || num_dacs > 5) {
 		printk(KERN_WARNING "hda_codec: Could not determine "
 		       "number of channels defaulting to DAC count\n");
-		spec->multiout.num_dacs = STAC92HD73_DAC_COUNT;
+		num_dacs = STAC92HD73_DAC_COUNT;
 	}
-
-	switch (spec->multiout.num_dacs) {
+	switch (num_dacs) {
 	case 0x3: /* 6 Channel */
-		spec->multiout.hp_nid = 0x17;
 		spec->mixer = stac92hd73xx_6ch_mixer;
 		spec->init = stac92hd73xx_6ch_core_init;
 		break;
 	case 0x4: /* 8 Channel */
-		spec->multiout.hp_nid = 0x18;
 		spec->mixer = stac92hd73xx_8ch_mixer;
 		spec->init = stac92hd73xx_8ch_core_init;
 		break;
 	case 0x5: /* 10 Channel */
-		spec->multiout.hp_nid = 0x19;
 		spec->mixer = stac92hd73xx_10ch_mixer;
 		spec->init = stac92hd73xx_10ch_core_init;
-	};
+	}
+	spec->multiout.dac_nids = spec->dac_nids;
 
-	spec->multiout.dac_nids = stac92hd73xx_dac_nids;
 	spec->aloopback_mask = 0x01;
 	spec->aloopback_shift = 8;
 
@@ -4310,9 +4582,8 @@ again:
 		spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
 		spec->eapd_switch = 0;
 		spec->num_amps = 1;
-		spec->multiout.hp_nid = 0; /* dual HPs */
 
-		if (!spec->init)
+		if (spec->board_config != STAC_DELL_EQ)
 			spec->init = dell_m6_core_init;
 		switch (spec->board_config) {
 		case STAC_DELL_M6_AMIC: /* Analog Mics */
@@ -4370,6 +4641,8 @@ again:
 
 	codec->patch_ops = stac92xx_patch_ops;
 
+	codec->proc_widget_hook = stac92hd7x_proc_hook;
+
 	return 0;
 }
 
@@ -4401,17 +4674,15 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->pwr_nids = stac92hd83xxx_pwr_nids;
 	spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
-	spec->multiout.dac_nids = stac92hd83xxx_dac_nids;
+	spec->multiout.dac_nids = spec->dac_nids;
 
 	spec->init = stac92hd83xxx_core_init;
 	switch (codec->vendor_id) {
 	case 0x111d7605:
-		spec->multiout.num_dacs = STAC92HD81_DAC_COUNT;
 		break;
 	default:
 		spec->num_pwrs--;
 		spec->init++; /* switch to config #2 */
-		spec->multiout.num_dacs = STAC92HD83_DAC_COUNT;
 	}
 
 	spec->mixer = stac92hd83xxx_mixer;
@@ -4430,14 +4701,12 @@ again:
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD83XXX, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else {
-		spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+				stac92hd83xxx_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	err = stac92xx_parse_auto_config(codec, 0x1d, 0);
@@ -4458,57 +4727,11 @@ again:
 
 	codec->patch_ops = stac92xx_patch_ops;
 
+	codec->proc_widget_hook = stac92hd_proc_hook;
+
 	return 0;
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
-static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	int i;
-	snd_hda_codec_write_cache(codec, codec->afg, 0,
-		AC_VERB_SET_POWER_STATE, pwr);
-
-	msleep(1);
-	for (i = 0; i < spec->num_adcs; i++) {
-		snd_hda_codec_write_cache(codec,
-			spec->adc_nids[i], 0,
-			AC_VERB_SET_POWER_STATE, pwr);
-	}
-};
-
-static int stac92hd71xx_resume(struct hda_codec *codec)
-{
-	stac92hd71xx_set_power_state(codec, AC_PWRST_D0);
-	return stac92xx_resume(codec);
-}
-
-static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state)
-{
-	struct sigmatel_spec *spec = codec->spec;
-
-	stac92hd71xx_set_power_state(codec, AC_PWRST_D3);
-	if (spec->eapd_mask)
-		stac_gpio_set(codec, spec->gpio_mask,
-				spec->gpio_dir, spec->gpio_data &
-				~spec->eapd_mask);
-	return 0;
-};
-
-#endif
-
-static struct hda_codec_ops stac92hd71bxx_patch_ops = {
-	.build_controls = stac92xx_build_controls,
-	.build_pcms = stac92xx_build_pcms,
-	.init = stac92xx_init,
-	.free = stac92xx_free,
-	.unsol_event = stac92xx_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
-	.resume = stac92hd71xx_resume,
-	.suspend = stac92hd71xx_suspend,
-#endif
-};
-
 static struct hda_input_mux stac92hd71bxx_dmux = {
 	.num_items = 4,
 	.items = {
@@ -4544,14 +4767,12 @@ again:
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD71BXX, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else {
-		spec->pin_configs = stac92hd71bxx_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+				stac92hd71bxx_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
@@ -4574,21 +4795,21 @@ again:
 		switch (spec->board_config) {
 		case STAC_HP_M4:
 			/* Enable VREF power saving on GPIO1 detect */
+			err = stac_add_event(spec, codec->afg,
+					     STAC_VREF_EVENT, 0x02);
+			if (err < 0)
+				return err;
 			snd_hda_codec_write_cache(codec, codec->afg, 0,
 				AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02);
 			snd_hda_codec_write_cache(codec, codec->afg, 0,
-					AC_VERB_SET_UNSOLICITED_ENABLE,
-					(AC_USRSP_EN | STAC_VREF_EVENT | 0x01));
+				AC_VERB_SET_UNSOLICITED_ENABLE,
+				AC_USRSP_EN | err);
 			spec->gpio_mask |= 0x02;
 			break;
 		}
 		if ((codec->revision_id & 0xf) == 0 ||
-				(codec->revision_id & 0xf) == 1) {
-#ifdef SND_HDA_NEEDS_RESUME
-			codec->patch_ops = stac92hd71bxx_patch_ops;
-#endif
+		    (codec->revision_id & 0xf) == 1)
 			spec->stream_delay = 40; /* 40 milliseconds */
-		}
 
 		/* no output amps */
 		spec->num_pwrs = 0;
@@ -4597,15 +4818,11 @@ again:
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
-		stac92xx_set_config_reg(codec, 0xf, 0x40f000f0);
+		stac_change_pin_config(codec, 0xf, 0x40f000f0);
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
-		if ((codec->revision_id & 0xf) == 1) {
-#ifdef SND_HDA_NEEDS_RESUME
-			codec->patch_ops = stac92hd71bxx_patch_ops;
-#endif
+		if ((codec->revision_id & 0xf) == 1)
 			spec->stream_delay = 40; /* 40 milliseconds */
-		}
 
 		/* no output amps */
 		spec->num_pwrs = 0;
@@ -4635,7 +4852,7 @@ again:
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
-		stac92xx_set_config_reg(codec, 0x0e, 0x01813040);
+		stac_change_pin_config(codec, 0x0e, 0x01813040);
 		stac92xx_auto_set_pinctl(codec, 0x0e,
 			AC_PINCTL_IN_EN | AC_PINCTL_VREF_80);
 		/* fallthru */
@@ -4656,9 +4873,7 @@ again:
 		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	};
 
-	spec->multiout.num_dacs = 1;
-	spec->multiout.hp_nid = 0x11;
-	spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
+	spec->multiout.dac_nids = spec->dac_nids;
 	if (spec->dinput_mux)
 		spec->private_dimux.num_items +=
 			spec->num_dmics -
@@ -4680,6 +4895,8 @@ again:
 		return err;
 	}
 
+	codec->proc_widget_hook = stac92hd7x_proc_hook;
+
 	return 0;
 };
 
@@ -4741,14 +4958,12 @@ static int patch_stac922x(struct hda_codec *codec)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
 			"using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else if (stac922x_brd_tbl[spec->board_config] != NULL) {
-		spec->pin_configs = stac922x_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+				stac922x_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	spec->adc_nids = stac922x_adc_nids;
@@ -4811,14 +5026,12 @@ static int patch_stac927x(struct hda_codec *codec)
 			snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 				    "STAC927x, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else {
-		spec->pin_configs = stac927x_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+				stac927x_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	spec->digbeep_nid = 0x23;
@@ -4848,15 +5061,15 @@ static int patch_stac927x(struct hda_codec *codec)
 		case 0x10280209:
 		case 0x1028022e:
 			/* correct the device field to SPDIF out */
-			stac92xx_set_config_reg(codec, 0x21, 0x01442070);
+			stac_change_pin_config(codec, 0x21, 0x01442070);
 			break;
 		};
 		/* configure the analog microphone on some laptops */
-		stac92xx_set_config_reg(codec, 0x0c, 0x90a79130);
+		stac_change_pin_config(codec, 0x0c, 0x90a79130);
 		/* correct the front output jack as a hp out */
-		stac92xx_set_config_reg(codec, 0x0f, 0x0227011f);
+		stac_change_pin_config(codec, 0x0f, 0x0227011f);
 		/* correct the front input jack as a mic */
-		stac92xx_set_config_reg(codec, 0x0e, 0x02a79130);
+		stac_change_pin_config(codec, 0x0e, 0x02a79130);
 		/* fallthru */
 	case STAC_DELL_3ST:
 		/* GPIO2 High = Enable EAPD */
@@ -4904,6 +5117,8 @@ static int patch_stac927x(struct hda_codec *codec)
 
 	codec->patch_ops = stac92xx_patch_ops;
 
+	codec->proc_widget_hook = stac927x_proc_hook;
+
 	/*
 	 * !!FIXME!!
 	 * The STAC927x seem to require fairly long delays for certain
@@ -4942,14 +5157,12 @@ static int patch_stac9205(struct hda_codec *codec)
 	if (spec->board_config < 0) {
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return err;
-		}
-		spec->pin_configs = spec->bios_pin_configs;
-	} else {
-		spec->pin_configs = stac9205_brd_tbl[spec->board_config];
-		stac92xx_set_config_regs(codec);
+	} else
+		err = stac_save_pin_cfgs(codec,
+					 stac9205_brd_tbl[spec->board_config]);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
 	}
 
 	spec->digbeep_nid = 0x23;
@@ -4976,15 +5189,18 @@ static int patch_stac9205(struct hda_codec *codec)
 	switch (spec->board_config){
 	case STAC_9205_DELL_M43:
 		/* Enable SPDIF in/out */
-		stac92xx_set_config_reg(codec, 0x1f, 0x01441030);
-		stac92xx_set_config_reg(codec, 0x20, 0x1c410030);
+		stac_change_pin_config(codec, 0x1f, 0x01441030);
+		stac_change_pin_config(codec, 0x20, 0x1c410030);
 
 		/* Enable unsol response for GPIO4/Dock HP connection */
+		err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01);
+		if (err < 0)
+			return err;
 		snd_hda_codec_write_cache(codec, codec->afg, 0,
 			AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10);
 		snd_hda_codec_write_cache(codec, codec->afg, 0,
 					  AC_VERB_SET_UNSOLICITED_ENABLE,
-					  (AC_USRSP_EN | STAC_HP_EVENT));
+					  AC_USRSP_EN | err);
 
 		spec->gpio_dir = 0x0b;
 		spec->eapd_mask = 0x01;
@@ -5022,6 +5238,8 @@ static int patch_stac9205(struct hda_codec *codec)
 
 	codec->patch_ops = stac92xx_patch_ops;
 
+	codec->proc_widget_hook = stac9205_proc_hook;
+
 	return 0;
 }
 
@@ -5078,29 +5296,11 @@ static struct hda_verb vaio_ar_init[] = {
 	{}
 };
 
-/* bind volumes of both NID 0x02 and 0x05 */
-static struct hda_bind_ctls vaio_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-/* bind volumes of both NID 0x02 and 0x05 */
-static struct hda_bind_ctls vaio_bind_master_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
-		0,
-	},
-};
-
 static struct snd_kcontrol_new vaio_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT),
 	/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
@@ -5116,8 +5316,10 @@ static struct snd_kcontrol_new vaio_mixer[] = {
 };
 
 static struct snd_kcontrol_new vaio_ar_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT),
 	/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
@@ -5158,7 +5360,7 @@ static int stac9872_vaio_init(struct hda_codec *codec)
 
 static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res)
 {
-	if (get_hp_pin_presence(codec, 0x0a)) {
+	if (get_pin_presence(codec, 0x0a)) {
 		stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
 		stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
 	} else {
@@ -5269,7 +5471,7 @@ static int patch_stac9872(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_sigmatel[] = {
+static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
  	{ .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
  	{ .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
  	{ .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
@@ -5333,3 +5535,27 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:8384*");
+MODULE_ALIAS("snd-hda-codec-id:111d*");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec");
+
+static struct hda_codec_preset_list sigmatel_list = {
+	.preset = snd_hda_preset_sigmatel,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_sigmatel_init(void)
+{
+	return snd_hda_add_codec_preset(&sigmatel_list);
+}
+
+static void __exit patch_sigmatel_exit(void)
+{
+	snd_hda_delete_codec_preset(&sigmatel_list);
+}
+
+module_init(patch_sigmatel_init)
+module_exit(patch_sigmatel_exit)
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 63e4871e5d8f..c761394cbe84 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -47,15 +47,11 @@
 #include <sound/asoundef.h>
 #include "hda_codec.h"
 #include "hda_local.h"
-#include "hda_patch.h"
 
 /* amp values */
 #define AMP_VAL_IDX_SHIFT	19
 #define AMP_VAL_IDX_MASK	(0x0f<<19)
 
-#define NUM_CONTROL_ALLOC	32
-#define NUM_VERB_ALLOC		32
-
 /* Pin Widget NID */
 #define VT1708_HP_NID		0x13
 #define VT1708_DIGOUT_NID	0x14
@@ -145,8 +141,6 @@ enum {
 	AUTO_SEQ_SIDE
 };
 
-#define get_amp_nid(kc)	((kc)->private_value & 0xffff)
-
 /* Some VT1708S based boards gets the micboost setting wrong, so we have
  * to apply some brute-force and re-write the TLV's by software. */
 static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,
@@ -227,8 +221,7 @@ struct via_spec {
 
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
-	unsigned int num_kctl_alloc, num_kctl_used;
-	struct snd_kcontrol_new *kctl_alloc;
+	struct snd_array kctls;
 	struct hda_input_mux private_imux[2];
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
@@ -272,33 +265,31 @@ static int via_add_control(struct via_spec *spec, int type, const char *name,
 {
 	struct snd_kcontrol_new *knew;
 
-	if (spec->num_kctl_used >= spec->num_kctl_alloc) {
-		int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
-
-		/* array + terminator */
-		knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL);
-		if (!knew)
-			return -ENOMEM;
-		if (spec->kctl_alloc) {
-			memcpy(knew, spec->kctl_alloc,
-			       sizeof(*knew) * spec->num_kctl_alloc);
-			kfree(spec->kctl_alloc);
-		}
-		spec->kctl_alloc = knew;
-		spec->num_kctl_alloc = num;
-	}
-
-	knew = &spec->kctl_alloc[spec->num_kctl_used];
+	snd_array_init(&spec->kctls, sizeof(*knew), 32);
+	knew = snd_array_new(&spec->kctls);
+	if (!knew)
+		return -ENOMEM;
 	*knew = vt1708_control_templates[type];
 	knew->name = kstrdup(name, GFP_KERNEL);
-
 	if (!knew->name)
 		return -ENOMEM;
 	knew->private_value = val;
-	spec->num_kctl_used++;
 	return 0;
 }
 
+static void via_free_kctls(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+
+	if (spec->kctls.list) {
+		struct snd_kcontrol_new *kctl = spec->kctls.list;
+		int i;
+		for (i = 0; i < spec->kctls.used; i++)
+			kfree(kctl[i].name);
+	}
+	snd_array_free(&spec->kctls);
+}
+
 /* create input playback/capture controls for the given pin */
 static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin,
 				const char *ctlname, int idx, int mix_nid)
@@ -896,6 +887,7 @@ static int via_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
+	via_free_kctls(codec); /* no longer needed */
 	return 0;
 }
 
@@ -941,17 +933,11 @@ static int via_build_pcms(struct hda_codec *codec)
 static void via_free(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
-	unsigned int i;
 
 	if (!spec)
 		return;
 
-	if (spec->kctl_alloc) {
-		for (i = 0; i < spec->num_kctl_used; i++)
-			kfree(spec->kctl_alloc[i].name);
-		kfree(spec->kctl_alloc);
-	}
-
+	via_free_kctls(codec);
 	kfree(codec->spec);
 }
 
@@ -1373,8 +1359,8 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708_DIGIN_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs;
 
@@ -1846,8 +1832,8 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1709_DIGIN_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->input_mux = &spec->private_imux[0];
 
@@ -2390,8 +2376,8 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708B_DIGIN_NID;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->input_mux = &spec->private_imux[0];
 
@@ -2855,8 +2841,8 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 
 	spec->extra_dig_out_nid = 0x15;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->input_mux = &spec->private_imux[0];
 
@@ -3174,8 +3160,8 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
 
 	spec->extra_dig_out_nid = 0x1B;
 
-	if (spec->kctl_alloc)
-		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+	if (spec->kctls.list)
+		spec->mixers[spec->num_mixers++] = spec->kctls.list;
 
 	spec->input_mux = &spec->private_imux[0];
 
@@ -3262,74 +3248,97 @@ static int patch_vt1702(struct hda_codec *codec)
 /*
  * patch entries
  */
-struct hda_codec_preset snd_hda_preset_via[] = {
-	{ .id = 0x11061708, .name = "VIA VT1708", .patch = patch_vt1708},
-	{ .id = 0x11061709, .name = "VIA VT1708", .patch = patch_vt1708},
-	{ .id = 0x1106170A, .name = "VIA VT1708", .patch = patch_vt1708},
-	{ .id = 0x1106170B, .name = "VIA VT1708", .patch = patch_vt1708},
-	{ .id = 0x1106E710, .name = "VIA VT1709 10-Ch",
+static struct hda_codec_preset snd_hda_preset_via[] = {
+	{ .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708},
+	{ .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708},
+	{ .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
+	{ .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708},
+	{ .id = 0x1106e710, .name = "VT1709 10-Ch",
 	  .patch = patch_vt1709_10ch},
-	{ .id = 0x1106E711, .name = "VIA VT1709 10-Ch",
+	{ .id = 0x1106e711, .name = "VT1709 10-Ch",
 	  .patch = patch_vt1709_10ch},
-	{ .id = 0x1106E712, .name = "VIA VT1709 10-Ch",
+	{ .id = 0x1106e712, .name = "VT1709 10-Ch",
 	  .patch = patch_vt1709_10ch},
-	{ .id = 0x1106E713, .name = "VIA VT1709 10-Ch",
+	{ .id = 0x1106e713, .name = "VT1709 10-Ch",
 	  .patch = patch_vt1709_10ch},
-	{ .id = 0x1106E714, .name = "VIA VT1709 6-Ch",
+	{ .id = 0x1106e714, .name = "VT1709 6-Ch",
 	  .patch = patch_vt1709_6ch},
-	{ .id = 0x1106E715, .name = "VIA VT1709 6-Ch",
+	{ .id = 0x1106e715, .name = "VT1709 6-Ch",
 	  .patch = patch_vt1709_6ch},
-	{ .id = 0x1106E716, .name = "VIA VT1709 6-Ch",
+	{ .id = 0x1106e716, .name = "VT1709 6-Ch",
 	  .patch = patch_vt1709_6ch},
-	{ .id = 0x1106E717, .name = "VIA VT1709 6-Ch",
+	{ .id = 0x1106e717, .name = "VT1709 6-Ch",
 	  .patch = patch_vt1709_6ch},
-	{ .id = 0x1106E720, .name = "VIA VT1708B 8-Ch",
+	{ .id = 0x1106e720, .name = "VT1708B 8-Ch",
 	  .patch = patch_vt1708B_8ch},
-	{ .id = 0x1106E721, .name = "VIA VT1708B 8-Ch",
+	{ .id = 0x1106e721, .name = "VT1708B 8-Ch",
 	  .patch = patch_vt1708B_8ch},
-	{ .id = 0x1106E722, .name = "VIA VT1708B 8-Ch",
+	{ .id = 0x1106e722, .name = "VT1708B 8-Ch",
 	  .patch = patch_vt1708B_8ch},
-	{ .id = 0x1106E723, .name = "VIA VT1708B 8-Ch",
+	{ .id = 0x1106e723, .name = "VT1708B 8-Ch",
 	  .patch = patch_vt1708B_8ch},
-	{ .id = 0x1106E724, .name = "VIA VT1708B 4-Ch",
+	{ .id = 0x1106e724, .name = "VT1708B 4-Ch",
 	  .patch = patch_vt1708B_4ch},
-	{ .id = 0x1106E725, .name = "VIA VT1708B 4-Ch",
+	{ .id = 0x1106e725, .name = "VT1708B 4-Ch",
 	  .patch = patch_vt1708B_4ch},
-	{ .id = 0x1106E726, .name = "VIA VT1708B 4-Ch",
+	{ .id = 0x1106e726, .name = "VT1708B 4-Ch",
 	  .patch = patch_vt1708B_4ch},
-	{ .id = 0x1106E727, .name = "VIA VT1708B 4-Ch",
+	{ .id = 0x1106e727, .name = "VT1708B 4-Ch",
 	  .patch = patch_vt1708B_4ch},
-	{ .id = 0x11060397, .name = "VIA VT1708S",
+	{ .id = 0x11060397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11061397, .name = "VIA VT1708S",
+	{ .id = 0x11061397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11062397, .name = "VIA VT1708S",
+	{ .id = 0x11062397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11063397, .name = "VIA VT1708S",
+	{ .id = 0x11063397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11064397, .name = "VIA VT1708S",
+	{ .id = 0x11064397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11065397, .name = "VIA VT1708S",
+	{ .id = 0x11065397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11066397, .name = "VIA VT1708S",
+	{ .id = 0x11066397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11067397, .name = "VIA VT1708S",
+	{ .id = 0x11067397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11060398, .name = "VIA VT1702",
+	{ .id = 0x11060398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11061398, .name = "VIA VT1702",
+	{ .id = 0x11061398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11062398, .name = "VIA VT1702",
+	{ .id = 0x11062398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11063398, .name = "VIA VT1702",
+	{ .id = 0x11063398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11064398, .name = "VIA VT1702",
+	{ .id = 0x11064398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11065398, .name = "VIA VT1702",
+	{ .id = 0x11065398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11066398, .name = "VIA VT1702",
+	{ .id = 0x11066398, .name = "VT1702",
 	  .patch = patch_vt1702},
-	{ .id = 0x11067398, .name = "VIA VT1702",
+	{ .id = 0x11067398, .name = "VT1702",
 	  .patch = patch_vt1702},
 	{} /* terminator */
 };
+
+MODULE_ALIAS("snd-hda-codec-id:1106*");
+
+static struct hda_codec_preset_list via_list = {
+	.preset = snd_hda_preset_via,
+	.owner = THIS_MODULE,
+};
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("VIA HD-audio codec");
+
+static int __init patch_via_init(void)
+{
+	return snd_hda_add_codec_preset(&via_list);
+}
+
+static void __exit patch_via_exit(void)
+{
+	snd_hda_delete_codec_preset(&via_list);
+}
+
+module_init(patch_via_init)
+module_exit(patch_via_exit)
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 1b3f11702713..0dfa0540ce2c 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -382,23 +382,25 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 	unsigned char status_mask =
 		VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM;
 	int handled = 0;
-#ifdef CONFIG_SND_DEBUG
 	int timeout = 0;
-#endif
 
 	while (1) {
 		status = inb(ICEREG1724(ice, IRQSTAT));
 		status &= status_mask;
 		if (status == 0)
 			break;
-#ifdef CONFIG_SND_DEBUG
 		if (++timeout > 10) {
-			printk(KERN_ERR
-			       "ice1724: Too long irq loop, status = 0x%x\n",
-			       status);
+			status = inb(ICEREG1724(ice, IRQSTAT));
+			printk(KERN_ERR "ice1724: Too long irq loop, "
+			       "status = 0x%x\n", status);
+			if (status & VT1724_IRQ_MPU_TX) {
+				printk(KERN_ERR "ice1724: Disabling MPU_TX\n");
+				outb(inb(ICEREG1724(ice, IRQMASK)) |
+				     VT1724_IRQ_MPU_TX,
+				     ICEREG1724(ice, IRQMASK));
+			}
 			break;
 		}
-#endif
 		handled = 1;
 		if (status & VT1724_IRQ_MPU_TX) {
 			spin_lock(&ice->reg_lock);
@@ -2351,7 +2353,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
 {
 	struct snd_ice1712 *ice;
 	int err;
-	unsigned char mask;
 	static struct snd_device_ops ops = {
 		.dev_free =	snd_vt1724_dev_free,
 	};
@@ -2412,9 +2413,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
 		return -EIO;
 	}
 
-	/* unmask used interrupts */
-	mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX;
-	outb(mask, ICEREG1724(ice, IRQMASK));
+	/* MPU_RX and TX irq masks are cleared later dynamically */
+	outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK));
+
 	/* don't handle FIFO overrun/underruns (just yet),
 	 * since they cause machine lockups
 	 */
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 9ff3f9e34404..59bbaf8f3e5b 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -1670,7 +1670,7 @@ static irqreturn_t snd_m3_interrupt(int irq, void *dev_id)
 		return IRQ_NONE;
 
 	if (status & HV_INT_PENDING)
-		tasklet_hi_schedule(&chip->hwvol_tq);
+		tasklet_schedule(&chip->hwvol_tq);
 
 	/*
 	 * ack an assp int if its running
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index ae7601f353a7..f23a73577c22 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1010,7 +1010,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card *
 		.dev_free = snd_mixart_chip_dev_free,
 	};
 
-	mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
 	if (! chip) {
 		snd_printk(KERN_ERR "cannot allocate chip\n");
 		return -ENOMEM;
@@ -1025,6 +1025,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card *
 		return err;
 	}
 
+	mgr->chip[idx] = chip;
 	snd_card_set_dev(card, &mgr->pci->dev);
 
 	return 0;
@@ -1377,6 +1378,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
 		sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i);
 
 		if ((err = snd_mixart_create(mgr, card, i)) < 0) {
+			snd_card_free(card);
 			snd_mixart_free(mgr);
 			return err;
 		}
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index b9a06c279397..d3350f383966 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -550,7 +550,7 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id)
 				mgr->msg_fifo[mgr->msg_fifo_writeptr] = msg;
 				mgr->msg_fifo_writeptr++;
 				mgr->msg_fifo_writeptr %= MSG_FIFO_SIZE;
-				tasklet_hi_schedule(&mgr->msg_taskq);
+				tasklet_schedule(&mgr->msg_taskq);
 			}
 			spin_unlock(&mgr->msg_lock);
 			break;
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index b60f6212745a..de999c6d6dd3 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -61,6 +61,7 @@ MODULE_PARM_DESC(enable, "enable card");
 enum {
 	MODEL_CMEDIA_REF,	/* C-Media's reference design */
 	MODEL_MERIDIAN,		/* AuzenTech X-Meridian */
+	MODEL_HALO,		/* HT-Omega Claro halo */
 };
 
 static struct pci_device_id oxygen_ids[] __devinitdata = {
@@ -74,6 +75,7 @@ static struct pci_device_id oxygen_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
 	{ OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN },
 	{ OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF },
+	{ OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, oxygen_ids);
@@ -301,6 +303,8 @@ static int generic_probe(struct oxygen *chip, unsigned long driver_data)
 					    PLAYBACK_1_TO_SPDIF |
 					    CAPTURE_0_FROM_I2S_2 |
 					    CAPTURE_1_FROM_SPDIF;
+	}
+	if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) {
 		chip->model.misc_flags = OXYGEN_MISC_MIDI;
 		chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
 	}
diff --git a/sound/pci/pcxhr/Makefile b/sound/pci/pcxhr/Makefile
index 10473c05918d..b06128e918ca 100644
--- a/sound/pci/pcxhr/Makefile
+++ b/sound/pci/pcxhr/Makefile
@@ -1,2 +1,2 @@
-snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o
+snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o
 obj-$(CONFIG_SND_PCXHR) += snd-pcxhr.o
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 73de6e989b3d..27cf2c28d113 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -40,18 +40,20 @@
 #include "pcxhr_mixer.h"
 #include "pcxhr_hwdep.h"
 #include "pcxhr_core.h"
+#include "pcxhr_mix22.h"
 
 #define DRIVER_NAME "pcxhr"
 
-MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>");
+MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>, "
+	      "Marc Titinger <titinger@digigram.com>");
 MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING);
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}");
 
-static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;		/* Index 0-MAX */
-static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;		/* ID for this card */
-static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
-static int mono[SNDRV_CARDS];					/* capture in mono only */
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */
+static int mono[SNDRV_CARDS];				/* capture  mono only */
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard");
@@ -67,18 +69,58 @@ enum {
 	PCI_ID_PCX882HR,
 	PCI_ID_VX881HR,
 	PCI_ID_PCX881HR,
+	PCI_ID_VX882E,
+	PCI_ID_PCX882E,
+	PCI_ID_VX881E,
+	PCI_ID_PCX881E,
+	PCI_ID_VX1222HR,
 	PCI_ID_PCX1222HR,
+	PCI_ID_VX1221HR,
 	PCI_ID_PCX1221HR,
+	PCI_ID_VX1222E,
+	PCI_ID_PCX1222E,
+	PCI_ID_VX1221E,
+	PCI_ID_PCX1221E,
+	PCI_ID_VX222HR,
+	PCI_ID_VX222E,
+	PCI_ID_PCX22HR,
+	PCI_ID_PCX22E,
+	PCI_ID_VX222HRMIC,
+	PCI_ID_VX222E_MIC,
+	PCI_ID_PCX924HR,
+	PCI_ID_PCX924E,
+	PCI_ID_PCX924HRMIC,
+	PCI_ID_PCX924E_MIC,
 	PCI_ID_LAST
 };
 
 static struct pci_device_id pcxhr_ids[] = {
-	{ 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, },   /* VX882HR */
-	{ 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, },  /* PCX882HR */
-	{ 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, },   /* VX881HR */
-	{ 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, },  /* PCX881HR */
-	{ 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, }, /* PCX1222HR */
-	{ 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, }, /* PCX1221HR */
+	{ 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb021, 0, 0, PCI_ID_VX882E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb121, 0, 0, PCI_ID_PCX882E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb221, 0, 0, PCI_ID_VX881E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb321, 0, 0, PCI_ID_PCX881E, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb401, 0, 0, PCI_ID_VX1222HR, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb601, 0, 0, PCI_ID_VX1221HR, },
+	{ 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb421, 0, 0, PCI_ID_VX1222E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb521, 0, 0, PCI_ID_PCX1222E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb621, 0, 0, PCI_ID_VX1221E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xb721, 0, 0, PCI_ID_PCX1221E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xba01, 0, 0, PCI_ID_VX222HR, },
+	{ 0x10b5, 0x9056, 0x1369, 0xba21, 0, 0, PCI_ID_VX222E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbd01, 0, 0, PCI_ID_PCX22HR, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbd21, 0, 0, PCI_ID_PCX22E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbc01, 0, 0, PCI_ID_VX222HRMIC, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbc21, 0, 0, PCI_ID_VX222E_MIC, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbb01, 0, 0, PCI_ID_PCX924HR, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbb21, 0, 0, PCI_ID_PCX924E, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbf01, 0, 0, PCI_ID_PCX924HRMIC, },
+	{ 0x10b5, 0x9056, 0x1369, 0xbf21, 0, 0, PCI_ID_PCX924E_MIC, },
 	{ 0, }
 };
 
@@ -88,27 +130,55 @@ struct board_parameters {
 	char* board_name;
 	short playback_chips;
 	short capture_chips;
+	short fw_file_set;
 	short firmware_num;
 };
 static struct board_parameters pcxhr_board_params[] = {
-[PCI_ID_VX882HR] =	{ "VX882HR",   4, 4, 41, },
-[PCI_ID_PCX882HR] =	{ "PCX882HR",  4, 4, 41, },
-[PCI_ID_VX881HR] =	{ "VX881HR",   4, 4, 41, },
-[PCI_ID_PCX881HR] =	{ "PCX881HR",  4, 4, 41, },
-[PCI_ID_PCX1222HR] =	{ "PCX1222HR", 6, 1, 42, },
-[PCI_ID_PCX1221HR] =	{ "PCX1221HR", 6, 1, 42, },
+[PCI_ID_VX882HR] =      { "VX882HR",      4, 4, 0, 41 },
+[PCI_ID_PCX882HR] =     { "PCX882HR",     4, 4, 0, 41 },
+[PCI_ID_VX881HR] =      { "VX881HR",      4, 4, 0, 41 },
+[PCI_ID_PCX881HR] =     { "PCX881HR",     4, 4, 0, 41 },
+[PCI_ID_VX882E] =       { "VX882e",       4, 4, 1, 41 },
+[PCI_ID_PCX882E] =      { "PCX882e",      4, 4, 1, 41 },
+[PCI_ID_VX881E] =       { "VX881e",       4, 4, 1, 41 },
+[PCI_ID_PCX881E] =      { "PCX881e",      4, 4, 1, 41 },
+[PCI_ID_VX1222HR] =     { "VX1222HR",     6, 1, 2, 42 },
+[PCI_ID_PCX1222HR] =    { "PCX1222HR",    6, 1, 2, 42 },
+[PCI_ID_VX1221HR] =     { "VX1221HR",     6, 1, 2, 42 },
+[PCI_ID_PCX1221HR] =    { "PCX1221HR",    6, 1, 2, 42 },
+[PCI_ID_VX1222E] =      { "VX1222e",      6, 1, 3, 42 },
+[PCI_ID_PCX1222E] =     { "PCX1222e",     6, 1, 3, 42 },
+[PCI_ID_VX1221E] =      { "VX1221e",      6, 1, 3, 42 },
+[PCI_ID_PCX1221E] =     { "PCX1221e",     6, 1, 3, 42 },
+[PCI_ID_VX222HR] =      { "VX222HR",      1, 1, 4, 44 },
+[PCI_ID_VX222E] =       { "VX222e",       1, 1, 4, 44 },
+[PCI_ID_PCX22HR] =      { "PCX22HR",      1, 0, 4, 44 },
+[PCI_ID_PCX22E] =       { "PCX22e",       1, 0, 4, 44 },
+[PCI_ID_VX222HRMIC] =   { "VX222HR-Mic",  1, 1, 5, 44 },
+[PCI_ID_VX222E_MIC] =   { "VX222e-Mic",   1, 1, 5, 44 },
+[PCI_ID_PCX924HR] =     { "PCX924HR",     1, 1, 5, 44 },
+[PCI_ID_PCX924E] =      { "PCX924e",      1, 1, 5, 44 },
+[PCI_ID_PCX924HRMIC] =  { "PCX924HR-Mic", 1, 1, 5, 44 },
+[PCI_ID_PCX924E_MIC] =  { "PCX924e-Mic",  1, 1, 5, 44 },
 };
 
+/* boards without hw AES1 and SRC onboard are all using fw_file_set==4 */
+/* VX222HR, VX222e, PCX22HR and PCX22e */
+#define PCXHR_BOARD_HAS_AES1(x) (x->fw_file_set != 4)
+/* some boards do not support 192kHz on digital AES input plugs */
+#define PCXHR_BOARD_AESIN_NO_192K(x) ((x->capture_chips == 0) || \
+				      (x->fw_file_set == 0)   || \
+				      (x->fw_file_set == 2))
 
 static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg,
 				   unsigned int* realfreq)
 {
 	unsigned int reg;
 
-	if (freq < 6900 || freq > 110250)
+	if (freq < 6900 || freq > 110000)
 		return -EINVAL;
-	reg = (28224000 * 10) / freq;
-	reg = (reg + 5) / 10;
+	reg = (28224000 * 2) / freq;
+	reg = (reg - 1) / 2;
 	if (reg < 0x200)
 		*pllreg = reg + 0x800;
 	else if (reg < 0x400)
@@ -121,7 +191,7 @@ static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg,
 		reg &= ~3;
 	}
 	if (realfreq)
-		*realfreq = ((28224000 * 10) / reg + 5) / 10;
+		*realfreq = (28224000 / (reg + 1));
 	return 0;
 }
 
@@ -151,11 +221,6 @@ static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg,
 #define PCXHR_FREQ_AES_3		0x03
 #define PCXHR_FREQ_AES_4		0x0d
 
-#define PCXHR_MODIFY_CLOCK_S_BIT	0x04
-
-#define PCXHR_IRQ_TIMER_FREQ		92000
-#define PCXHR_IRQ_TIMER_PERIOD		48
-
 static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate,
 			       unsigned int *reg, unsigned int *freq)
 {
@@ -196,19 +261,32 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate,
 			err = pcxhr_send_msg(mgr, &rmh);
 			if (err < 0) {
 				snd_printk(KERN_ERR
-					   "error CMD_ACCESS_IO_WRITE for PLL register : %x!\n",
-					   err );
+					   "error CMD_ACCESS_IO_WRITE "
+					   "for PLL register : %x!\n", err);
 				return err;
 			}
 		}
 		break;
-	case PCXHR_CLOCK_TYPE_WORD_CLOCK :	val = PCXHR_FREQ_WORD_CLOCK;	break;
-	case PCXHR_CLOCK_TYPE_AES_SYNC :	val = PCXHR_FREQ_SYNC_AES;	break;
-	case PCXHR_CLOCK_TYPE_AES_1 :		val = PCXHR_FREQ_AES_1;		break;
-	case PCXHR_CLOCK_TYPE_AES_2 :		val = PCXHR_FREQ_AES_2;		break;
-	case PCXHR_CLOCK_TYPE_AES_3 :		val = PCXHR_FREQ_AES_3;		break;
-	case PCXHR_CLOCK_TYPE_AES_4 :		val = PCXHR_FREQ_AES_4;		break;
-	default : return -EINVAL;
+	case PCXHR_CLOCK_TYPE_WORD_CLOCK:
+		val = PCXHR_FREQ_WORD_CLOCK;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_SYNC:
+		val = PCXHR_FREQ_SYNC_AES;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_1:
+		val = PCXHR_FREQ_AES_1;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_2:
+		val = PCXHR_FREQ_AES_2;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_3:
+		val = PCXHR_FREQ_AES_3;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_4:
+		val = PCXHR_FREQ_AES_4;
+		break;
+	default:
+		return -EINVAL;
 	}
 	*reg = val;
 	*freq = realfreq;
@@ -216,14 +294,13 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate,
 }
 
 
-int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate)
+static int pcxhr_sub_set_clock(struct pcxhr_mgr *mgr,
+			       unsigned int rate,
+			       int *changed)
 {
 	unsigned int val, realfreq, speed;
 	struct pcxhr_rmh rmh;
-	int err, changed;
-
-	if (rate == 0)
-		return 0; /* nothing to do */
+	int err;
 
 	err = pcxhr_get_clock_reg(mgr, rate, &val, &realfreq);
 	if (err)
@@ -237,13 +314,17 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate)
 	else
 		speed = 2;	/* quad speed */
 	if (mgr->codec_speed != speed) {
-		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);	/* mute outputs */
+		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* mute outputs */
 		rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+		if (DSP_EXT_CMD_SET(mgr)) {
+			rmh.cmd[1]  = 1;
+			rmh.cmd_len = 2;
+		}
 		err = pcxhr_send_msg(mgr, &rmh);
 		if (err)
 			return err;
 
-		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);	/* set speed ratio */
+		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set speed ratio */
 		rmh.cmd[0] |= IO_NUM_SPEED_RATIO;
 		rmh.cmd[1] = speed;
 		rmh.cmd_len = 2;
@@ -253,25 +334,57 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate)
 	}
 	/* set the new frequency */
 	snd_printdd("clock register : set %x\n", val);
-	err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK, val, &changed);
+	err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK,
+					  val, changed);
 	if (err)
 		return err;
+
 	mgr->sample_rate_real = realfreq;
 	mgr->cur_clock_type = mgr->use_clock_type;
 
 	/* unmute after codec speed modes */
 	if (mgr->codec_speed != speed) {
-		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);	/* unmute outputs */
+		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* unmute outputs */
 		rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+		if (DSP_EXT_CMD_SET(mgr)) {
+			rmh.cmd[1]  = 1;
+			rmh.cmd_len = 2;
+		}
 		err = pcxhr_send_msg(mgr, &rmh);
 		if (err)
 			return err;
-		mgr->codec_speed = speed;			/* save new codec speed */
+		mgr->codec_speed = speed;	/* save new codec speed */
 	}
 
+	snd_printdd("pcxhr_sub_set_clock to %dHz (realfreq=%d)\n",
+		    rate, realfreq);
+	return 0;
+}
+
+#define PCXHR_MODIFY_CLOCK_S_BIT	0x04
+
+#define PCXHR_IRQ_TIMER_FREQ		92000
+#define PCXHR_IRQ_TIMER_PERIOD		48
+
+int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate)
+{
+	struct pcxhr_rmh rmh;
+	int err, changed;
+
+	if (rate == 0)
+		return 0; /* nothing to do */
+
+	if (mgr->is_hr_stereo)
+		err = hr222_sub_set_clock(mgr, rate, &changed);
+	else
+		err = pcxhr_sub_set_clock(mgr, rate, &changed);
+
+	if (err)
+		return err;
+
 	if (changed) {
 		pcxhr_init_rmh(&rmh, CMD_MODIFY_CLOCK);
-		rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT;		/* resync fifos  */
+		rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT; /* resync fifos  */
 		if (rate < PCXHR_IRQ_TIMER_FREQ)
 			rmh.cmd[1] = PCXHR_IRQ_TIMER_PERIOD;
 		else
@@ -282,26 +395,39 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate)
 		if (err)
 			return err;
 	}
-	snd_printdd("pcxhr_set_clock to %dHz (realfreq=%d)\n", rate, realfreq);
 	return 0;
 }
 
 
-int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type,
-			     int *sample_rate)
+static int pcxhr_sub_get_external_clock(struct pcxhr_mgr *mgr,
+					enum pcxhr_clock_type clock_type,
+					int *sample_rate)
 {
 	struct pcxhr_rmh rmh;
 	unsigned char reg;
 	int err, rate;
 
 	switch (clock_type) {
-	case PCXHR_CLOCK_TYPE_WORD_CLOCK :	reg = REG_STATUS_WORD_CLOCK;	break;
-	case PCXHR_CLOCK_TYPE_AES_SYNC :	reg = REG_STATUS_AES_SYNC;	break;
-	case PCXHR_CLOCK_TYPE_AES_1 :		reg = REG_STATUS_AES_1;		break;
-	case PCXHR_CLOCK_TYPE_AES_2 :		reg = REG_STATUS_AES_2;		break;
-	case PCXHR_CLOCK_TYPE_AES_3 :		reg = REG_STATUS_AES_3;		break;
-	case PCXHR_CLOCK_TYPE_AES_4 :		reg = REG_STATUS_AES_4;		break;
-	default : return -EINVAL;
+	case PCXHR_CLOCK_TYPE_WORD_CLOCK:
+		reg = REG_STATUS_WORD_CLOCK;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_SYNC:
+		reg = REG_STATUS_AES_SYNC;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_1:
+		reg = REG_STATUS_AES_1;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_2:
+		reg = REG_STATUS_AES_2;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_3:
+		reg = REG_STATUS_AES_3;
+		break;
+	case PCXHR_CLOCK_TYPE_AES_4:
+		reg = REG_STATUS_AES_4;
+		break;
+	default:
+		return -EINVAL;
 	}
 	pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
 	rmh.cmd_len = 2;
@@ -311,7 +437,7 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_
 		err = pcxhr_send_msg(mgr, &rmh);
 		if (err)
 			return err;
-		udelay(100);		/* wait minimum 2 sample_frames at 32kHz ! */
+		udelay(100);	/* wait minimum 2 sample_frames at 32kHz ! */
 		mgr->last_reg_stat = reg;
 	}
 	rmh.cmd[1]  = REG_STATUS_CURRENT;
@@ -336,6 +462,18 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_
 }
 
 
+int pcxhr_get_external_clock(struct pcxhr_mgr *mgr,
+			     enum pcxhr_clock_type clock_type,
+			     int *sample_rate)
+{
+	if (mgr->is_hr_stereo)
+		return hr222_get_external_clock(mgr, clock_type,
+						sample_rate);
+	else
+		return pcxhr_sub_get_external_clock(mgr, clock_type,
+						    sample_rate);
+}
+
 /*
  *  start or stop playback/capture substream
  */
@@ -350,7 +488,8 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream)
 		start = 1;
 	else {
 		if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) {
-			snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state CANNOT be stopped\n");
+			snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state "
+				   "CANNOT be stopped\n");
 			return -EINVAL;
 		}
 		start = 0;
@@ -359,11 +498,12 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream)
 		return -EINVAL;
 
 	stream->timer_abs_periods = 0;
-	stream->timer_period_frag = 0;            /* reset theoretical stream pos */
+	stream->timer_period_frag = 0;	/* reset theoretical stream pos */
 	stream->timer_buf_periods = 0;
 	stream->timer_is_synced = 0;
 
-	stream_mask = stream->pipe->is_capture ? 1 : 1<<stream->substream->number;
+	stream_mask =
+	  stream->pipe->is_capture ? 1 : 1<<stream->substream->number;
 
 	pcxhr_init_rmh(&rmh, start ? CMD_START_STREAM : CMD_STOP_STREAM);
 	pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture,
@@ -373,8 +513,10 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream)
 
 	err = pcxhr_send_msg(chip->mgr, &rmh);
 	if (err)
-		snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n", err);
-	stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED;
+		snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n",
+			   err);
+	stream->status =
+	  start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED;
 	return err;
 }
 
@@ -399,13 +541,15 @@ static int pcxhr_set_format(struct pcxhr_stream *stream)
 		header = HEADER_FMT_BASE_LIN;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
-		header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS | HEADER_FMT_INTEL;
+		header = HEADER_FMT_BASE_LIN |
+			 HEADER_FMT_16BITS | HEADER_FMT_INTEL;
 		break;
 	case SNDRV_PCM_FORMAT_S16_BE:
 		header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS;
 		break;
 	case SNDRV_PCM_FORMAT_S24_3LE:
-		header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS | HEADER_FMT_INTEL;
+		header = HEADER_FMT_BASE_LIN |
+			 HEADER_FMT_24BITS | HEADER_FMT_INTEL;
 		break;
 	case SNDRV_PCM_FORMAT_S24_3BE:
 		header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS;
@@ -414,7 +558,8 @@ static int pcxhr_set_format(struct pcxhr_stream *stream)
 		header = HEADER_FMT_BASE_FLOAT | HEADER_FMT_INTEL;
 		break;
 	default:
-		snd_printk(KERN_ERR "error pcxhr_set_format() : unknown format\n");
+		snd_printk(KERN_ERR
+			   "error pcxhr_set_format() : unknown format\n");
 		return -EINVAL;
 	}
 	chip = snd_pcm_substream_chip(stream->substream);
@@ -432,14 +577,31 @@ static int pcxhr_set_format(struct pcxhr_stream *stream)
 	is_capture = stream->pipe->is_capture;
 	stream_num = is_capture ? 0 : stream->substream->number;
 
-	pcxhr_init_rmh(&rmh, is_capture ? CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT);
-	pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0);
-	if (is_capture)
-		rmh.cmd[0] |= 1<<12;
+	pcxhr_init_rmh(&rmh, is_capture ?
+		       CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT);
+	pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio,
+				  stream_num, 0);
+	if (is_capture) {
+		/* bug with old dsp versions: */
+		/* bit 12 also sets the format of the playback stream */
+		if (DSP_EXT_CMD_SET(chip->mgr))
+			rmh.cmd[0] |= 1<<10;
+		else
+			rmh.cmd[0] |= 1<<12;
+	}
 	rmh.cmd[1] = 0;
-	rmh.cmd[2] = header >> 8;
-	rmh.cmd[3] = (header & 0xff) << 16;
-	rmh.cmd_len = 4;
+	rmh.cmd_len = 2;
+	if (DSP_EXT_CMD_SET(chip->mgr)) {
+		/* add channels and set bit 19 if channels>2 */
+		rmh.cmd[1] = stream->channels;
+		if (!is_capture) {
+			/* playback : add channel mask to command */
+			rmh.cmd[2] = (stream->channels == 1) ? 0x01 : 0x03;
+			rmh.cmd_len = 3;
+		}
+	}
+	rmh.cmd[rmh.cmd_len++] = header >> 8;
+	rmh.cmd[rmh.cmd_len++] = (header & 0xff) << 16;
 	err = pcxhr_send_msg(chip->mgr, &rmh);
 	if (err)
 		snd_printk(KERN_ERR "ERROR pcxhr_set_format err=%x;\n", err);
@@ -456,30 +618,38 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream)
 	is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE);
 	stream_num = is_capture ? 0 : subs->number;
 
-	snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n",
+	snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : "
+		    "addr(%p) bytes(%zx) subs(%d)\n",
 		    is_capture ? 'c' : 'p',
 		    chip->chip_idx, (void *)(long)subs->runtime->dma_addr,
 		    subs->runtime->dma_bytes, subs->number);
 
 	pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS);
-	pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0);
+	pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio,
+				  stream_num, 0);
 
 	/* max buffer size is 2 MByte */
 	snd_BUG_ON(subs->runtime->dma_bytes >= 0x200000);
-	rmh.cmd[1] = subs->runtime->dma_bytes * 8;		/* size in bits */
-	rmh.cmd[2] = subs->runtime->dma_addr >> 24;		/* most significant byte */
-	rmh.cmd[2] |= 1<<19;					/* this is a circular buffer */
-	rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD;	/* least 3 significant bytes */
+	/* size in bits */
+	rmh.cmd[1] = subs->runtime->dma_bytes * 8;
+	/* most significant byte */
+	rmh.cmd[2] = subs->runtime->dma_addr >> 24;
+	/* this is a circular buffer */
+	rmh.cmd[2] |= 1<<19;
+	/* least 3 significant bytes */
+	rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD;
 	rmh.cmd_len = 4;
 	err = pcxhr_send_msg(chip->mgr, &rmh);
 	if (err)
-		snd_printk(KERN_ERR "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err);
+		snd_printk(KERN_ERR
+			   "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err);
 	return err;
 }
 
 
 #if 0
-static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, snd_pcm_uframes_t *sample_count)
+static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream,
+				   snd_pcm_uframes_t *sample_count)
 {
 	struct pcxhr_rmh rmh;
 	int err;
@@ -533,8 +703,8 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 		for (j = 0; j < chip->nb_streams_play; j++) {
 			if (pcxhr_stream_scheduled_get_pipe(&chip->playback_stream[j], &pipe)) {
 				playback_mask |= (1 << pipe->first_audio);
-				break;	/* add only once, as all playback streams of
-					 * one chip use the same pipe
+				break;	/* add only once, as all playback
+					 * streams of one chip use the same pipe
 					 */
 			}
 		}
@@ -545,19 +715,21 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 		return;
 	}
 
-	snd_printdd("pcxhr_trigger_tasklet : playback_mask=%x capture_mask=%x\n",
+	snd_printdd("pcxhr_trigger_tasklet : "
+		    "playback_mask=%x capture_mask=%x\n",
 		    playback_mask, capture_mask);
 
 	/* synchronous stop of all the pipes concerned */
 	err = pcxhr_set_pipe_state(mgr,  playback_mask, capture_mask, 0);
 	if (err) {
 		mutex_unlock(&mgr->setup_mutex);
-		snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error stop pipes (P%x C%x)\n",
+		snd_printk(KERN_ERR "pcxhr_trigger_tasklet : "
+			   "error stop pipes (P%x C%x)\n",
 			   playback_mask, capture_mask);
 		return;
 	}
 
-	/* unfortunately the dsp lost format and buffer info with the stop pipe */
+	/* the dsp lost format and buffer info with the stop pipe */
 	for (i = 0; i < mgr->num_cards; i++) {
 		struct pcxhr_stream *stream;
 		chip = mgr->chip[i];
@@ -596,12 +768,15 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 	err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1);
 	if (err) {
 		mutex_unlock(&mgr->setup_mutex);
-		snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error start pipes (P%x C%x)\n",
+		snd_printk(KERN_ERR "pcxhr_trigger_tasklet : "
+			   "error start pipes (P%x C%x)\n",
 			   playback_mask, capture_mask);
 		return;
 	}
 
-	/* put the streams into the running state now (increment pointer by interrupt) */
+	/* put the streams into the running state now
+	 * (increment pointer by interrupt)
+	 */
 	spin_lock_irqsave(&mgr->lock, flags);
 	for ( i =0; i < mgr->num_cards; i++) {
 		struct pcxhr_stream *stream;
@@ -615,7 +790,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 			stream = &chip->playback_stream[j];
 			if (stream->status == PCXHR_STREAM_STATUS_STARTED) {
 				/* playback will already have advanced ! */
-				stream->timer_period_frag += PCXHR_GRANULARITY;
+				stream->timer_period_frag += mgr->granularity;
 				stream->status = PCXHR_STREAM_STATUS_RUNNING;
 			}
 		}
@@ -653,7 +828,7 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
 					PCXHR_STREAM_STATUS_SCHEDULE_RUN;
 				snd_pcm_trigger_done(s, subs);
 			}
-			tasklet_hi_schedule(&chip->mgr->trigger_taskq);
+			tasklet_schedule(&chip->mgr->trigger_taskq);
 		} else {
 			stream = subs->runtime->private_data;
 			snd_printdd("Only one Substream %c %d\n",
@@ -697,12 +872,14 @@ static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start)
 
 	pcxhr_init_rmh(&rmh, CMD_SET_TIMER_INTERRUPT);
 	if (start) {
-		mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;	/* last dsp time invalid */
-		rmh.cmd[0] |= PCXHR_GRANULARITY;
+		/* last dsp time invalid */
+		mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
+		rmh.cmd[0] |= mgr->granularity;
 	}
 	err = pcxhr_send_msg(mgr, &rmh);
 	if (err < 0)
-		snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n", err);
+		snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n",
+			   err);
 	return err;
 }
 
@@ -713,38 +890,16 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs)
 {
 	struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
 	struct pcxhr_mgr *mgr = chip->mgr;
-	/*
-	struct pcxhr_stream *stream = (pcxhr_stream_t*)subs->runtime->private_data;
-	*/
 	int err = 0;
 
 	snd_printdd("pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n",
 		    subs->runtime->period_size, subs->runtime->periods,
 		    subs->runtime->buffer_size);
 
-	/*
-	if(subs->runtime->period_size <= PCXHR_GRANULARITY) {
-		snd_printk(KERN_ERR "pcxhr_prepare : error period_size too small (%x)\n",
-			   (unsigned int)subs->runtime->period_size);
-		return -EINVAL;
-	}
-	*/
-
 	mutex_lock(&mgr->setup_mutex);
 
 	do {
-		/* if the stream was stopped before, format and buffer were reset */
-		/*
-		if(stream->status == PCXHR_STREAM_STATUS_STOPPED) {
-			err = pcxhr_set_format(stream);
-			if(err) break;
-			err = pcxhr_update_r_buffer(stream);
-			if(err) break;
-		}
-		*/
-
 		/* only the first stream can choose the sample rate */
-		/* the further opened streams will be limited to its frequency (see open) */
 		/* set the clock only once (first stream) */
 		if (mgr->sample_rate != subs->runtime->rate) {
 			err = pcxhr_set_clock(mgr, subs->runtime->rate);
@@ -787,22 +942,9 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs,
 	stream->channels = channels;
 	stream->format = format;
 
-	/* set the format to the board */
-	/*
-	err = pcxhr_set_format(stream);
-	if(err) {
-		mutex_unlock(&mgr->setup_mutex);
-		return err;
-	}
-	*/
 	/* allocate buffer */
 	err = snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw));
 
-	/*
-	if (err > 0) {
-		err = pcxhr_update_r_buffer(stream);
-	}
-	*/
 	mutex_unlock(&mgr->setup_mutex);
 
 	return err;
@@ -820,14 +962,18 @@ static int pcxhr_hw_free(struct snd_pcm_substream *subs)
  */
 static struct snd_pcm_hardware pcxhr_caps =
 {
-	.info             = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
-			      SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
-			      0 /*SNDRV_PCM_INFO_PAUSE*/),
-	.formats	  = ( SNDRV_PCM_FMTBIT_U8 |
-			      SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
-			      SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
-			      SNDRV_PCM_FMTBIT_FLOAT_LE ),
-	.rates            = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,
+	.info             = (SNDRV_PCM_INFO_MMAP |
+			     SNDRV_PCM_INFO_INTERLEAVED |
+			     SNDRV_PCM_INFO_MMAP_VALID |
+			     SNDRV_PCM_INFO_SYNC_START),
+	.formats	  = (SNDRV_PCM_FMTBIT_U8 |
+			     SNDRV_PCM_FMTBIT_S16_LE |
+			     SNDRV_PCM_FMTBIT_S16_BE |
+			     SNDRV_PCM_FMTBIT_S24_3LE |
+			     SNDRV_PCM_FMTBIT_S24_3BE |
+			     SNDRV_PCM_FMTBIT_FLOAT_LE),
+	.rates            = (SNDRV_PCM_RATE_CONTINUOUS |
+			     SNDRV_PCM_RATE_8000_192000),
 	.rate_min         = 8000,
 	.rate_max         = 192000,
 	.channels_min     = 1,
@@ -847,6 +993,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
 	struct pcxhr_mgr       *mgr = chip->mgr;
 	struct snd_pcm_runtime *runtime = subs->runtime;
 	struct pcxhr_stream    *stream;
+	int err;
 
 	mutex_lock(&mgr->setup_mutex);
 
@@ -874,6 +1021,18 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
 		return -EBUSY;
 	}
 
+	/* float format support is in some cases buggy on stereo cards */
+	if (mgr->is_hr_stereo)
+		runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_FLOAT_LE;
+
+	/* buffer-size should better be multiple of period-size */
+	err = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0) {
+		mutex_unlock(&mgr->setup_mutex);
+		return err;
+	}
+
 	/* if a sample rate is already used or fixed by external clock,
 	 * the stream cannot change
 	 */
@@ -889,7 +1048,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
 				mutex_unlock(&mgr->setup_mutex);
 				return -EBUSY;
 			}
-			runtime->hw.rate_min = runtime->hw.rate_max = external_rate;
+			runtime->hw.rate_min = external_rate;
+			runtime->hw.rate_max = external_rate;
 		}
 	}
 
@@ -899,9 +1059,11 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
 
 	runtime->private_data = stream;
 
-	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4);
-	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
-
+	/* better get a divisor of granularity values (96 or 192) */
+	snd_pcm_hw_constraint_step(runtime, 0,
+				   SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32);
+	snd_pcm_hw_constraint_step(runtime, 0,
+				   SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32);
 	snd_pcm_set_sync(subs);
 
 	mgr->ref_count_rate++;
@@ -919,11 +1081,12 @@ static int pcxhr_close(struct snd_pcm_substream *subs)
 
 	mutex_lock(&mgr->setup_mutex);
 
-	snd_printdd("pcxhr_close chip%d subs%d\n", chip->chip_idx, subs->number);
+	snd_printdd("pcxhr_close chip%d subs%d\n",
+		    chip->chip_idx, subs->number);
 
 	/* sample rate released */
 	if (--mgr->ref_count_rate == 0) {
-		mgr->sample_rate = 0;		/* the sample rate is no more locked */
+		mgr->sample_rate = 0;	/* the sample rate is no more locked */
 		pcxhr_hardware_timer(mgr, 0);	/* stop the DSP-timer */
 	}
 
@@ -1016,7 +1179,8 @@ static int pcxhr_chip_dev_free(struct snd_device *device)
 
 /*
  */
-static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, int idx)
+static int __devinit pcxhr_create(struct pcxhr_mgr *mgr,
+				  struct snd_card *card, int idx)
 {
 	int err;
 	struct snd_pcxhr *chip;
@@ -1024,7 +1188,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card,
 		.dev_free = pcxhr_chip_dev_free,
 	};
 
-	mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
 	if (! chip) {
 		snd_printk(KERN_ERR "cannot allocate chip\n");
 		return -ENOMEM;
@@ -1040,7 +1204,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card,
 
 	if (idx < mgr->capture_chips) {
 		if (mgr->mono_capture)
-			chip->nb_streams_capt = 2;	/* 2 mono streams (left+right) */
+			chip->nb_streams_capt = 2;	/* 2 mono streams */
 		else
 			chip->nb_streams_capt = 1;	/* or 1 stereo stream */
 	}
@@ -1050,13 +1214,15 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card,
 		return err;
 	}
 
+	mgr->chip[idx] = chip;
 	snd_card_set_dev(card, &mgr->pci->dev);
 
 	return 0;
 }
 
 /* proc interface */
-static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+static void pcxhr_proc_info(struct snd_info_entry *entry,
+			    struct snd_info_buffer *buffer)
 {
 	struct snd_pcxhr *chip = entry->private_data;
 	struct pcxhr_mgr *mgr = chip->mgr;
@@ -1069,8 +1235,10 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer
 		short ver_maj = (mgr->dsp_version >> 16) & 0xff;
 		short ver_min = (mgr->dsp_version >> 8) & 0xff;
 		short ver_build = mgr->dsp_version & 0xff;
-		snd_iprintf(buffer, "module version %s\n", PCXHR_DRIVER_VERSION_STRING);
-		snd_iprintf(buffer, "dsp version %d.%d.%d\n", ver_maj, ver_min, ver_build);
+		snd_iprintf(buffer, "module version %s\n",
+			    PCXHR_DRIVER_VERSION_STRING);
+		snd_iprintf(buffer, "dsp version %d.%d.%d\n",
+			    ver_maj, ver_min, ver_build);
 		if (mgr->board_has_analog)
 			snd_iprintf(buffer, "analog io available\n");
 		else
@@ -1084,18 +1252,22 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer
 			if (ref > 0) {
 				if (mgr->sample_rate_real != 0 &&
 				    mgr->sample_rate_real != 48000) {
-					ref = (ref * 48000) / mgr->sample_rate_real;
-					if (mgr->sample_rate_real >= PCXHR_IRQ_TIMER_FREQ)
+					ref = (ref * 48000) /
+					  mgr->sample_rate_real;
+					if (mgr->sample_rate_real >=
+					    PCXHR_IRQ_TIMER_FREQ)
 						ref *= 2;
 				}
 				cur = 100 - (100 * cur) / ref;
 				snd_iprintf(buffer, "cpu load    %d%%\n", cur);
-				snd_iprintf(buffer, "buffer pool %d/%d kWords\n",
+				snd_iprintf(buffer, "buffer pool %d/%d\n",
 					    rmh.stat[2], rmh.stat[3]);
 			}
 		}
-		snd_iprintf(buffer, "dma granularity : %d\n", PCXHR_GRANULARITY);
-		snd_iprintf(buffer, "dsp time errors : %d\n", mgr->dsp_time_err);
+		snd_iprintf(buffer, "dma granularity : %d\n",
+			    mgr->granularity);
+		snd_iprintf(buffer, "dsp time errors : %d\n",
+			    mgr->dsp_time_err);
 		snd_iprintf(buffer, "dsp async pipe xrun errors : %d\n",
 			    mgr->async_err_pipe_xrun);
 		snd_iprintf(buffer, "dsp async stream xrun errors : %d\n",
@@ -1110,33 +1282,52 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer
 		rmh.cmd_idx = CMD_LAST_INDEX;
 		if( ! pcxhr_send_msg(mgr, &rmh) ) {
 			int i;
+			if (rmh.stat_len > 8)
+				rmh.stat_len = 8;
 			for (i = 0; i < rmh.stat_len; i++)
-				snd_iprintf(buffer, "debug[%02d] = %06x\n", i,  rmh.stat[i]);
+				snd_iprintf(buffer, "debug[%02d] = %06x\n",
+					    i,  rmh.stat[i]);
 		}
 	} else
 		snd_iprintf(buffer, "no firmware loaded\n");
 	snd_iprintf(buffer, "\n");
 }
-static void pcxhr_proc_sync(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+static void pcxhr_proc_sync(struct snd_info_entry *entry,
+			    struct snd_info_buffer *buffer)
 {
 	struct snd_pcxhr *chip = entry->private_data;
 	struct pcxhr_mgr *mgr = chip->mgr;
-	static char *texts[7] = {
-		"Internal", "Word", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4"
+	static const char *textsHR22[3] = {
+		"Internal", "AES Sync", "AES 1"
 	};
+	static const char *textsPCXHR[7] = {
+		"Internal", "Word", "AES Sync",
+		"AES 1", "AES 2", "AES 3", "AES 4"
+	};
+	const char **texts;
+	int max_clock;
+	if (mgr->is_hr_stereo) {
+		texts = textsHR22;
+		max_clock = HR22_CLOCK_TYPE_MAX;
+	} else {
+		texts = textsPCXHR;
+		max_clock = PCXHR_CLOCK_TYPE_MAX;
+	}
 
 	snd_iprintf(buffer, "\n%s\n", mgr->longname);
-	snd_iprintf(buffer, "Current Sample Clock\t: %s\n", texts[mgr->cur_clock_type]);
-	snd_iprintf(buffer, "Current Sample Rate\t= %d\n", mgr->sample_rate_real);
-
+	snd_iprintf(buffer, "Current Sample Clock\t: %s\n",
+		    texts[mgr->cur_clock_type]);
+	snd_iprintf(buffer, "Current Sample Rate\t= %d\n",
+		    mgr->sample_rate_real);
 	/* commands available when embedded DSP is running */
 	if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
 		int i, err, sample_rate;
-		for (i = PCXHR_CLOCK_TYPE_WORD_CLOCK; i< (3 + mgr->capture_chips); i++) {
+		for (i = 1; i <= max_clock; i++) {
 			err = pcxhr_get_external_clock(mgr, i, &sample_rate);
 			if (err)
 				break;
-			snd_iprintf(buffer, "%s Clock\t\t= %d\n", texts[i], sample_rate);
+			snd_iprintf(buffer, "%s Clock\t\t= %d\n",
+				    texts[i], sample_rate);
 		}
 	} else
 		snd_iprintf(buffer, "no firmware loaded\n");
@@ -1194,7 +1385,8 @@ static int pcxhr_free(struct pcxhr_mgr *mgr)
 /*
  *    probe function - creates the card manager
  */
-static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+static int __devinit pcxhr_probe(struct pci_dev *pci,
+				 const struct pci_device_id *pci_id)
 {
 	static int dev;
 	struct pcxhr_mgr *mgr;
@@ -1217,7 +1409,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 
 	/* check if we can restrict PCI DMA transfers to 32 bits */
 	if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) {
-		snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n");
+		snd_printk(KERN_ERR "architecture does not support "
+			   "32bit PCI busmaster DMA\n");
 		pci_disable_device(pci);
 		return -ENXIO;
 	}
@@ -1234,11 +1427,25 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 		pci_disable_device(pci);
 		return -ENODEV;
 	}
-	card_name = pcxhr_board_params[pci_id->driver_data].board_name;
-	mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips;
-	mgr->capture_chips  = pcxhr_board_params[pci_id->driver_data].capture_chips;
-	mgr->firmware_num  = pcxhr_board_params[pci_id->driver_data].firmware_num;
+	card_name =
+		pcxhr_board_params[pci_id->driver_data].board_name;
+	mgr->playback_chips =
+		pcxhr_board_params[pci_id->driver_data].playback_chips;
+	mgr->capture_chips  =
+		pcxhr_board_params[pci_id->driver_data].capture_chips;
+	mgr->fw_file_set =
+		pcxhr_board_params[pci_id->driver_data].fw_file_set;
+	mgr->firmware_num  =
+		pcxhr_board_params[pci_id->driver_data].firmware_num;
 	mgr->mono_capture = mono[dev];
+	mgr->is_hr_stereo = (mgr->playback_chips == 1);
+	mgr->board_has_aes1 = PCXHR_BOARD_HAS_AES1(mgr);
+	mgr->board_aes_in_192k = !PCXHR_BOARD_AESIN_NO_192K(mgr);
+
+	if (mgr->is_hr_stereo)
+		mgr->granularity = PCXHR_GRANULARITY_HR22;
+	else
+		mgr->granularity = PCXHR_GRANULARITY;
 
 	/* resource assignment */
 	if ((err = pci_request_regions(pci, card_name)) < 0) {
@@ -1261,7 +1468,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 	mgr->irq = pci->irq;
 
 	sprintf(mgr->shortname, "Digigram %s", card_name);
-	sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i", mgr->shortname,
+	sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i",
+		mgr->shortname,
 		mgr->port[0], mgr->port[1], mgr->port[2], mgr->irq);
 
 	/* ISR spinlock  */
@@ -1272,10 +1480,14 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 	mutex_init(&mgr->setup_mutex);
 
 	/* init taslket */
-	tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, (unsigned long) mgr);
-	tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, (unsigned long) mgr);
+	tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet,
+		     (unsigned long) mgr);
+	tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet,
+		     (unsigned long) mgr);
+
 	mgr->prmh = kmalloc(sizeof(*mgr->prmh) + 
-			    sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS),
+			    sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS -
+					   PCXHR_SIZE_MAX_STATUS),
 			    GFP_KERNEL);
 	if (! mgr->prmh) {
 		pcxhr_free(mgr);
@@ -1296,7 +1508,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 		else
 			idx = index[dev] + i;
 
-		snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : card_name, i);
+		snprintf(tmpid, sizeof(tmpid), "%s-%d",
+			 id[dev] ? id[dev] : card_name, i);
 		card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
 
 		if (! card) {
@@ -1310,6 +1523,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 		sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i);
 
 		if ((err = pcxhr_create(mgr, card, i)) < 0) {
+			snd_card_free(card);
 			pcxhr_free(mgr);
 			return err;
 		}
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 652064787a55..84131a916c92 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -27,15 +27,18 @@
 #include <linux/mutex.h>
 #include <sound/pcm.h>
 
-#define PCXHR_DRIVER_VERSION		0x000804	/* 0.8.4 */
-#define PCXHR_DRIVER_VERSION_STRING	"0.8.4"		/* 0.8.4 */
+#define PCXHR_DRIVER_VERSION		0x000905	/* 0.9.5 */
+#define PCXHR_DRIVER_VERSION_STRING	"0.9.5"		/* 0.9.5 */
 
 
-#define PCXHR_MAX_CARDS			6
-#define PCXHR_PLAYBACK_STREAMS		4
+#define PCXHR_MAX_CARDS		6
+#define PCXHR_PLAYBACK_STREAMS	4
 
-#define PCXHR_GRANULARITY		96	/* transfer granularity (should be min 96 and multiple of 48) */
-#define PCXHR_GRANULARITY_MIN		96	/* transfer granularity of pipes and the dsp time (MBOX4) */
+#define PCXHR_GRANULARITY	96	/* min 96 and multiple of 48 */
+/* transfer granularity of pipes and the dsp time (MBOX4) */
+#define PCXHR_GRANULARITY_MIN	96
+/* TODO : granularity could be 64 or 128 */
+#define PCXHR_GRANULARITY_HR22	192	/* granularity for stereo cards */
 
 struct snd_pcxhr;
 struct pcxhr_mgr;
@@ -51,6 +54,11 @@ enum pcxhr_clock_type {
 	PCXHR_CLOCK_TYPE_AES_2,
 	PCXHR_CLOCK_TYPE_AES_3,
 	PCXHR_CLOCK_TYPE_AES_4,
+	PCXHR_CLOCK_TYPE_MAX = PCXHR_CLOCK_TYPE_AES_4,
+	HR22_CLOCK_TYPE_INTERNAL = PCXHR_CLOCK_TYPE_INTERNAL,
+	HR22_CLOCK_TYPE_AES_SYNC,
+	HR22_CLOCK_TYPE_AES_1,
+	HR22_CLOCK_TYPE_MAX = HR22_CLOCK_TYPE_AES_1,
 };
 
 struct pcxhr_mgr {
@@ -61,6 +69,8 @@ struct pcxhr_mgr {
 
 	int irq;
 
+	int granularity;
+
 	/* card access with 1 mem bar and 2 io bar's */
 	unsigned long port[3];
 
@@ -83,11 +93,16 @@ struct pcxhr_mgr {
 	/* hardware interface */
 	unsigned int dsp_loaded;	/* bit flags of loaded dsp indices */
 	unsigned int dsp_version;	/* read from embedded once firmware is loaded */
-	int board_has_analog;		/* if 0 the board is digital only */
-	int mono_capture;		/* if 1 the board does mono capture */
-	int playback_chips;		/* 4 or 6 */
-	int capture_chips;		/* 4 or 1 */
-	int firmware_num;		/* 41 or 42 */
+	int playback_chips;
+	int capture_chips;
+	int fw_file_set;
+	int firmware_num;
+	int is_hr_stereo:1;
+	int board_has_aes1:1;	/* if 1 board has AES1 plug and SRC */
+	int board_has_analog:1;	/* if 0 the board is digital only */
+	int board_has_mic:1;	/* if 1 the board has microphone input */
+	int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
+	int mono_capture:1;	/* if 1 the board does mono capture */
 
 	struct snd_dma_buffer hostport;
 
@@ -106,6 +121,9 @@ struct pcxhr_mgr {
 	int async_err_stream_xrun;
 	int async_err_pipe_xrun;
 	int async_err_other_last;
+
+	unsigned char xlx_cfg;		/* copy of PCXHR_XLX_CFG register */
+	unsigned char xlx_selmic;	/* copy of PCXHR_XLX_SELMIC register */
 };
 
 
@@ -155,24 +173,30 @@ struct snd_pcxhr {
 
 	struct snd_pcm *pcm;		/* PCM */
 
-	struct pcxhr_pipe playback_pipe;		/* 1 stereo pipe only */
-	struct pcxhr_pipe capture_pipe[2];		/* 1 stereo pipe or 2 mono pipes */
+	struct pcxhr_pipe playback_pipe;	/* 1 stereo pipe only */
+	struct pcxhr_pipe capture_pipe[2];	/* 1 stereo or 2 mono pipes */
 
 	struct pcxhr_stream playback_stream[PCXHR_PLAYBACK_STREAMS];
-	struct pcxhr_stream capture_stream[2];	/* 1 stereo stream or 2 mono streams */
+	struct pcxhr_stream capture_stream[2];	/* 1 stereo or 2 mono streams */
 	int nb_streams_play;
 	int nb_streams_capt;
 
-	int analog_playback_active[2];		/* Mixer : Master Playback active (!mute) */
-	int analog_playback_volume[2];		/* Mixer : Master Playback Volume */
-	int analog_capture_volume[2];		/* Mixer : Master Capture Volume */
-	int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2];	/* Mixer : Digital Playback Active [streams][stereo]*/
-	int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2];	/* Mixer : Digital Playback Volume [streams][stereo]*/
-	int digital_capture_volume[2];		/* Mixer : Digital Capture Volume [stereo] */
-	int monitoring_active[2];		/* Mixer : Monitoring Active */
-	int monitoring_volume[2];		/* Mixer : Monitoring Volume */
-	int audio_capture_source;		/* Mixer : Audio Capture Source */
-	unsigned char aes_bits[5];		/* Mixer : IEC958_AES bits */
+	int analog_playback_active[2];	/* Mixer : Master Playback !mute */
+	int analog_playback_volume[2];	/* Mixer : Master Playback Volume */
+	int analog_capture_volume[2];	/* Mixer : Master Capture Volume */
+	int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2];
+	int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2];
+	int digital_capture_volume[2];	/* Mixer : Digital Capture Volume */
+	int monitoring_active[2];	/* Mixer : Monitoring Active */
+	int monitoring_volume[2];	/* Mixer : Monitoring Volume */
+	int audio_capture_source;	/* Mixer : Audio Capture Source */
+	int mic_volume;			/* used by cards with MIC only */
+	int mic_boost;			/* used by cards with MIC only */
+	int mic_active;			/* used by cards with MIC only */
+	int analog_capture_active;	/* used by cards with MIC only */
+	int phantom_power;		/* used by cards with MIC only */
+
+	unsigned char aes_bits[5];	/* Mixer : IEC958_AES bits */
 };
 
 struct pcxhr_hostport
@@ -184,6 +208,8 @@ struct pcxhr_hostport
 /* exported */
 int pcxhr_create_pcm(struct snd_pcxhr *chip);
 int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate);
-int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, int *sample_rate);
+int pcxhr_get_external_clock(struct pcxhr_mgr *mgr,
+			     enum pcxhr_clock_type clock_type,
+			     int *sample_rate);
 
 #endif /* __SOUND_PCXHR_H */
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 7143259cfe34..833e7180ad2d 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -132,13 +132,15 @@ static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg,
 		*read = PCXHR_INPB(mgr, reg);
 		if ((*read & mask) == bit) {
 			if (i > 100)
-				snd_printdd("ATTENTION! check_reg(%x) loopcount=%d\n",
+				snd_printdd("ATTENTION! check_reg(%x) "
+					    "loopcount=%d\n",
 					    reg, i);
 			return 0;
 		}
 		i++;
 	} while (time_after_eq(end_time, jiffies));
-	snd_printk(KERN_ERR "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=0x%x\n",
+	snd_printk(KERN_ERR
+		   "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=%x\n",
 		   reg, mask, *read);
 	return -EIO;
 }
@@ -159,18 +161,22 @@ static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg,
 #define PCXHR_IT_TEST_XILINX		(0x0000003C | PCXHR_MASK_IT_HF1 | \
 					 PCXHR_MASK_IT_MANAGE_HF5)
 #define PCXHR_IT_DOWNLOAD_BOOT		(0x0000000C | PCXHR_MASK_IT_HF1 | \
-					 PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT)
+					 PCXHR_MASK_IT_MANAGE_HF5 | \
+					 PCXHR_MASK_IT_WAIT)
 #define PCXHR_IT_RESET_BOARD_FUNC	(0x0000000C | PCXHR_MASK_IT_HF0 | \
-					 PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT_EXTRA)
+					 PCXHR_MASK_IT_MANAGE_HF5 | \
+					 PCXHR_MASK_IT_WAIT_EXTRA)
 #define PCXHR_IT_DOWNLOAD_DSP		(0x0000000C | \
-					 PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT)
+					 PCXHR_MASK_IT_MANAGE_HF5 | \
+					 PCXHR_MASK_IT_WAIT)
 #define PCXHR_IT_DEBUG			(0x0000005A | PCXHR_MASK_IT_NO_HF0_HF1)
 #define PCXHR_IT_RESET_SEMAPHORE	(0x0000005C | PCXHR_MASK_IT_NO_HF0_HF1)
 #define PCXHR_IT_MESSAGE		(0x00000074 | PCXHR_MASK_IT_NO_HF0_HF1)
 #define PCXHR_IT_RESET_CHK		(0x00000076 | PCXHR_MASK_IT_NO_HF0_HF1)
 #define PCXHR_IT_UPDATE_RBUFFER		(0x00000078 | PCXHR_MASK_IT_NO_HF0_HF1)
 
-static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atomic)
+static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr,
+			     unsigned int itdsp, int atomic)
 {
 	int err;
 	unsigned char reg;
@@ -178,17 +184,21 @@ static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atom
 	if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) {
 		/* clear hf5 bit */
 		PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0,
-			    PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & ~PCXHR_MBOX0_HF5);
+			    PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) &
+			    ~PCXHR_MBOX0_HF5);
 	}
 	if ((itdsp & PCXHR_MASK_IT_NO_HF0_HF1) == 0) {
-		reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ;
+		reg = (PCXHR_ICR_HI08_RREQ |
+		       PCXHR_ICR_HI08_TREQ |
+		       PCXHR_ICR_HI08_HDRQ);
 		if (itdsp & PCXHR_MASK_IT_HF0)
 			reg |= PCXHR_ICR_HI08_HF0;
 		if (itdsp & PCXHR_MASK_IT_HF1)
 			reg |= PCXHR_ICR_HI08_HF1;
 		PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg);
 	}
-	reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) | PCXHR_CVR_HI08_HC);
+	reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) |
+			      PCXHR_CVR_HI08_HC);
 	PCXHR_OUTPB(mgr, PCXHR_DSP_CVR, reg);
 	if (itdsp & PCXHR_MASK_IT_WAIT) {
 		if (atomic)
@@ -211,10 +221,14 @@ static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atom
 	}
 	if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) {
 		/* wait for hf5 bit */
-		err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, PCXHR_MBOX0_HF5,
-					  PCXHR_MBOX0_HF5, PCXHR_TIMEOUT_DSP, &reg);
+		err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0,
+					  PCXHR_MBOX0_HF5,
+					  PCXHR_MBOX0_HF5,
+					  PCXHR_TIMEOUT_DSP,
+					  &reg);
 		if (err) {
-			snd_printk(KERN_ERR "pcxhr_send_it_dsp : TIMEOUT HF5\n");
+			snd_printk(KERN_ERR
+				   "pcxhr_send_it_dsp : TIMEOUT HF5\n");
 			return err;
 		}
 	}
@@ -263,7 +277,8 @@ void pcxhr_enable_dsp(struct pcxhr_mgr *mgr)
 /*
  * load the xilinx image
  */
-int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilinx, int second)
+int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr,
+			     const struct firmware *xilinx, int second)
 {
 	unsigned int i;
 	unsigned int chipsc;
@@ -274,7 +289,9 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin
 	/* test first xilinx */
 	chipsc = PCXHR_INPL(mgr, PCXHR_PLX_CHIPSC);
 	/* REV01 cards do not support the PCXHR_CHIPSC_GPI_USERI bit anymore */
-	/* this bit will always be 1; no possibility to test presence of first xilinx */
+	/* this bit will always be 1;
+	 * no possibility to test presence of first xilinx
+	 */
 	if(second) {
 		if ((chipsc & PCXHR_CHIPSC_GPI_USERI) == 0) {
 			snd_printk(KERN_ERR "error loading first xilinx\n");
@@ -290,7 +307,8 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin
 		data = *image;
 		mask = 0x80;
 		while (mask) {
-			chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | PCXHR_CHIPSC_DATA_IN);
+			chipsc &= ~(PCXHR_CHIPSC_DATA_CLK |
+				    PCXHR_CHIPSC_DATA_IN);
 			if (data & mask)
 				chipsc |= PCXHR_CHIPSC_DATA_IN;
 			PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc);
@@ -330,15 +348,20 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp)
 		data = dsp->data + i;
 		if (i == 0) {
 			/* test data header consistency */
-			len = (unsigned int)((data[0]<<16) + (data[1]<<8) + data[2]);
-			if (len && dsp->size != (len + 2) * 3)
+			len = (unsigned int)((data[0]<<16) +
+					     (data[1]<<8) +
+					     data[2]);
+			if (len && (dsp->size != (len + 2) * 3))
 				return -EINVAL;
 		}
 		/* wait DSP ready for new transfer */
-		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY,
-					  PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &dummy);
+		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR,
+					  PCXHR_ISR_HI08_TRDY,
+					  PCXHR_ISR_HI08_TRDY,
+					  PCXHR_TIMEOUT_DSP, &dummy);
 		if (err) {
-			snd_printk(KERN_ERR "dsp loading error at position %d\n", i);
+			snd_printk(KERN_ERR
+				   "dsp loading error at position %d\n", i);
 			return err;
 		}
 		/* send host data */
@@ -357,7 +380,8 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp)
 /*
  * load the eeprom image
  */
-int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eeprom)
+int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr,
+			     const struct firmware *eeprom)
 {
 	int err;
 	unsigned char reg;
@@ -365,7 +389,9 @@ int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eepro
 	/* init value of the ICR register */
 	reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ;
 	if (PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & PCXHR_MBOX0_BOOT_HERE) {
-		/* no need to load the eeprom binary, but init the HI08 interface */
+		/* no need to load the eeprom binary,
+		 * but init the HI08 interface
+		 */
 		PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg | PCXHR_ICR_HI08_INIT);
 		msleep(PCXHR_WAIT_DEFAULT);
 		PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg);
@@ -429,8 +455,10 @@ int pcxhr_load_dsp_binary(struct pcxhr_mgr *mgr, const struct firmware *dsp)
 	if (err)
 		return err;
 	/* wait for chk bit */
-	return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK,
-				   PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &dummy);
+	return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR,
+				   PCXHR_ISR_HI08_CHK,
+				   PCXHR_ISR_HI08_CHK,
+				   PCXHR_TIMEOUT_DSP, &dummy);
 }
 
 
@@ -443,8 +471,8 @@ struct pcxhr_cmd_info {
 /* RMH status type */
 enum {
 	RMH_SSIZE_FIXED = 0,	/* status size fix (st_length = 0..x) */
-	RMH_SSIZE_ARG = 1,	/* status size given in the LSB byte (used with st_length = 1) */
-	RMH_SSIZE_MASK = 2,	/* status size given in bitmask  (used with st_length = 1) */
+	RMH_SSIZE_ARG = 1,	/* status size given in the LSB byte */
+	RMH_SSIZE_MASK = 2,	/* status size given in bitmask */
 };
 
 /*
@@ -474,7 +502,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = {
 [CMD_UPDATE_R_BUFFERS] =		{ 0x840000, 0, RMH_SSIZE_FIXED },
 [CMD_FORMAT_STREAM_OUT] =		{ 0x860000, 0, RMH_SSIZE_FIXED },
 [CMD_FORMAT_STREAM_IN] =		{ 0x870000, 0, RMH_SSIZE_FIXED },
-[CMD_STREAM_SAMPLE_COUNT] =		{ 0x902000, 2, RMH_SSIZE_FIXED },	/* stat_len = nb_streams * 2 */
+[CMD_STREAM_SAMPLE_COUNT] =		{ 0x902000, 2, RMH_SSIZE_FIXED },
 [CMD_AUDIO_LEVEL_ADJUST] =		{ 0xc22000, 0, RMH_SSIZE_FIXED },
 };
 
@@ -524,10 +552,13 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 
 	for (i = 0; i < rmh->stat_len; i++) {
 		/* wait for receiver full */
-		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF,
-					  PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, &reg);
+		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR,
+					  PCXHR_ISR_HI08_RXDF,
+					  PCXHR_ISR_HI08_RXDF,
+					  PCXHR_TIMEOUT_DSP, &reg);
 		if (err) {
-			snd_printk(KERN_ERR "ERROR RMH stat: ISR:RXDF=1 (ISR = %x; i=%d )\n",
+			snd_printk(KERN_ERR "ERROR RMH stat: "
+				   "ISR:RXDF=1 (ISR = %x; i=%d )\n",
 				   reg, i);
 			return err;
 		}
@@ -537,10 +568,10 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL);
 
 		/* need to update rmh->stat_len on the fly ?? */
-		if (i==0) {
+		if (!i) {
 			if (rmh->dsp_stat != RMH_SSIZE_FIXED) {
 				if (rmh->dsp_stat == RMH_SSIZE_ARG) {
-					rmh->stat_len = (u16)(data & 0x0000ff) + 1;
+					rmh->stat_len = (data & 0x0000ff) + 1;
 					data &= 0xffff00;
 				} else {
 					/* rmh->dsp_stat == RMH_SSIZE_MASK */
@@ -562,7 +593,8 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 			rmh->stat[i] = data;
 	}
 	if (rmh->stat_len > max_stat_len) {
-		snd_printdd("PCXHR : rmh->stat_len=%x too big\n", rmh->stat_len);
+		snd_printdd("PCXHR : rmh->stat_len=%x too big\n",
+			    rmh->stat_len);
 		rmh->stat_len = max_stat_len;
 	}
 	return 0;
@@ -605,7 +637,8 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		data &= 0xff7fff;	/* MASK_1_WORD_COMMAND */
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 	if (rmh->cmd_idx < CMD_LAST_INDEX)
-		snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]);
+		snd_printdd("MSG cmd[0]=%x (%s)\n",
+			    data, cmd_names[rmh->cmd_idx]);
 #endif
 
 	err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY,
@@ -619,8 +652,10 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 	if (rmh->cmd_len > 1) {
 		/* send length */
 		data = rmh->cmd_len - 1;
-		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY,
-					  PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &reg);
+		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR,
+					  PCXHR_ISR_HI08_TRDY,
+					  PCXHR_ISR_HI08_TRDY,
+					  PCXHR_TIMEOUT_DSP, &reg);
 		if (err)
 			return err;
 		PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, (data>>16)&0xFF);
@@ -653,8 +688,10 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 	/* test status ISR */
 	if (reg & PCXHR_ISR_HI08_ERR) {
 		/* ERROR, wait for receiver full */
-		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF,
-					  PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, &reg);
+		err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR,
+					  PCXHR_ISR_HI08_RXDF,
+					  PCXHR_ISR_HI08_RXDF,
+					  PCXHR_TIMEOUT_DSP, &reg);
 		if (err) {
 			snd_printk(KERN_ERR "ERROR RMH: ISR:RXDF=1 (ISR = %x)\n", reg);
 			return err;
@@ -663,7 +700,8 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		data  = PCXHR_INPB(mgr, PCXHR_DSP_TXH) << 16;
 		data |= PCXHR_INPB(mgr, PCXHR_DSP_TXM) << 8;
 		data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL);
-		snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n", rmh->cmd_idx, data);
+		snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n",
+			   rmh->cmd_idx, data);
 		err = -EINVAL;
 	} else {
 		/* read the response data */
@@ -732,8 +770,9 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr)
 {
 	int start_mask = PCXHR_INPL(mgr, PCXHR_PLX_MBOX2);
-	/* least segnificant 12 bits are the pipe states for the playback audios */
-	/* next 12 bits are the pipe states for the capture audios
+	/* least segnificant 12 bits are the pipe states
+	 * for the playback audios
+	 * next 12 bits are the pipe states for the capture audios
 	 * (PCXHR_PIPE_STATE_CAPTURE_OFFSET)
 	 */
 	start_mask &= 0xffffff;
@@ -744,7 +783,8 @@ static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr)
 #define PCXHR_PIPE_STATE_CAPTURE_OFFSET		12
 #define MAX_WAIT_FOR_DSP			20
 
-static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int *retry)
+static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr,
+				    int audio_mask, int *retry)
 {
 	struct pcxhr_rmh rmh;
 	int err;
@@ -760,17 +800,20 @@ static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int *
 			} else {
 				/* can start capture pipe */
 				pcxhr_set_pipe_cmd_params(&rmh, 1, audio -
-							  PCXHR_PIPE_STATE_CAPTURE_OFFSET,
-							  0, 0);
+						PCXHR_PIPE_STATE_CAPTURE_OFFSET,
+						0, 0);
 			}
 			err = pcxhr_send_msg(mgr, &rmh);
 			if (err) {
 				snd_printk(KERN_ERR
-					   "error pipe start (CMD_CAN_START_PIPE) err=%x!\n",
+					   "error pipe start "
+					   "(CMD_CAN_START_PIPE) err=%x!\n",
 					   err);
 				return err;
 			}
-			/* if the pipe couldn't be prepaired for start, retry it later */
+			/* if the pipe couldn't be prepaired for start,
+			 * retry it later
+			 */
 			if (rmh.stat[0] == 0)
 				*retry |= (1<<audio);
 		}
@@ -795,14 +838,14 @@ static int pcxhr_stop_pipes(struct pcxhr_mgr *mgr, int audio_mask)
 			} else {
 				/* stop capture pipe */
 				pcxhr_set_pipe_cmd_params(&rmh, 1, audio -
-							  PCXHR_PIPE_STATE_CAPTURE_OFFSET,
-							  0, 0);
+						PCXHR_PIPE_STATE_CAPTURE_OFFSET,
+						0, 0);
 			}
 			err = pcxhr_send_msg(mgr, &rmh);
 			if (err) {
 				snd_printk(KERN_ERR
-					   "error pipe stop (CMD_STOP_PIPE) err=%x!\n",
-					   err);
+					   "error pipe stop "
+					   "(CMD_STOP_PIPE) err=%x!\n", err);
 				return err;
 			}
 		}
@@ -822,15 +865,16 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask)
 		if (audio_mask & 1) {
 			pcxhr_init_rmh(&rmh, CMD_CONF_PIPE);
 			if (audio < PCXHR_PIPE_STATE_CAPTURE_OFFSET)
-				pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0, 1 << audio);
+				pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0,
+							  1 << audio);
 			else
 				pcxhr_set_pipe_cmd_params(&rmh, 1, 0, 0,
 							  1 << (audio - PCXHR_PIPE_STATE_CAPTURE_OFFSET));
 			err = pcxhr_send_msg(mgr, &rmh);
 			if (err) {
 				snd_printk(KERN_ERR
-					   "error pipe start (CMD_CONF_PIPE) err=%x!\n",
-					   err);
+					   "error pipe start "
+					   "(CMD_CONF_PIPE) err=%x!\n", err);
 				return err;
 			}
 		}
@@ -841,7 +885,9 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask)
 	pcxhr_init_rmh(&rmh, CMD_SEND_IRQA);
 	err = pcxhr_send_msg(mgr, &rmh);
 	if (err) {
-		snd_printk(KERN_ERR "error pipe start (CMD_SEND_IRQA) err=%x!\n", err );
+		snd_printk(KERN_ERR
+			   "error pipe start (CMD_SEND_IRQA) err=%x!\n",
+			   err);
 		return err;
 	}
 	return 0;
@@ -849,7 +895,8 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask)
 
 
 
-int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_mask, int start)
+int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask,
+			 int capture_mask, int start)
 {
 	int state, i, err;
 	int audio_mask;
@@ -858,21 +905,23 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
 	struct timeval my_tv1, my_tv2;
 	do_gettimeofday(&my_tv1);
 #endif
-	audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET));
+	audio_mask = (playback_mask |
+		      (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET));
 	/* current pipe state (playback + record) */
 	state = pcxhr_pipes_running(mgr);
 	snd_printdd("pcxhr_set_pipe_state %s (mask %x current %x)\n",
 		    start ? "START" : "STOP", audio_mask, state);
 	if (start) {
-		audio_mask &= ~state;	/* start only pipes that are not yet started */
+		/* start only pipes that are not yet started */
+		audio_mask &= ~state;
 		state = audio_mask;
 		for (i = 0; i < MAX_WAIT_FOR_DSP; i++) {
 			err = pcxhr_prepair_pipe_start(mgr, state, &state);
 			if (err)
 				return err;
 			if (state == 0)
-				break;	/* success, all pipes prepaired for start */
-			mdelay(1);		/* otherwise wait 1 millisecond and retry */
+				break;	/* success, all pipes prepaired */
+			mdelay(1);	/* wait 1 millisecond and retry */
 		}
 	} else {
 		audio_mask &= state;	/* stop only pipes that are started */
@@ -891,7 +940,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
 		if ((state & audio_mask) == (start ? audio_mask : 0))
 			break;
 		if (++i >= MAX_WAIT_FOR_DSP * 100) {
-			snd_printk(KERN_ERR "error pipe start/stop (ED_NO_RESPONSE_AT_IRQA)\n");
+			snd_printk(KERN_ERR "error pipe start/stop\n");
 			return -EBUSY;
 		}
 		udelay(10);			/* wait 10 microseconds */
@@ -918,7 +967,8 @@ int pcxhr_write_io_num_reg_cont(struct pcxhr_mgr *mgr, unsigned int mask,
 
 	spin_lock_irqsave(&mgr->msg_lock, flags);
 	if ((mgr->io_num_reg_cont & mask) == value) {
-		snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n", mask, value);
+		snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n",
+			    mask, value);
 		if (changed)
 			*changed = 0;
 		spin_unlock_irqrestore(&mgr->msg_lock, flags);
@@ -971,7 +1021,8 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err,
 		err = ((err >> 12) & 0xfff);
 	if (!err)
 		return 0;
-	snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n", err_src_name[err_src],
+	snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n",
+		    err_src_name[err_src],
 		    is_capture ? "Record" : "Play", pipe, err);
 	if (err == 0xe01)
 		mgr->async_err_stream_xrun++;
@@ -996,6 +1047,13 @@ void pcxhr_msg_tasklet(unsigned long arg)
 		snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n");
 	if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY)
 		snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n");
+	if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) {
+		/* clear events FREQ_CHANGE and TIME_CODE */
+		pcxhr_init_rmh(prmh, CMD_TEST_IT);
+		err = pcxhr_send_msg(mgr, prmh);
+		snd_printdd("CMD_TEST_IT : err=%x, stat=%x\n",
+			    err, prmh->stat[0]);
+	}
 	if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) {
 		snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n");
 
@@ -1005,18 +1063,22 @@ void pcxhr_msg_tasklet(unsigned long arg)
 		prmh->stat_len = PCXHR_SIZE_MAX_LONG_STATUS;
 		err = pcxhr_send_msg(mgr, prmh);
 		if (err)
-			snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n", err);
+			snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n",
+				   err);
 		i = 1;
 		while (i < prmh->stat_len) {
-			int nb_audio = (prmh->stat[i] >> FIELD_SIZE) & MASK_FIRST_FIELD;
-			int nb_stream = (prmh->stat[i] >> (2*FIELD_SIZE)) & MASK_FIRST_FIELD;
+			int nb_audio = ((prmh->stat[i] >> FIELD_SIZE) &
+					MASK_FIRST_FIELD);
+			int nb_stream = ((prmh->stat[i] >> (2*FIELD_SIZE)) &
+					 MASK_FIRST_FIELD);
 			int pipe = prmh->stat[i] & MASK_FIRST_FIELD;
 			int is_capture = prmh->stat[i] & 0x400000;
 			u32 err2;
 
 			if (prmh->stat[i] & 0x800000) {	/* if BIT_END */
 				snd_printdd("TASKLET : End%sPipe %d\n",
-					    is_capture ? "Record" : "Play", pipe);
+					    is_capture ? "Record" : "Play",
+					    pipe);
 			}
 			i++;
 			err2 = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1];
@@ -1062,7 +1124,7 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr,
 	pcxhr_init_rmh(&rmh, CMD_STREAM_SAMPLE_COUNT);
 	pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture,
 				  stream->pipe->first_audio, 0, stream_mask);
-	/* rmh.stat_len = 2; */		/* 2 resp data for each stream of the pipe */
+	/* rmh.stat_len = 2; */	/* 2 resp data for each stream of the pipe */
 
 	err = pcxhr_send_msg(mgr, &rmh);
 	if (err)
@@ -1072,18 +1134,21 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr,
 	hw_sample_count += (u_int64_t)rmh.stat[1];
 
 	snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n",
-		    stream->pipe->is_capture ? 'C':'P', stream->substream->number,
+		    stream->pipe->is_capture ? 'C' : 'P',
+		    stream->substream->number,
 		    (long unsigned int)hw_sample_count,
 		    (long unsigned int)(stream->timer_abs_periods +
-					stream->timer_period_frag + PCXHR_GRANULARITY));
-
+					stream->timer_period_frag +
+					mgr->granularity));
 	return hw_sample_count;
 }
 
 static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr,
-				   struct pcxhr_stream *stream, int samples_to_add)
+				   struct pcxhr_stream *stream,
+				   int samples_to_add)
 {
-	if (stream->substream && (stream->status == PCXHR_STREAM_STATUS_RUNNING)) {
+	if (stream->substream &&
+	    (stream->status == PCXHR_STREAM_STATUS_RUNNING)) {
 		u_int64_t new_sample_count;
 		int elapsed = 0;
 		int hardware_read = 0;
@@ -1092,20 +1157,22 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr,
 		if (samples_to_add < 0) {
 			stream->timer_is_synced = 0;
 			/* add default if no hardware_read possible */
-			samples_to_add = PCXHR_GRANULARITY;
+			samples_to_add = mgr->granularity;
 		}
 
 		if (!stream->timer_is_synced) {
-			if (stream->timer_abs_periods != 0 ||
-			    stream->timer_period_frag + PCXHR_GRANULARITY >=
-			    runtime->period_size) {
-				new_sample_count = pcxhr_stream_read_position(mgr, stream);
+			if ((stream->timer_abs_periods != 0) ||
+			    ((stream->timer_period_frag + samples_to_add) >=
+			    runtime->period_size)) {
+				new_sample_count =
+				  pcxhr_stream_read_position(mgr, stream);
 				hardware_read = 1;
-				if (new_sample_count >= PCXHR_GRANULARITY_MIN) {
-					/* sub security offset because of jitter and
-					 * finer granularity of dsp time (MBOX4)
+				if (new_sample_count >= mgr->granularity) {
+					/* sub security offset because of
+					 * jitter and finer granularity of
+					 * dsp time (MBOX4)
 					 */
-					new_sample_count -= PCXHR_GRANULARITY_MIN;
+					new_sample_count -= mgr->granularity;
 					stream->timer_is_synced = 1;
 				}
 			}
@@ -1128,12 +1195,15 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr,
 				stream->timer_buf_periods = 0;
 			stream->timer_abs_periods = new_elapse_pos;
 		}
-		if (new_sample_count >= stream->timer_abs_periods)
-			stream->timer_period_frag = (u_int32_t)(new_sample_count -
-								stream->timer_abs_periods);
-		else
-			snd_printk(KERN_ERR "ERROR new_sample_count too small ??? %lx\n",
+		if (new_sample_count >= stream->timer_abs_periods) {
+			stream->timer_period_frag =
+				(u_int32_t)(new_sample_count -
+					    stream->timer_abs_periods);
+		} else {
+			snd_printk(KERN_ERR
+				   "ERROR new_sample_count too small ??? %ld\n",
 				   (long unsigned int)new_sample_count);
+		}
 
 		if (elapsed) {
 			spin_unlock(&mgr->lock);
@@ -1143,7 +1213,6 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr,
 	}
 }
 
-
 irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 {
 	struct pcxhr_mgr *mgr = dev_id;
@@ -1156,7 +1225,8 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 	reg = PCXHR_INPL(mgr, PCXHR_PLX_IRQCS);
 	if (! (reg & PCXHR_IRQCS_ACTIVE_PCIDB)) {
 		spin_unlock(&mgr->lock);
-		return IRQ_NONE;	/* this device did not cause the interrupt */
+		/* this device did not cause the interrupt */
+		return IRQ_NONE;
 	}
 
 	/* clear interrupt */
@@ -1167,10 +1237,12 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 	if (reg & PCXHR_IRQ_TIMER) {
 		int timer_toggle = reg & PCXHR_IRQ_TIMER;
 		/* is a 24 bit counter */
-		int dsp_time_new = PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK;
+		int dsp_time_new =
+			PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK;
 		int dsp_time_diff = dsp_time_new - mgr->dsp_time_last;
 
-		if (dsp_time_diff < 0 && mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID) {
+		if ((dsp_time_diff < 0) &&
+		    (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) {
 			snd_printdd("ERROR DSP TIME old(%d) new(%d) -> "
 				    "resynchronize all streams\n",
 				    mgr->dsp_time_last, dsp_time_new);
@@ -1178,42 +1250,51 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 		}
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (dsp_time_diff == 0)
-			snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new);
-		else if (dsp_time_diff >= (2*PCXHR_GRANULARITY))
+			snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n",
+				    dsp_time_new);
+		else if (dsp_time_diff >= (2*mgr->granularity))
 			snd_printdd("ERROR DSP TIME TOO BIG old(%d) add(%d)\n",
-				    mgr->dsp_time_last, dsp_time_new - mgr->dsp_time_last);
+				    mgr->dsp_time_last,
+				    dsp_time_new - mgr->dsp_time_last);
+		else if (dsp_time_diff % mgr->granularity)
+			snd_printdd("ERROR DSP TIME increased by %d\n",
+				    dsp_time_diff);
 #endif
 		mgr->dsp_time_last = dsp_time_new;
 
-		if (timer_toggle == mgr->timer_toggle)
+		if (timer_toggle == mgr->timer_toggle) {
 			snd_printdd("ERROR TIMER TOGGLE\n");
+			mgr->dsp_time_err++;
+		}
 		mgr->timer_toggle = timer_toggle;
 
 		reg &= ~PCXHR_IRQ_TIMER;
 		for (i = 0; i < mgr->num_cards; i++) {
 			chip = mgr->chip[i];
 			for (j = 0; j < chip->nb_streams_capt; j++)
-				pcxhr_update_timer_pos(mgr, &chip->capture_stream[j],
-						       dsp_time_diff);
+				pcxhr_update_timer_pos(mgr,
+						&chip->capture_stream[j],
+						dsp_time_diff);
 		}
 		for (i = 0; i < mgr->num_cards; i++) {
 			chip = mgr->chip[i];
 			for (j = 0; j < chip->nb_streams_play; j++)
-				pcxhr_update_timer_pos(mgr, &chip->playback_stream[j],
-						       dsp_time_diff);
+				pcxhr_update_timer_pos(mgr,
+						&chip->playback_stream[j],
+						dsp_time_diff);
 		}
 	}
 	/* other irq's handled in the tasklet */
 	if (reg & PCXHR_IRQ_MASK) {
-
-		/* as we didn't request any notifications, some kind of xrun error
-		 * will probably occured
-		 */
-		/* better resynchronize all streams next interrupt : */
-		mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
-		
+		if (reg & PCXHR_IRQ_ASYNC) {
+			/* as we didn't request any async notifications,
+			 * some kind of xrun error will probably occured
+			 */
+			/* better resynchronize all streams next interrupt : */
+			mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
+		}
 		mgr->src_it_dsp = reg;
-		tasklet_hi_schedule(&mgr->msg_taskq);
+		tasklet_schedule(&mgr->msg_taskq);
 	}
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 	if (reg & PCXHR_FATAL_DSP_ERR)
diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
index d9a4ab609875..bbbd66d13a64 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -65,7 +65,7 @@ enum {
 	CMD_RESYNC_AUDIO_INPUTS,	/* cmd_len = 1	stat_len = 0 */
 	CMD_GET_DSP_RESOURCES,		/* cmd_len = 1	stat_len = 4 */
 	CMD_SET_TIMER_INTERRUPT,	/* cmd_len = 1	stat_len = 0 */
-	CMD_RES_PIPE,			/* cmd_len = 2	stat_len = 0 */
+	CMD_RES_PIPE,			/* cmd_len >=2	stat_len = 0 */
 	CMD_FREE_PIPE,			/* cmd_len = 1	stat_len = 0 */
 	CMD_CONF_PIPE,			/* cmd_len = 2	stat_len = 0 */
 	CMD_STOP_PIPE,			/* cmd_len = 1	stat_len = 0 */
@@ -96,6 +96,8 @@ void pcxhr_init_rmh(struct pcxhr_rmh *rmh, int cmd);
 void pcxhr_set_pipe_cmd_params(struct pcxhr_rmh* rmh, int capture, unsigned int param1,
 			       unsigned int param2, unsigned int param3);
 
+#define DSP_EXT_CMD_SET(x) (x->dsp_version > 0x012800)
+
 /*
  send the rmh
  */
@@ -110,6 +112,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh);
 #define IO_NUM_REG_STATUS		5
 #define IO_NUM_REG_CUER			10
 #define IO_NUM_UER_CHIP_REG		11
+#define IO_NUM_REG_CONFIG_SRC		12
 #define IO_NUM_REG_OUT_ANA_LEVEL	20
 #define IO_NUM_REG_IN_ANA_LEVEL		21
 
diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c
index 96640d9c227d..592743a298b0 100644
--- a/sound/pci/pcxhr/pcxhr_hwdep.c
+++ b/sound/pci/pcxhr/pcxhr_hwdep.c
@@ -31,6 +31,7 @@
 #include "pcxhr_mixer.h"
 #include "pcxhr_hwdep.h"
 #include "pcxhr_core.h"
+#include "pcxhr_mix22.h"
 
 
 #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE)
@@ -40,10 +41,10 @@
 #endif
 
 
+static int pcxhr_sub_init(struct pcxhr_mgr *mgr);
 /*
  * get basic information and init pcxhr card
  */
-
 static int pcxhr_init_board(struct pcxhr_mgr *mgr)
 {
 	int err;
@@ -68,7 +69,7 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr)
 	if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2)
 		return -EINVAL;
 	/* test 8 or 2 phys in */
-	if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) !=
+	if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) <
 	    mgr->capture_chips * 2)
 		return -EINVAL;
 	/* test max nb substream per board */
@@ -77,20 +78,34 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr)
 	/* test max nb substream per pipe */
 	if (((rmh.stat[1] >> 7) & 0x5F) < PCXHR_PLAYBACK_STREAMS)
 		return -EINVAL;
+	snd_printdd("supported formats : playback=%x capture=%x\n",
+		    rmh.stat[2], rmh.stat[3]);
 
 	pcxhr_init_rmh(&rmh, CMD_VERSION);
 	/* firmware num for DSP */
 	rmh.cmd[0] |= mgr->firmware_num;
 	/* transfer granularity in samples (should be multiple of 48) */
-	rmh.cmd[1] = (1<<23) + PCXHR_GRANULARITY;
+	rmh.cmd[1] = (1<<23) + mgr->granularity;
 	rmh.cmd_len = 2;
 	err = pcxhr_send_msg(mgr, &rmh);
 	if (err)
 		return err;
-	snd_printdd("PCXHR DSP version is %d.%d.%d\n",
-		    (rmh.stat[0]>>16)&0xff, (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff);
+	snd_printdd("PCXHR DSP version is %d.%d.%d\n", (rmh.stat[0]>>16)&0xff,
+		    (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff);
 	mgr->dsp_version = rmh.stat[0];
 
+	if (mgr->is_hr_stereo)
+		err = hr222_sub_init(mgr);
+	else
+		err = pcxhr_sub_init(mgr);
+	return err;
+}
+
+static int pcxhr_sub_init(struct pcxhr_mgr *mgr)
+{
+	int err;
+	struct pcxhr_rmh rmh;
+
 	/* get options */
 	pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
 	rmh.cmd[0] |= IO_NUM_REG_STATUS;
@@ -100,20 +115,22 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr)
 	if (err)
 		return err;
 
-	if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) == REG_STATUS_OPT_ANALOG_BOARD)
-		mgr->board_has_analog = 1;	/* analog addon board available */
-	else
-		/* analog addon board not available -> no support for instance */
-		return -EINVAL;	
+	if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) ==
+	    REG_STATUS_OPT_ANALOG_BOARD)
+		mgr->board_has_analog = 1;	/* analog addon board found */
 
 	/* unmute inputs */
 	err = pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS,
 					  REG_CONT_UNMUTE_INPUTS, NULL);
 	if (err)
 		return err;
-	/* unmute outputs */
-	pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* a write to IO_NUM_REG_MUTE_OUT mutes! */
+	/* unmute outputs (a write to IO_NUM_REG_MUTE_OUT mutes!) */
+	pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
 	rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+	if (DSP_EXT_CMD_SET(mgr)) {
+		rmh.cmd[1]  = 1;	/* unmute digital plugs */
+		rmh.cmd_len = 2;
+	}
 	err = pcxhr_send_msg(mgr, &rmh);
 	return err;
 }
@@ -124,19 +141,25 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr)
 
 	if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
 		/* mute outputs */
+	    if (!mgr->is_hr_stereo) {
 		/* a read to IO_NUM_REG_MUTE_OUT register unmutes! */
 		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
 		rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
 		pcxhr_send_msg(mgr, &rmh);
 		/* mute inputs */
-		pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, 0, NULL);
+		pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS,
+					    0, NULL);
+	    }
+		/* stereo cards mute with reset of dsp */
 	}
 	/* reset pcxhr dsp */
-	if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX))
+	if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX))
 		pcxhr_reset_dsp(mgr);
 	/* reset second xilinx */
-	if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_XLX_COM_INDEX))
+	if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_XLX_COM_INDEX)) {
 		pcxhr_reset_xilinx_com(mgr);
+		mgr->dsp_loaded = 1;
+	}
 	return;
 }
 
@@ -144,8 +167,9 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr)
 /*
  *  allocate a playback/capture pipe (pcmp0/pcmc0)
  */
-static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe,
-				    int is_capture, int pin)
+static int pcxhr_dsp_allocate_pipe(struct pcxhr_mgr *mgr,
+				   struct pcxhr_pipe *pipe,
+				   int is_capture, int pin)
 {
 	int stream_count, audio_count;
 	int err;
@@ -161,15 +185,23 @@ static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pi
 		stream_count = PCXHR_PLAYBACK_STREAMS;
 		audio_count = 2;	/* always stereo */
 	}
-	snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n", pin, is_capture ? 'c' : 'p');
+	snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n",
+		    pin, is_capture ? 'c' : 'p');
 	pipe->is_capture = is_capture;
 	pipe->first_audio = pin;
 	/* define pipe (P_PCM_ONLY_MASK (0x020000) is not necessary) */
 	pcxhr_init_rmh(&rmh, CMD_RES_PIPE);
-	pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, audio_count, stream_count); 
+	pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin,
+				  audio_count, stream_count);
+	rmh.cmd[1] |= 0x020000; /* add P_PCM_ONLY_MASK */
+	if (DSP_EXT_CMD_SET(mgr)) {
+		/* add channel mask to command */
+	  rmh.cmd[rmh.cmd_len++] = (audio_count == 1) ? 0x01 : 0x03;
+	}
 	err = pcxhr_send_msg(mgr, &rmh);
 	if (err < 0) {
-		snd_printk(KERN_ERR "error pipe allocation (CMD_RES_PIPE) err=%x!\n", err );
+		snd_printk(KERN_ERR "error pipe allocation "
+			   "(CMD_RES_PIPE) err=%x!\n", err);
 		return err;
 	}
 	pipe->status = PCXHR_PIPE_DEFINED;
@@ -199,10 +231,12 @@ static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe)
 		snd_printk(KERN_ERR "error stopping pipe!\n");
 	/* release the pipe */
 	pcxhr_init_rmh(&rmh, CMD_FREE_PIPE);
-	pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, 0, 0);
+	pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio,
+				  0, 0);
 	err = pcxhr_send_msg(mgr, &rmh);
 	if (err < 0)
-		snd_printk(KERN_ERR "error pipe release (CMD_FREE_PIPE) err(%x)\n", err);
+		snd_printk(KERN_ERR "error pipe release "
+			   "(CMD_FREE_PIPE) err(%x)\n", err);
 	pipe->status = PCXHR_PIPE_UNDEFINED;
 	return err;
 }
@@ -248,15 +282,16 @@ static int pcxhr_start_pipes(struct pcxhr_mgr *mgr)
 	for (i = 0; i < mgr->num_cards; i++) {
 		chip = mgr->chip[i];
 		if (chip->nb_streams_play)
-			playback_mask |= (1 << chip->playback_pipe.first_audio);
+			playback_mask |= 1 << chip->playback_pipe.first_audio;
 		for (j = 0; j < chip->nb_streams_capt; j++)
-			capture_mask |= (1 << chip->capture_pipe[j].first_audio);
+			capture_mask |= 1 << chip->capture_pipe[j].first_audio;
 	}
 	return pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1);
 }
 
 
-static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmware *dsp)
+static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index,
+			  const struct firmware *dsp)
 {
 	int err, card_index;
 
@@ -330,22 +365,33 @@ static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmwar
 
 int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 {
-	static char *fw_files[5] = {
-		"xi_1_882.dat",
-		"xc_1_882.dat",
-		"e321_512.e56",
-		"b321_512.b56",
-		"d321_512.d56"
+	static char *fw_files[][5] = {
+	[0] = { "xlxint.dat", "xlxc882hr.dat",
+		"dspe882.e56", "dspb882hr.b56", "dspd882.d56" },
+	[1] = { "xlxint.dat", "xlxc882e.dat",
+		"dspe882.e56", "dspb882e.b56", "dspd882.d56" },
+	[2] = { "xlxint.dat", "xlxc1222hr.dat",
+		"dspe882.e56", "dspb1222hr.b56", "dspd1222.d56" },
+	[3] = { "xlxint.dat", "xlxc1222e.dat",
+		"dspe882.e56", "dspb1222e.b56", "dspd1222.d56" },
+	[4] = { NULL, "xlxc222.dat",
+		"dspe924.e56", "dspb924.b56", "dspd222.d56" },
+	[5] = { NULL, "xlxc924.dat",
+		"dspe924.e56", "dspb924.b56", "dspd222.d56" },
 	};
 	char path[32];
 
 	const struct firmware *fw_entry;
 	int i, err;
+	int fw_set = mgr->fw_file_set;
 
-	for (i = 0; i < ARRAY_SIZE(fw_files); i++) {
-		sprintf(path, "pcxhr/%s", fw_files[i]);
+	for (i = 0; i < 5; i++) {
+		if (!fw_files[fw_set][i])
+			continue;
+		sprintf(path, "pcxhr/%s", fw_files[fw_set][i]);
 		if (request_firmware(&fw_entry, path, &mgr->pci->dev)) {
-			snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n", path);
+			snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n",
+				   path);
 			return -ENOENT;
 		}
 		/* fake hwdep dsp record */
@@ -358,11 +404,26 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 	return 0;
 }
 
-MODULE_FIRMWARE("pcxhr/xi_1_882.dat");
-MODULE_FIRMWARE("pcxhr/xc_1_882.dat");
-MODULE_FIRMWARE("pcxhr/e321_512.e56");
-MODULE_FIRMWARE("pcxhr/b321_512.b56");
-MODULE_FIRMWARE("pcxhr/d321_512.d56");
+MODULE_FIRMWARE("pcxhr/xlxint.dat");
+MODULE_FIRMWARE("pcxhr/xlxc882hr.dat");
+MODULE_FIRMWARE("pcxhr/xlxc882e.dat");
+MODULE_FIRMWARE("pcxhr/dspe882.e56");
+MODULE_FIRMWARE("pcxhr/dspb882hr.b56");
+MODULE_FIRMWARE("pcxhr/dspb882e.b56");
+MODULE_FIRMWARE("pcxhr/dspd882.d56");
+
+MODULE_FIRMWARE("pcxhr/xlxc1222hr.dat");
+MODULE_FIRMWARE("pcxhr/xlxc1222e.dat");
+MODULE_FIRMWARE("pcxhr/dspb1222hr.b56");
+MODULE_FIRMWARE("pcxhr/dspb1222e.b56");
+MODULE_FIRMWARE("pcxhr/dspd1222.d56");
+
+MODULE_FIRMWARE("pcxhr/xlxc222.dat");
+MODULE_FIRMWARE("pcxhr/xlxc924.dat");
+MODULE_FIRMWARE("pcxhr/dspe924.e56");
+MODULE_FIRMWARE("pcxhr/dspb924.b56");
+MODULE_FIRMWARE("pcxhr/dspd222.d56");
+
 
 #else /* old style firmware loading */
 
@@ -373,7 +434,8 @@ MODULE_FIRMWARE("pcxhr/d321_512.d56");
 static int pcxhr_hwdep_dsp_status(struct snd_hwdep *hw,
 				  struct snd_hwdep_dsp_status *info)
 {
-	strcpy(info->id, "pcxhr");
+	struct pcxhr_mgr *mgr = hw->private_data;
+	sprintf(info->id, "pcxhr%d", mgr->fw_file_set);
         info->num_dsps = PCXHR_FIRMWARE_FILES_MAX_INDEX;
 
 	if (hw->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))
@@ -393,8 +455,8 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw,
 	fw.size = dsp->length;
 	fw.data = vmalloc(fw.size);
 	if (! fw.data) {
-		snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%lu bytes)\n",
-			   (unsigned long)fw.size);
+		snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image "
+			   "(%lu bytes)\n", (unsigned long)fw.size);
 		return -ENOMEM;
 	}
 	if (copy_from_user((void *)fw.data, dsp->image, dsp->length)) {
@@ -424,8 +486,11 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 	int err;
 	struct snd_hwdep *hw;
 
-	/* only create hwdep interface for first cardX (see "index" module parameter)*/
-	if ((err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw)) < 0)
+	/* only create hwdep interface for first cardX
+	 * (see "index" module parameter)
+	 */
+	err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw);
+	if (err < 0)
 		return err;
 
 	hw->iface = SNDRV_HWDEP_IFACE_PCXHR;
@@ -435,10 +500,13 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 	hw->ops.dsp_status = pcxhr_hwdep_dsp_status;
 	hw->ops.dsp_load = pcxhr_hwdep_dsp_load;
 	hw->exclusive = 1;
+	/* stereo cards don't need fw_file_0 -> dsp_loaded = 1 */
+	hw->dsp_loaded = mgr->is_hr_stereo ? 1 : 0;
 	mgr->dsp_loaded = 0;
 	sprintf(hw->name, PCXHR_HWDEP_ID);
 
-	if ((err = snd_card_register(mgr->chip[0]->card)) < 0)
+	err = snd_card_register(mgr->chip[0]->card);
+	if (err < 0)
 		return err;
 	return 0;
 }
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
new file mode 100644
index 000000000000..ff019126b672
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -0,0 +1,820 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * mixer interface for stereo cards
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/asoundef.h>
+#include "pcxhr.h"
+#include "pcxhr_core.h"
+#include "pcxhr_mix22.h"
+
+
+/* registers used on the DSP and Xilinx (port 2) : HR stereo cards only */
+#define PCXHR_DSP_RESET		0x20
+#define PCXHR_XLX_CFG		0x24
+#define PCXHR_XLX_RUER		0x28
+#define PCXHR_XLX_DATA		0x2C
+#define PCXHR_XLX_STATUS	0x30
+#define PCXHR_XLX_LOFREQ	0x34
+#define PCXHR_XLX_HIFREQ	0x38
+#define PCXHR_XLX_CSUER		0x3C
+#define PCXHR_XLX_SELMIC	0x40
+
+#define PCXHR_DSP 2
+
+/* byte access only ! */
+#define PCXHR_INPB(mgr, x)	inb((mgr)->port[PCXHR_DSP] + (x))
+#define PCXHR_OUTPB(mgr, x, data) outb((data), (mgr)->port[PCXHR_DSP] + (x))
+
+
+/* values for PCHR_DSP_RESET register */
+#define PCXHR_DSP_RESET_DSP	0x01
+#define PCXHR_DSP_RESET_MUTE	0x02
+#define PCXHR_DSP_RESET_CODEC	0x08
+
+/* values for PCHR_XLX_CFG register */
+#define PCXHR_CFG_SYNCDSP_MASK		0x80
+#define PCXHR_CFG_DEPENDENCY_MASK	0x60
+#define PCXHR_CFG_INDEPENDANT_SEL	0x00
+#define PCXHR_CFG_MASTER_SEL		0x40
+#define PCXHR_CFG_SLAVE_SEL		0x20
+#define PCXHR_CFG_DATA_UER1_SEL_MASK	0x10	/* 0 (UER0), 1(UER1) */
+#define PCXHR_CFG_DATAIN_SEL_MASK	0x08	/* 0 (ana), 1 (UER) */
+#define PCXHR_CFG_SRC_MASK		0x04	/* 0 (Bypass), 1 (SRC Actif) */
+#define PCXHR_CFG_CLOCK_UER1_SEL_MASK	0x02	/* 0 (UER0), 1(UER1) */
+#define PCXHR_CFG_CLOCKIN_SEL_MASK	0x01	/* 0 (internal), 1 (AES/EBU) */
+
+/* values for PCHR_XLX_DATA register */
+#define PCXHR_DATA_CODEC	0x80
+#define AKM_POWER_CONTROL_CMD	0xA007
+#define AKM_RESET_ON_CMD	0xA100
+#define AKM_RESET_OFF_CMD	0xA103
+#define AKM_CLOCK_INF_55K_CMD	0xA240
+#define AKM_CLOCK_SUP_55K_CMD	0xA24D
+#define AKM_MUTE_CMD		0xA38D
+#define AKM_UNMUTE_CMD		0xA30D
+#define AKM_LEFT_LEVEL_CMD	0xA600
+#define AKM_RIGHT_LEVEL_CMD	0xA700
+
+/* values for PCHR_XLX_STATUS register - READ */
+#define PCXHR_STAT_SRC_LOCK		0x01
+#define PCXHR_STAT_LEVEL_IN		0x02
+#define PCXHR_STAT_MIC_CAPS		0x10
+/* values for PCHR_XLX_STATUS register - WRITE */
+#define PCXHR_STAT_FREQ_SYNC_MASK	0x01
+#define PCXHR_STAT_FREQ_UER1_MASK	0x02
+#define PCXHR_STAT_FREQ_SAVE_MASK	0x80
+
+/* values for PCHR_XLX_CSUER register */
+#define PCXHR_SUER1_BIT_U_READ_MASK	0x80
+#define PCXHR_SUER1_BIT_C_READ_MASK	0x40
+#define PCXHR_SUER1_DATA_PRESENT_MASK	0x20
+#define PCXHR_SUER1_CLOCK_PRESENT_MASK	0x10
+#define PCXHR_SUER_BIT_U_READ_MASK	0x08
+#define PCXHR_SUER_BIT_C_READ_MASK	0x04
+#define PCXHR_SUER_DATA_PRESENT_MASK	0x02
+#define PCXHR_SUER_CLOCK_PRESENT_MASK	0x01
+
+#define PCXHR_SUER_BIT_U_WRITE_MASK	0x02
+#define PCXHR_SUER_BIT_C_WRITE_MASK	0x01
+
+/* values for PCXHR_XLX_SELMIC register - WRITE */
+#define PCXHR_SELMIC_PREAMPLI_OFFSET	2
+#define PCXHR_SELMIC_PREAMPLI_MASK	0x0C
+#define PCXHR_SELMIC_PHANTOM_ALIM	0x80
+
+
+static const unsigned char g_hr222_p_level[] = {
+    0x00,   /* [000] -49.5 dB:	AKM[000] = -1.#INF dB	(mute) */
+    0x01,   /* [001] -49.0 dB:	AKM[001] = -48.131 dB	(diff=0.86920 dB) */
+    0x01,   /* [002] -48.5 dB:	AKM[001] = -48.131 dB	(diff=0.36920 dB) */
+    0x01,   /* [003] -48.0 dB:	AKM[001] = -48.131 dB	(diff=0.13080 dB) */
+    0x01,   /* [004] -47.5 dB:	AKM[001] = -48.131 dB	(diff=0.63080 dB) */
+    0x01,   /* [005] -46.5 dB:	AKM[001] = -48.131 dB	(diff=1.63080 dB) */
+    0x01,   /* [006] -47.0 dB:	AKM[001] = -48.131 dB	(diff=1.13080 dB) */
+    0x01,   /* [007] -46.0 dB:	AKM[001] = -48.131 dB	(diff=2.13080 dB) */
+    0x01,   /* [008] -45.5 dB:	AKM[001] = -48.131 dB	(diff=2.63080 dB) */
+    0x02,   /* [009] -45.0 dB:	AKM[002] = -42.110 dB	(diff=2.88980 dB) */
+    0x02,   /* [010] -44.5 dB:	AKM[002] = -42.110 dB	(diff=2.38980 dB) */
+    0x02,   /* [011] -44.0 dB:	AKM[002] = -42.110 dB	(diff=1.88980 dB) */
+    0x02,   /* [012] -43.5 dB:	AKM[002] = -42.110 dB	(diff=1.38980 dB) */
+    0x02,   /* [013] -43.0 dB:	AKM[002] = -42.110 dB	(diff=0.88980 dB) */
+    0x02,   /* [014] -42.5 dB:	AKM[002] = -42.110 dB	(diff=0.38980 dB) */
+    0x02,   /* [015] -42.0 dB:	AKM[002] = -42.110 dB	(diff=0.11020 dB) */
+    0x02,   /* [016] -41.5 dB:	AKM[002] = -42.110 dB	(diff=0.61020 dB) */
+    0x02,   /* [017] -41.0 dB:	AKM[002] = -42.110 dB	(diff=1.11020 dB) */
+    0x02,   /* [018] -40.5 dB:	AKM[002] = -42.110 dB	(diff=1.61020 dB) */
+    0x03,   /* [019] -40.0 dB:	AKM[003] = -38.588 dB	(diff=1.41162 dB) */
+    0x03,   /* [020] -39.5 dB:	AKM[003] = -38.588 dB	(diff=0.91162 dB) */
+    0x03,   /* [021] -39.0 dB:	AKM[003] = -38.588 dB	(diff=0.41162 dB) */
+    0x03,   /* [022] -38.5 dB:	AKM[003] = -38.588 dB	(diff=0.08838 dB) */
+    0x03,   /* [023] -38.0 dB:	AKM[003] = -38.588 dB	(diff=0.58838 dB) */
+    0x03,   /* [024] -37.5 dB:	AKM[003] = -38.588 dB	(diff=1.08838 dB) */
+    0x04,   /* [025] -37.0 dB:	AKM[004] = -36.090 dB	(diff=0.91040 dB) */
+    0x04,   /* [026] -36.5 dB:	AKM[004] = -36.090 dB	(diff=0.41040 dB) */
+    0x04,   /* [027] -36.0 dB:	AKM[004] = -36.090 dB	(diff=0.08960 dB) */
+    0x04,   /* [028] -35.5 dB:	AKM[004] = -36.090 dB	(diff=0.58960 dB) */
+    0x05,   /* [029] -35.0 dB:	AKM[005] = -34.151 dB	(diff=0.84860 dB) */
+    0x05,   /* [030] -34.5 dB:	AKM[005] = -34.151 dB	(diff=0.34860 dB) */
+    0x05,   /* [031] -34.0 dB:	AKM[005] = -34.151 dB	(diff=0.15140 dB) */
+    0x05,   /* [032] -33.5 dB:	AKM[005] = -34.151 dB	(diff=0.65140 dB) */
+    0x06,   /* [033] -33.0 dB:	AKM[006] = -32.568 dB	(diff=0.43222 dB) */
+    0x06,   /* [034] -32.5 dB:	AKM[006] = -32.568 dB	(diff=0.06778 dB) */
+    0x06,   /* [035] -32.0 dB:	AKM[006] = -32.568 dB	(diff=0.56778 dB) */
+    0x07,   /* [036] -31.5 dB:	AKM[007] = -31.229 dB	(diff=0.27116 dB) */
+    0x07,   /* [037] -31.0 dB:	AKM[007] = -31.229 dB	(diff=0.22884 dB) */
+    0x08,   /* [038] -30.5 dB:	AKM[008] = -30.069 dB	(diff=0.43100 dB) */
+    0x08,   /* [039] -30.0 dB:	AKM[008] = -30.069 dB	(diff=0.06900 dB) */
+    0x09,   /* [040] -29.5 dB:	AKM[009] = -29.046 dB	(diff=0.45405 dB) */
+    0x09,   /* [041] -29.0 dB:	AKM[009] = -29.046 dB	(diff=0.04595 dB) */
+    0x0a,   /* [042] -28.5 dB:	AKM[010] = -28.131 dB	(diff=0.36920 dB) */
+    0x0a,   /* [043] -28.0 dB:	AKM[010] = -28.131 dB	(diff=0.13080 dB) */
+    0x0b,   /* [044] -27.5 dB:	AKM[011] = -27.303 dB	(diff=0.19705 dB) */
+    0x0b,   /* [045] -27.0 dB:	AKM[011] = -27.303 dB	(diff=0.30295 dB) */
+    0x0c,   /* [046] -26.5 dB:	AKM[012] = -26.547 dB	(diff=0.04718 dB) */
+    0x0d,   /* [047] -26.0 dB:	AKM[013] = -25.852 dB	(diff=0.14806 dB) */
+    0x0e,   /* [048] -25.5 dB:	AKM[014] = -25.208 dB	(diff=0.29176 dB) */
+    0x0e,   /* [049] -25.0 dB:	AKM[014] = -25.208 dB	(diff=0.20824 dB) */
+    0x0f,   /* [050] -24.5 dB:	AKM[015] = -24.609 dB	(diff=0.10898 dB) */
+    0x10,   /* [051] -24.0 dB:	AKM[016] = -24.048 dB	(diff=0.04840 dB) */
+    0x11,   /* [052] -23.5 dB:	AKM[017] = -23.522 dB	(diff=0.02183 dB) */
+    0x12,   /* [053] -23.0 dB:	AKM[018] = -23.025 dB	(diff=0.02535 dB) */
+    0x13,   /* [054] -22.5 dB:	AKM[019] = -22.556 dB	(diff=0.05573 dB) */
+    0x14,   /* [055] -22.0 dB:	AKM[020] = -22.110 dB	(diff=0.11020 dB) */
+    0x15,   /* [056] -21.5 dB:	AKM[021] = -21.686 dB	(diff=0.18642 dB) */
+    0x17,   /* [057] -21.0 dB:	AKM[023] = -20.896 dB	(diff=0.10375 dB) */
+    0x18,   /* [058] -20.5 dB:	AKM[024] = -20.527 dB	(diff=0.02658 dB) */
+    0x1a,   /* [059] -20.0 dB:	AKM[026] = -19.831 dB	(diff=0.16866 dB) */
+    0x1b,   /* [060] -19.5 dB:	AKM[027] = -19.504 dB	(diff=0.00353 dB) */
+    0x1d,   /* [061] -19.0 dB:	AKM[029] = -18.883 dB	(diff=0.11716 dB) */
+    0x1e,   /* [062] -18.5 dB:	AKM[030] = -18.588 dB	(diff=0.08838 dB) */
+    0x20,   /* [063] -18.0 dB:	AKM[032] = -18.028 dB	(diff=0.02780 dB) */
+    0x22,   /* [064] -17.5 dB:	AKM[034] = -17.501 dB	(diff=0.00123 dB) */
+    0x24,   /* [065] -17.0 dB:	AKM[036] = -17.005 dB	(diff=0.00475 dB) */
+    0x26,   /* [066] -16.5 dB:	AKM[038] = -16.535 dB	(diff=0.03513 dB) */
+    0x28,   /* [067] -16.0 dB:	AKM[040] = -16.090 dB	(diff=0.08960 dB) */
+    0x2b,   /* [068] -15.5 dB:	AKM[043] = -15.461 dB	(diff=0.03857 dB) */
+    0x2d,   /* [069] -15.0 dB:	AKM[045] = -15.067 dB	(diff=0.06655 dB) */
+    0x30,   /* [070] -14.5 dB:	AKM[048] = -14.506 dB	(diff=0.00598 dB) */
+    0x33,   /* [071] -14.0 dB:	AKM[051] = -13.979 dB	(diff=0.02060 dB) */
+    0x36,   /* [072] -13.5 dB:	AKM[054] = -13.483 dB	(diff=0.01707 dB) */
+    0x39,   /* [073] -13.0 dB:	AKM[057] = -13.013 dB	(diff=0.01331 dB) */
+    0x3c,   /* [074] -12.5 dB:	AKM[060] = -12.568 dB	(diff=0.06778 dB) */
+    0x40,   /* [075] -12.0 dB:	AKM[064] = -12.007 dB	(diff=0.00720 dB) */
+    0x44,   /* [076] -11.5 dB:	AKM[068] = -11.481 dB	(diff=0.01937 dB) */
+    0x48,   /* [077] -11.0 dB:	AKM[072] = -10.984 dB	(diff=0.01585 dB) */
+    0x4c,   /* [078] -10.5 dB:	AKM[076] = -10.515 dB	(diff=0.01453 dB) */
+    0x51,   /* [079] -10.0 dB:	AKM[081] = -9.961 dB	(diff=0.03890 dB) */
+    0x55,   /* [080] -9.5 dB:	AKM[085] = -9.542 dB	(diff=0.04243 dB) */
+    0x5a,   /* [081] -9.0 dB:	AKM[090] = -9.046 dB	(diff=0.04595 dB) */
+    0x60,   /* [082] -8.5 dB:	AKM[096] = -8.485 dB	(diff=0.01462 dB) */
+    0x66,   /* [083] -8.0 dB:	AKM[102] = -7.959 dB	(diff=0.04120 dB) */
+    0x6c,   /* [084] -7.5 dB:	AKM[108] = -7.462 dB	(diff=0.03767 dB) */
+    0x72,   /* [085] -7.0 dB:	AKM[114] = -6.993 dB	(diff=0.00729 dB) */
+    0x79,   /* [086] -6.5 dB:	AKM[121] = -6.475 dB	(diff=0.02490 dB) */
+    0x80,   /* [087] -6.0 dB:	AKM[128] = -5.987 dB	(diff=0.01340 dB) */
+    0x87,   /* [088] -5.5 dB:	AKM[135] = -5.524 dB	(diff=0.02413 dB) */
+    0x8f,   /* [089] -5.0 dB:	AKM[143] = -5.024 dB	(diff=0.02408 dB) */
+    0x98,   /* [090] -4.5 dB:	AKM[152] = -4.494 dB	(diff=0.00607 dB) */
+    0xa1,   /* [091] -4.0 dB:	AKM[161] = -3.994 dB	(diff=0.00571 dB) */
+    0xaa,   /* [092] -3.5 dB:	AKM[170] = -3.522 dB	(diff=0.02183 dB) */
+    0xb5,   /* [093] -3.0 dB:	AKM[181] = -2.977 dB	(diff=0.02277 dB) */
+    0xbf,   /* [094] -2.5 dB:	AKM[191] = -2.510 dB	(diff=0.01014 dB) */
+    0xcb,   /* [095] -2.0 dB:	AKM[203] = -1.981 dB	(diff=0.01912 dB) */
+    0xd7,   /* [096] -1.5 dB:	AKM[215] = -1.482 dB	(diff=0.01797 dB) */
+    0xe3,   /* [097] -1.0 dB:	AKM[227] = -1.010 dB	(diff=0.01029 dB) */
+    0xf1,   /* [098] -0.5 dB:	AKM[241] = -0.490 dB	(diff=0.00954 dB) */
+    0xff,   /* [099] +0.0 dB:	AKM[255] = +0.000 dB	(diff=0.00000 dB) */
+};
+
+
+static void hr222_config_akm(struct pcxhr_mgr *mgr, unsigned short data)
+{
+	unsigned short mask = 0x8000;
+	/* activate access to codec registers */
+	PCXHR_INPB(mgr, PCXHR_XLX_HIFREQ);
+
+	while (mask) {
+		PCXHR_OUTPB(mgr, PCXHR_XLX_DATA,
+			    data & mask ? PCXHR_DATA_CODEC : 0);
+		mask >>= 1;
+	}
+	/* termiate access to codec registers */
+	PCXHR_INPB(mgr, PCXHR_XLX_RUER);
+}
+
+
+static int hr222_set_hw_playback_level(struct pcxhr_mgr *mgr,
+				       int idx, int level)
+{
+	unsigned short cmd;
+	if (idx > 1 ||
+	    level < 0 ||
+	    level >= ARRAY_SIZE(g_hr222_p_level))
+		return -EINVAL;
+
+	if (idx == 0)
+		cmd = AKM_LEFT_LEVEL_CMD;
+	else
+		cmd = AKM_RIGHT_LEVEL_CMD;
+
+	/* conversion from PmBoardCodedLevel to AKM nonlinear programming */
+	cmd += g_hr222_p_level[level];
+
+	hr222_config_akm(mgr, cmd);
+	return 0;
+}
+
+
+static int hr222_set_hw_capture_level(struct pcxhr_mgr *mgr,
+				      int level_l, int level_r, int level_mic)
+{
+	/* program all input levels at the same time */
+	unsigned int data;
+	int i;
+
+	if (!mgr->capture_chips)
+		return -EINVAL;	/* no PCX22 */
+
+	data  = ((level_mic & 0xff) << 24);	/* micro is mono, but apply */
+	data |= ((level_mic & 0xff) << 16);	/* level on both channels */
+	data |= ((level_r & 0xff) << 8);	/* line input right channel */
+	data |= (level_l & 0xff);		/* line input left channel */
+
+	PCXHR_INPB(mgr, PCXHR_XLX_DATA);	/* activate input codec */
+	/* send 32 bits (4 x 8 bits) */
+	for (i = 0; i < 32; i++, data <<= 1) {
+		PCXHR_OUTPB(mgr, PCXHR_XLX_DATA,
+			    (data & 0x80000000) ? PCXHR_DATA_CODEC : 0);
+	}
+	PCXHR_INPB(mgr, PCXHR_XLX_RUER);	/* close input level codec */
+	return 0;
+}
+
+static void hr222_micro_boost(struct pcxhr_mgr *mgr, int level);
+
+int hr222_sub_init(struct pcxhr_mgr *mgr)
+{
+	unsigned char reg;
+
+	mgr->board_has_analog = 1;	/* analog always available */
+	mgr->xlx_cfg = PCXHR_CFG_SYNCDSP_MASK;
+
+	reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS);
+	if (reg & PCXHR_STAT_MIC_CAPS)
+		mgr->board_has_mic = 1;	/* microphone available */
+	snd_printdd("MIC input available = %d\n", mgr->board_has_mic);
+
+	/* reset codec */
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET,
+		    PCXHR_DSP_RESET_DSP);
+	msleep(5);
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET,
+		    PCXHR_DSP_RESET_DSP  |
+		    PCXHR_DSP_RESET_MUTE |
+		    PCXHR_DSP_RESET_CODEC);
+	msleep(5);
+
+	/* config AKM */
+	hr222_config_akm(mgr, AKM_POWER_CONTROL_CMD);
+	hr222_config_akm(mgr, AKM_CLOCK_INF_55K_CMD);
+	hr222_config_akm(mgr, AKM_UNMUTE_CMD);
+	hr222_config_akm(mgr, AKM_RESET_OFF_CMD);
+
+	/* init micro boost */
+	hr222_micro_boost(mgr, 0);
+
+	return 0;
+}
+
+
+/* calc PLL register */
+/* TODO : there is a very similar fct in pcxhr.c */
+static int hr222_pll_freq_register(unsigned int freq,
+				   unsigned int *pllreg,
+				   unsigned int *realfreq)
+{
+	unsigned int reg;
+
+	if (freq < 6900 || freq > 219000)
+		return -EINVAL;
+	reg = (28224000 * 2) / freq;
+	reg = (reg - 1) / 2;
+	if (reg < 0x100)
+		*pllreg = reg + 0xC00;
+	else if (reg < 0x200)
+		*pllreg = reg + 0x800;
+	else if (reg < 0x400)
+		*pllreg = reg & 0x1ff;
+	else if (reg < 0x800) {
+		*pllreg = ((reg >> 1) & 0x1ff) + 0x200;
+		reg &= ~1;
+	} else {
+		*pllreg = ((reg >> 2) & 0x1ff) + 0x400;
+		reg &= ~3;
+	}
+	if (realfreq)
+		*realfreq = (28224000 / (reg + 1));
+	return 0;
+}
+
+int hr222_sub_set_clock(struct pcxhr_mgr *mgr,
+			unsigned int rate,
+			int *changed)
+{
+	unsigned int speed, pllreg = 0;
+	int err;
+	unsigned realfreq = rate;
+
+	switch (mgr->use_clock_type) {
+	case HR22_CLOCK_TYPE_INTERNAL:
+		err = hr222_pll_freq_register(rate, &pllreg, &realfreq);
+		if (err)
+			return err;
+
+		mgr->xlx_cfg &= ~(PCXHR_CFG_CLOCKIN_SEL_MASK |
+				  PCXHR_CFG_CLOCK_UER1_SEL_MASK);
+		break;
+	case HR22_CLOCK_TYPE_AES_SYNC:
+		mgr->xlx_cfg |= PCXHR_CFG_CLOCKIN_SEL_MASK;
+		mgr->xlx_cfg &= ~PCXHR_CFG_CLOCK_UER1_SEL_MASK;
+		break;
+	case HR22_CLOCK_TYPE_AES_1:
+		if (!mgr->board_has_aes1)
+			return -EINVAL;
+
+		mgr->xlx_cfg |= (PCXHR_CFG_CLOCKIN_SEL_MASK |
+				 PCXHR_CFG_CLOCK_UER1_SEL_MASK);
+		break;
+	default:
+		return -EINVAL;
+	}
+	hr222_config_akm(mgr, AKM_MUTE_CMD);
+
+	if (mgr->use_clock_type == HR22_CLOCK_TYPE_INTERNAL) {
+		PCXHR_OUTPB(mgr, PCXHR_XLX_HIFREQ, pllreg >> 8);
+		PCXHR_OUTPB(mgr, PCXHR_XLX_LOFREQ, pllreg & 0xff);
+	}
+
+	/* set clock source */
+	PCXHR_OUTPB(mgr, PCXHR_XLX_CFG, mgr->xlx_cfg);
+
+	/* codec speed modes */
+	speed = rate < 55000 ? 0 : 1;
+	if (mgr->codec_speed != speed) {
+		mgr->codec_speed = speed;
+		if (speed == 0)
+			hr222_config_akm(mgr, AKM_CLOCK_INF_55K_CMD);
+		else
+			hr222_config_akm(mgr, AKM_CLOCK_SUP_55K_CMD);
+	}
+
+	mgr->sample_rate_real = realfreq;
+	mgr->cur_clock_type = mgr->use_clock_type;
+
+	if (changed)
+		*changed = 1;
+
+	hr222_config_akm(mgr, AKM_UNMUTE_CMD);
+
+	snd_printdd("set_clock to %dHz (realfreq=%d pllreg=%x)\n",
+		    rate, realfreq, pllreg);
+	return 0;
+}
+
+int hr222_get_external_clock(struct pcxhr_mgr *mgr,
+			     enum pcxhr_clock_type clock_type,
+			     int *sample_rate)
+{
+	int rate, calc_rate = 0;
+	unsigned int ticks;
+	unsigned char mask, reg;
+
+	if (clock_type == HR22_CLOCK_TYPE_AES_SYNC) {
+
+		mask = (PCXHR_SUER_CLOCK_PRESENT_MASK |
+			PCXHR_SUER_DATA_PRESENT_MASK);
+		reg = PCXHR_STAT_FREQ_SYNC_MASK;
+
+	} else if (clock_type == HR22_CLOCK_TYPE_AES_1 && mgr->board_has_aes1) {
+
+		mask = (PCXHR_SUER1_CLOCK_PRESENT_MASK |
+			PCXHR_SUER1_DATA_PRESENT_MASK);
+		reg = PCXHR_STAT_FREQ_UER1_MASK;
+
+	} else {
+		snd_printdd("get_external_clock : type %d not supported\n",
+			    clock_type);
+		return -EINVAL; /* other clocks not supported */
+	}
+
+	if ((PCXHR_INPB(mgr, PCXHR_XLX_CSUER) & mask) != mask) {
+		snd_printdd("get_external_clock(%d) = 0 Hz\n", clock_type);
+		*sample_rate = 0;
+		return 0; /* no external clock locked */
+	}
+
+	PCXHR_OUTPB(mgr, PCXHR_XLX_STATUS, reg); /* calculate freq */
+
+	/* save the measured clock frequency */
+	reg |= PCXHR_STAT_FREQ_SAVE_MASK;
+
+	if (mgr->last_reg_stat != reg) {
+		udelay(500);	/* wait min 2 cycles of lowest freq (8000) */
+		mgr->last_reg_stat = reg;
+	}
+
+	PCXHR_OUTPB(mgr, PCXHR_XLX_STATUS, reg); /* save */
+
+	/* get the frequency */
+	ticks = (unsigned int)PCXHR_INPB(mgr, PCXHR_XLX_CFG);
+	ticks = (ticks & 0x03) << 8;
+	ticks |= (unsigned int)PCXHR_INPB(mgr, PCXHR_DSP_RESET);
+
+	if (ticks != 0)
+		calc_rate = 28224000 / ticks;
+	/* rounding */
+	if (calc_rate > 184200)
+		rate = 192000;
+	else if (calc_rate > 152200)
+		rate = 176400;
+	else if (calc_rate > 112000)
+		rate = 128000;
+	else if (calc_rate > 92100)
+		rate = 96000;
+	else if (calc_rate > 76100)
+		rate = 88200;
+	else if (calc_rate > 56000)
+		rate = 64000;
+	else if (calc_rate > 46050)
+		rate = 48000;
+	else if (calc_rate > 38050)
+		rate = 44100;
+	else if (calc_rate > 28000)
+		rate = 32000;
+	else if (calc_rate > 23025)
+		rate = 24000;
+	else if (calc_rate > 19025)
+		rate = 22050;
+	else if (calc_rate > 14000)
+		rate = 16000;
+	else if (calc_rate > 11512)
+		rate = 12000;
+	else if (calc_rate > 9512)
+		rate = 11025;
+	else if (calc_rate > 7000)
+		rate = 8000;
+	else
+		rate = 0;
+
+	snd_printdd("External clock is at %d Hz (measured %d Hz)\n",
+		    rate, calc_rate);
+	*sample_rate = rate;
+	return 0;
+}
+
+
+int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
+				    int is_capture, int channel)
+{
+	snd_printdd("hr222_update_analog_audio_level(%s chan=%d)\n",
+		    is_capture ? "capture" : "playback", channel);
+	if (is_capture) {
+		int level_l, level_r, level_mic;
+		/* we have to update all levels */
+		if (chip->analog_capture_active) {
+			level_l = chip->analog_capture_volume[0];
+			level_r = chip->analog_capture_volume[1];
+		} else {
+			level_l = HR222_LINE_CAPTURE_LEVEL_MIN;
+			level_r = HR222_LINE_CAPTURE_LEVEL_MIN;
+		}
+		if (chip->mic_active)
+			level_mic = chip->mic_volume;
+		else
+			level_mic = HR222_MICRO_CAPTURE_LEVEL_MIN;
+		return hr222_set_hw_capture_level(chip->mgr,
+						 level_l, level_r, level_mic);
+	} else {
+		int vol;
+		if (chip->analog_playback_active[channel])
+			vol = chip->analog_playback_volume[channel];
+		else
+			vol = HR222_LINE_PLAYBACK_LEVEL_MIN;
+		return hr222_set_hw_playback_level(chip->mgr, channel, vol);
+	}
+}
+
+
+/*texts[5] = {"Line", "Digital", "Digi+SRC", "Mic", "Line+Mic"}*/
+#define SOURCE_LINE	0
+#define SOURCE_DIGITAL	1
+#define SOURCE_DIGISRC	2
+#define SOURCE_MIC	3
+#define SOURCE_LINEMIC	4
+
+int hr222_set_audio_source(struct snd_pcxhr *chip)
+{
+	int digital = 0;
+	/* default analog source */
+	chip->mgr->xlx_cfg &= ~(PCXHR_CFG_SRC_MASK |
+				PCXHR_CFG_DATAIN_SEL_MASK |
+				PCXHR_CFG_DATA_UER1_SEL_MASK);
+
+	if (chip->audio_capture_source == SOURCE_DIGISRC) {
+		chip->mgr->xlx_cfg |= PCXHR_CFG_SRC_MASK;
+		digital = 1;
+	} else {
+		if (chip->audio_capture_source == SOURCE_DIGITAL)
+			digital = 1;
+	}
+	if (digital) {
+		chip->mgr->xlx_cfg |=  PCXHR_CFG_DATAIN_SEL_MASK;
+		if (chip->mgr->board_has_aes1) {
+			/* get data from the AES1 plug */
+			chip->mgr->xlx_cfg |= PCXHR_CFG_DATA_UER1_SEL_MASK;
+		}
+		/* chip->mic_active = 0; */
+		/* chip->analog_capture_active = 0; */
+	} else {
+		int update_lvl = 0;
+		chip->analog_capture_active = 0;
+		chip->mic_active = 0;
+		if (chip->audio_capture_source == SOURCE_LINE ||
+		    chip->audio_capture_source == SOURCE_LINEMIC) {
+			if (chip->analog_capture_active == 0)
+				update_lvl = 1;
+			chip->analog_capture_active = 1;
+		}
+		if (chip->audio_capture_source == SOURCE_MIC ||
+		    chip->audio_capture_source == SOURCE_LINEMIC) {
+			if (chip->mic_active == 0)
+				update_lvl = 1;
+			chip->mic_active = 1;
+		}
+		if (update_lvl) {
+			/* capture: update all 3 mutes/unmutes with one call */
+			hr222_update_analog_audio_level(chip, 1, 0);
+		}
+	}
+	/* set the source infos (max 3 bits modified) */
+	PCXHR_OUTPB(chip->mgr, PCXHR_XLX_CFG, chip->mgr->xlx_cfg);
+	return 0;
+}
+
+
+int hr222_iec958_capture_byte(struct snd_pcxhr *chip,
+			     int aes_idx, unsigned char *aes_bits)
+{
+	unsigned char idx = (unsigned char)(aes_idx * 8);
+	unsigned char temp = 0;
+	unsigned char mask = chip->mgr->board_has_aes1 ?
+		PCXHR_SUER1_BIT_C_READ_MASK : PCXHR_SUER_BIT_C_READ_MASK;
+	int i;
+	for (i = 0; i < 8; i++) {
+		PCXHR_OUTPB(chip->mgr, PCXHR_XLX_RUER, idx++); /* idx < 192 */
+		temp <<= 1;
+		if (PCXHR_INPB(chip->mgr, PCXHR_XLX_CSUER) & mask)
+			temp |= 1;
+	}
+	snd_printdd("read iec958 AES %d byte %d = 0x%x\n",
+		    chip->chip_idx, aes_idx, temp);
+	*aes_bits = temp;
+	return 0;
+}
+
+
+int hr222_iec958_update_byte(struct snd_pcxhr *chip,
+			     int aes_idx, unsigned char aes_bits)
+{
+	int i;
+	unsigned char new_bits = aes_bits;
+	unsigned char old_bits = chip->aes_bits[aes_idx];
+	unsigned char idx = (unsigned char)(aes_idx * 8);
+	for (i = 0; i < 8; i++) {
+		if ((old_bits & 0x01) != (new_bits & 0x01)) {
+			/* idx < 192 */
+			PCXHR_OUTPB(chip->mgr, PCXHR_XLX_RUER, idx);
+			/* write C and U bit */
+			PCXHR_OUTPB(chip->mgr, PCXHR_XLX_CSUER, new_bits&0x01 ?
+				    PCXHR_SUER_BIT_C_WRITE_MASK : 0);
+		}
+		idx++;
+		old_bits >>= 1;
+		new_bits >>= 1;
+	}
+	chip->aes_bits[aes_idx] = aes_bits;
+	return 0;
+}
+
+static void hr222_micro_boost(struct pcxhr_mgr *mgr, int level)
+{
+	unsigned char boost_mask;
+	boost_mask = (unsigned char) (level << PCXHR_SELMIC_PREAMPLI_OFFSET);
+	if (boost_mask & (~PCXHR_SELMIC_PREAMPLI_MASK))
+		return; /* only values form 0 to 3 accepted */
+
+	mgr->xlx_selmic &= ~PCXHR_SELMIC_PREAMPLI_MASK;
+	mgr->xlx_selmic |= boost_mask;
+
+	PCXHR_OUTPB(mgr, PCXHR_XLX_SELMIC, mgr->xlx_selmic);
+
+	snd_printdd("hr222_micro_boost : set %x\n", boost_mask);
+}
+
+static void hr222_phantom_power(struct pcxhr_mgr *mgr, int power)
+{
+	if (power)
+		mgr->xlx_selmic |= PCXHR_SELMIC_PHANTOM_ALIM;
+	else
+		mgr->xlx_selmic &= ~PCXHR_SELMIC_PHANTOM_ALIM;
+
+	PCXHR_OUTPB(mgr, PCXHR_XLX_SELMIC, mgr->xlx_selmic);
+
+	snd_printdd("hr222_phantom_power : set %d\n", power);
+}
+
+
+/* mic level */
+static const DECLARE_TLV_DB_SCALE(db_scale_mic_hr222, -9850, 50, 650);
+
+static int hr222_mic_vol_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = HR222_MICRO_CAPTURE_LEVEL_MIN; /* -98 dB */
+	/* gains from 9 dB to 31.5 dB not recommended; use micboost instead */
+	uinfo->value.integer.max = HR222_MICRO_CAPTURE_LEVEL_MAX; /*  +7 dB */
+	return 0;
+}
+
+static int hr222_mic_vol_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+	mutex_lock(&chip->mgr->mixer_mutex);
+	ucontrol->value.integer.value[0] = chip->mic_volume;
+	mutex_unlock(&chip->mgr->mixer_mutex);
+	return 0;
+}
+
+static int hr222_mic_vol_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+	int changed = 0;
+	mutex_lock(&chip->mgr->mixer_mutex);
+	if (chip->mic_volume != ucontrol->value.integer.value[0]) {
+		changed = 1;
+		chip->mic_volume = ucontrol->value.integer.value[0];
+		hr222_update_analog_audio_level(chip, 1, 0);
+	}
+	mutex_unlock(&chip->mgr->mixer_mutex);
+	return changed;
+}
+
+static struct snd_kcontrol_new hr222_control_mic_level = {
+	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.name =		"Mic Capture Volume",
+	.info =		hr222_mic_vol_info,
+	.get =		hr222_mic_vol_get,
+	.put =		hr222_mic_vol_put,
+	.tlv = { .p = db_scale_mic_hr222 },
+};
+
+
+/* mic boost level */
+static const DECLARE_TLV_DB_SCALE(db_scale_micboost_hr222, 0, 1800, 5400);
+
+static int hr222_mic_boost_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;	/*  0 dB */
+	uinfo->value.integer.max = 3;	/* 54 dB */
+	return 0;
+}
+
+static int hr222_mic_boost_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+	mutex_lock(&chip->mgr->mixer_mutex);
+	ucontrol->value.integer.value[0] = chip->mic_boost;
+	mutex_unlock(&chip->mgr->mixer_mutex);
+	return 0;
+}
+
+static int hr222_mic_boost_put(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+	int changed = 0;
+	mutex_lock(&chip->mgr->mixer_mutex);
+	if (chip->mic_boost != ucontrol->value.integer.value[0]) {
+		changed = 1;
+		chip->mic_boost = ucontrol->value.integer.value[0];
+		hr222_micro_boost(chip->mgr, chip->mic_boost);
+	}
+	mutex_unlock(&chip->mgr->mixer_mutex);
+	return changed;
+}
+
+static struct snd_kcontrol_new hr222_control_mic_boost = {
+	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.name =		"MicBoost Capture Volume",
+	.info =		hr222_mic_boost_info,
+	.get =		hr222_mic_boost_get,
+	.put =		hr222_mic_boost_put,
+	.tlv = { .p = db_scale_micboost_hr222 },
+};
+
+
+/******************* Phantom power switch *******************/
+#define hr222_phantom_power_info	snd_ctl_boolean_mono_info
+
+static int hr222_phantom_power_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+	mutex_lock(&chip->mgr->mixer_mutex);
+	ucontrol->value.integer.value[0] = chip->phantom_power;
+	mutex_unlock(&chip->mgr->mixer_mutex);
+	return 0;
+}
+
+static int hr222_phantom_power_put(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+	int power, changed = 0;
+
+	mutex_lock(&chip->mgr->mixer_mutex);
+	power = !!ucontrol->value.integer.value[0];
+	if (chip->phantom_power != power) {
+		hr222_phantom_power(chip->mgr, power);
+		chip->phantom_power = power;
+		changed = 1;
+	}
+	mutex_unlock(&chip->mgr->mixer_mutex);
+	return changed;
+}
+
+static struct snd_kcontrol_new hr222_phantom_power_switch = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Phantom Power Switch",
+	.info = hr222_phantom_power_info,
+	.get = hr222_phantom_power_get,
+	.put = hr222_phantom_power_put,
+};
+
+
+int hr222_add_mic_controls(struct snd_pcxhr *chip)
+{
+	int err;
+	if (!chip->mgr->board_has_mic)
+		return 0;
+
+	/* controls */
+	err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_control_mic_level,
+						   chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_control_mic_boost,
+						   chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_phantom_power_switch,
+						   chip));
+	return err;
+}
diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h
new file mode 100644
index 000000000000..6b318b2f0100
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -0,0 +1,56 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * low level interface with interrupt ans message handling
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#ifndef __SOUND_PCXHR_MIX22_H
+#define __SOUND_PCXHR_MIX22_H
+
+struct pcxhr_mgr;
+
+int hr222_sub_init(struct pcxhr_mgr *mgr);
+int hr222_sub_set_clock(struct pcxhr_mgr *mgr, unsigned int rate,
+			int *changed);
+int hr222_get_external_clock(struct pcxhr_mgr *mgr,
+			     enum pcxhr_clock_type clock_type,
+			     int *sample_rate);
+
+#define HR222_LINE_PLAYBACK_LEVEL_MIN		0	/* -25.5 dB */
+#define HR222_LINE_PLAYBACK_ZERO_LEVEL		51	/* 0.0 dB */
+#define HR222_LINE_PLAYBACK_LEVEL_MAX		99	/* +24.0 dB */
+
+#define HR222_LINE_CAPTURE_LEVEL_MIN		0	/* -111.5 dB */
+#define HR222_LINE_CAPTURE_ZERO_LEVEL		223	/* 0.0 dB */
+#define HR222_LINE_CAPTURE_LEVEL_MAX		255	/* +16 dB */
+#define HR222_MICRO_CAPTURE_LEVEL_MIN		0	/* -98.5 dB */
+#define HR222_MICRO_CAPTURE_LEVEL_MAX		210	/* +6.5 dB */
+
+int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
+				    int is_capture,
+				    int channel);
+int hr222_set_audio_source(struct snd_pcxhr *chip);
+int hr222_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx,
+			      unsigned char *aes_bits);
+int hr222_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx,
+			     unsigned char aes_bits);
+
+int hr222_add_mic_controls(struct snd_pcxhr *chip);
+
+#endif /* __SOUND_PCXHR_MIX22_H */
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index aabc7bc5321e..2436e374586f 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -33,20 +33,24 @@
 #include <sound/tlv.h>
 #include <sound/asoundef.h>
 #include "pcxhr_mixer.h"
+#include "pcxhr_mix22.h"
 
+#define PCXHR_LINE_CAPTURE_LEVEL_MIN   0	/* -112.0 dB */
+#define PCXHR_LINE_CAPTURE_LEVEL_MAX   255	/* +15.5 dB */
+#define PCXHR_LINE_CAPTURE_ZERO_LEVEL  224	/* 0.0 dB ( 0 dBu -> 0 dBFS ) */
 
-#define PCXHR_ANALOG_CAPTURE_LEVEL_MIN   0	/* -96.0 dB */
-#define PCXHR_ANALOG_CAPTURE_LEVEL_MAX   255	/* +31.5 dB */
-#define PCXHR_ANALOG_CAPTURE_ZERO_LEVEL  224	/* +16.0 dB ( +31.5 dB - fix level +15.5 dB ) */
+#define PCXHR_LINE_PLAYBACK_LEVEL_MIN  0	/* -104.0 dB */
+#define PCXHR_LINE_PLAYBACK_LEVEL_MAX  128	/* +24.0 dB */
+#define PCXHR_LINE_PLAYBACK_ZERO_LEVEL 104	/* 0.0 dB ( 0 dBFS -> 0 dBu ) */
 
-#define PCXHR_ANALOG_PLAYBACK_LEVEL_MIN  0	/* -128.0 dB */
-#define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX  128	/*    0.0 dB */
-#define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104	/*  -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */
-
-static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 3150);
+static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -11200, 50, 1550);
 static const DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -10400, 100, 2400);
 
-static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel)
+static const DECLARE_TLV_DB_SCALE(db_scale_a_hr222_capture, -11150, 50, 1600);
+static const DECLARE_TLV_DB_SCALE(db_scale_a_hr222_playback, -2550, 50, 2400);
+
+static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip,
+					   int is_capture, int channel)
 {
 	int err, vol;
 	struct pcxhr_rmh rmh;
@@ -60,15 +64,17 @@ static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_captur
 		if (chip->analog_playback_active[channel])
 			vol = chip->analog_playback_volume[channel];
 		else
-			vol = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN;
-		rmh.cmd[2] = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX - vol;	/* playback analog levels are inversed */
+			vol = PCXHR_LINE_PLAYBACK_LEVEL_MIN;
+		/* playback analog levels are inversed */
+		rmh.cmd[2] = PCXHR_LINE_PLAYBACK_LEVEL_MAX - vol;
 	}
 	rmh.cmd[1]  = 1 << ((2 * chip->chip_idx) + channel);	/* audio mask */
 	rmh.cmd_len = 3;
 	err = pcxhr_send_msg(chip->mgr, &rmh);
 	if (err < 0) {
-		snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d) "
-			   "is_capture(%d) err(%x)\n", chip->chip_idx, is_capture, err);
+		snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d)"
+			   " is_capture(%d) err(%x)\n",
+			   chip->chip_idx, is_capture, err);
 		return -EINVAL;
 	}
 	return 0;
@@ -80,14 +86,34 @@ static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_captur
 static int pcxhr_analog_vol_info(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = 2;
 	if (kcontrol->private_value == 0) {	/* playback */
-		uinfo->value.integer.min = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN;	/* -128 dB */
-		uinfo->value.integer.max = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX;	/* 0 dB */
+	    if (chip->mgr->is_hr_stereo) {
+		uinfo->value.integer.min =
+			HR222_LINE_PLAYBACK_LEVEL_MIN;	/* -25 dB */
+		uinfo->value.integer.max =
+			HR222_LINE_PLAYBACK_LEVEL_MAX;	/* +24 dB */
+	    } else {
+		uinfo->value.integer.min =
+			PCXHR_LINE_PLAYBACK_LEVEL_MIN;	/*-104 dB */
+		uinfo->value.integer.max =
+			PCXHR_LINE_PLAYBACK_LEVEL_MAX;	/* +24 dB */
+	    }
 	} else {				/* capture */
-		uinfo->value.integer.min = PCXHR_ANALOG_CAPTURE_LEVEL_MIN;	/* -96 dB */
-		uinfo->value.integer.max = PCXHR_ANALOG_CAPTURE_LEVEL_MAX;	/* 31.5 dB */
+	    if (chip->mgr->is_hr_stereo) {
+		uinfo->value.integer.min =
+			HR222_LINE_CAPTURE_LEVEL_MIN;	/*-112 dB */
+		uinfo->value.integer.max =
+			HR222_LINE_CAPTURE_LEVEL_MAX;	/* +15.5 dB */
+	    } else {
+		uinfo->value.integer.min =
+			PCXHR_LINE_CAPTURE_LEVEL_MIN;	/*-112 dB */
+		uinfo->value.integer.max =
+			PCXHR_LINE_CAPTURE_LEVEL_MAX;	/* +15.5 dB */
+	    }
 	}
 	return 0;
 }
@@ -98,11 +124,11 @@ static int pcxhr_analog_vol_get(struct snd_kcontrol *kcontrol,
 	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
 	mutex_lock(&chip->mgr->mixer_mutex);
 	if (kcontrol->private_value == 0) {	/* playback */
-		ucontrol->value.integer.value[0] = chip->analog_playback_volume[0];
-		ucontrol->value.integer.value[1] = chip->analog_playback_volume[1];
+	  ucontrol->value.integer.value[0] = chip->analog_playback_volume[0];
+	  ucontrol->value.integer.value[1] = chip->analog_playback_volume[1];
 	} else {				/* capture */
-		ucontrol->value.integer.value[0] = chip->analog_capture_volume[0];
-		ucontrol->value.integer.value[1] = chip->analog_capture_volume[1];
+	  ucontrol->value.integer.value[0] = chip->analog_capture_volume[0];
+	  ucontrol->value.integer.value[1] = chip->analog_capture_volume[1];
 	}
 	mutex_unlock(&chip->mgr->mixer_mutex);
 	return 0;
@@ -123,18 +149,35 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol,
 			&chip->analog_capture_volume[i] :
 			&chip->analog_playback_volume[i];
 		if (is_capture) {
-			if (new_volume < PCXHR_ANALOG_CAPTURE_LEVEL_MIN ||
-			    new_volume > PCXHR_ANALOG_CAPTURE_LEVEL_MAX)
-				continue;
+			if (chip->mgr->is_hr_stereo) {
+				if (new_volume < HR222_LINE_CAPTURE_LEVEL_MIN ||
+				    new_volume > HR222_LINE_CAPTURE_LEVEL_MAX)
+					continue;
+			} else {
+				if (new_volume < PCXHR_LINE_CAPTURE_LEVEL_MIN ||
+				    new_volume > PCXHR_LINE_CAPTURE_LEVEL_MAX)
+					continue;
+			}
 		} else {
-			if (new_volume < PCXHR_ANALOG_PLAYBACK_LEVEL_MIN ||
-			    new_volume > PCXHR_ANALOG_PLAYBACK_LEVEL_MAX)
-				continue;
+			if (chip->mgr->is_hr_stereo) {
+				if (new_volume < HR222_LINE_PLAYBACK_LEVEL_MIN ||
+				    new_volume > HR222_LINE_PLAYBACK_LEVEL_MAX)
+					continue;
+			} else {
+				if (new_volume < PCXHR_LINE_PLAYBACK_LEVEL_MIN ||
+				    new_volume > PCXHR_LINE_PLAYBACK_LEVEL_MAX)
+					continue;
+			}
 		}
 		if (*stored_volume != new_volume) {
 			*stored_volume = new_volume;
 			changed = 1;
-			pcxhr_update_analog_audio_level(chip, is_capture, i);
+			if (chip->mgr->is_hr_stereo)
+				hr222_update_analog_audio_level(chip,
+								is_capture, i);
+			else
+				pcxhr_update_analog_audio_level(chip,
+								is_capture, i);
 		}
 	}
 	mutex_unlock(&chip->mgr->mixer_mutex);
@@ -153,6 +196,7 @@ static struct snd_kcontrol_new pcxhr_control_analog_level = {
 };
 
 /* shared */
+
 #define pcxhr_sw_info		snd_ctl_boolean_stereo_info
 
 static int pcxhr_audio_sw_get(struct snd_kcontrol *kcontrol,
@@ -180,7 +224,10 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol,
 				!!ucontrol->value.integer.value[i];
 			changed = 1;
 			/* update playback levels */
-			pcxhr_update_analog_audio_level(chip, 0, i);
+			if (chip->mgr->is_hr_stereo)
+				hr222_update_analog_audio_level(chip, 0, i);
+			else
+				pcxhr_update_analog_audio_level(chip, 0, i);
 		}
 	}
 	mutex_unlock(&chip->mgr->mixer_mutex);
@@ -251,7 +298,8 @@ static int pcxhr_update_playback_stream_level(struct snd_pcxhr* chip, int idx)
 #define VALID_AUDIO_IO_MUTE_LEVEL	0x000004
 #define VALID_AUDIO_IO_MUTE_MONITOR_1	0x000008
 
-static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, int channel)
+static int pcxhr_update_audio_pipe_level(struct snd_pcxhr *chip,
+					 int capture, int channel)
 {
 	int err;
 	struct pcxhr_rmh rmh;
@@ -264,18 +312,20 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in
 
 	pcxhr_init_rmh(&rmh, CMD_AUDIO_LEVEL_ADJUST);
 	/* add channel mask */
-	pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0, 1 << (channel + pipe->first_audio));
-	/* TODO : if mask (3 << pipe->first_audio) is used, left and right channel
-	 * will be programmed to the same params
-	 */
+	pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0,
+				  1 << (channel + pipe->first_audio));
+	/* TODO : if mask (3 << pipe->first_audio) is used, left and right
+	 * channel will be programmed to the same params */
 	if (capture) {
 		rmh.cmd[0] |= VALID_AUDIO_IO_DIGITAL_LEVEL;
-		/* VALID_AUDIO_IO_MUTE_LEVEL not yet handled (capture pipe level) */
+		/* VALID_AUDIO_IO_MUTE_LEVEL not yet handled
+		 * (capture pipe level) */
 		rmh.cmd[2] = chip->digital_capture_volume[channel];
 	} else {
-		rmh.cmd[0] |= VALID_AUDIO_IO_MONITOR_LEVEL | VALID_AUDIO_IO_MUTE_MONITOR_1;
-		/* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL not yet
-		 * handled (playback pipe level)
+		rmh.cmd[0] |=	VALID_AUDIO_IO_MONITOR_LEVEL |
+				VALID_AUDIO_IO_MUTE_MONITOR_1;
+		/* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL
+		 * not yet handled (playback pipe level)
 		 */
 		rmh.cmd[2] = chip->monitoring_volume[channel] << 10;
 		if (chip->monitoring_active[channel] == 0)
@@ -284,8 +334,8 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in
 	rmh.cmd_len = 3;
 
 	err = pcxhr_send_msg(chip->mgr, &rmh);
-	if(err<0) {
-		snd_printk(KERN_DEBUG "error update_audio_level card(%d) err(%x)\n",
+	if (err < 0) {
+		snd_printk(KERN_DEBUG "error update_audio_level(%d) err=%x\n",
 			   chip->chip_idx, err);
 		return -EINVAL;
 	}
@@ -309,15 +359,15 @@ static int pcxhr_pcm_vol_get(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
-	int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);		/* index */
+	int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);	/* index */
 	int *stored_volume;
 	int is_capture = kcontrol->private_value;
 
 	mutex_lock(&chip->mgr->mixer_mutex);
-	if (is_capture)
-		stored_volume = chip->digital_capture_volume;		/* digital capture */
-	else
-		stored_volume = chip->digital_playback_volume[idx];	/* digital playback */
+	if (is_capture)		/* digital capture */
+		stored_volume = chip->digital_capture_volume;
+	else			/* digital playback */
+		stored_volume = chip->digital_playback_volume[idx];
 	ucontrol->value.integer.value[0] = stored_volume[0];
 	ucontrol->value.integer.value[1] = stored_volume[1];
 	mutex_unlock(&chip->mgr->mixer_mutex);
@@ -328,7 +378,7 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
-	int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);		/* index */
+	int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);	/* index */
 	int changed = 0;
 	int is_capture = kcontrol->private_value;
 	int *stored_volume;
@@ -384,7 +434,8 @@ static int pcxhr_pcm_sw_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
 	int changed = 0;
@@ -444,8 +495,8 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol,
 		if (chip->monitoring_volume[i] !=
 		    ucontrol->value.integer.value[i]) {
 			chip->monitoring_volume[i] =
-				!!ucontrol->value.integer.value[i];
-			if(chip->monitoring_active[i])
+				ucontrol->value.integer.value[i];
+			if (chip->monitoring_active[i])
 				/* update monitoring volume and mute */
 				/* do only when monitoring is unmuted */
 				pcxhr_update_audio_pipe_level(chip, 0, i);
@@ -460,7 +511,7 @@ static struct snd_kcontrol_new pcxhr_control_monitor_vol = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
-	.name =         "Monitoring Volume",
+	.name =         "Monitoring Playback Volume",
 	.info =		pcxhr_digital_vol_info,		/* shared */
 	.get =		pcxhr_monitor_vol_get,
 	.put =		pcxhr_monitor_vol_put,
@@ -511,7 +562,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol,
 
 static struct snd_kcontrol_new pcxhr_control_monitor_sw = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name =         "Monitoring Switch",
+	.name =         "Monitoring Playback Switch",
 	.info =         pcxhr_sw_info,		/* shared */
 	.get =          pcxhr_monitor_sw_get,
 	.put =          pcxhr_monitor_sw_put
@@ -533,7 +584,7 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip)
 	struct pcxhr_rmh rmh;
 	unsigned int mask, reg;
 	unsigned int codec;
-	int err, use_src, changed;
+	int err, changed;
 
 	switch (chip->chip_idx) {
 	case 0 : mask = PCXHR_SOURCE_AUDIO01_UER; codec = CS8420_01_CS; break;
@@ -542,13 +593,10 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip)
 	case 3 : mask = PCXHR_SOURCE_AUDIO67_UER; codec = CS8420_67_CS; break;
 	default: return -EINVAL;
 	}
-	reg = 0;	/* audio source from analog plug */
-	use_src = 0;	/* do not activate codec SRC */
-
 	if (chip->audio_capture_source != 0) {
 		reg = mask;	/* audio source from digital plug */
-		if (chip->audio_capture_source == 2)
-			use_src = 1;
+	} else {
+		reg = 0;	/* audio source from analog plug */
 	}
 	/* set the input source */
 	pcxhr_write_io_num_reg_cont(chip->mgr, mask, reg, &changed);
@@ -560,29 +608,61 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip)
 		if (err)
 			return err;
 	}
-	pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);	/* set codec SRC on off */
-	rmh.cmd_len = 3;
-	rmh.cmd[0] |= IO_NUM_UER_CHIP_REG;
-	rmh.cmd[1] = codec;
-	rmh.cmd[2] = (CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x54);
-	err = pcxhr_send_msg(chip->mgr, &rmh);
-	if(err)
-		return err;
-	rmh.cmd[2] = (CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x49);
-	err = pcxhr_send_msg(chip->mgr, &rmh);
+	if (chip->mgr->board_aes_in_192k) {
+		int i;
+		unsigned int src_config = 0xC0;
+		/* update all src configs with one call */
+		for (i = 0; (i < 4) && (i < chip->mgr->capture_chips); i++) {
+			if (chip->mgr->chip[i]->audio_capture_source == 2)
+				src_config |= (1 << (3 - i));
+		}
+		/* set codec SRC on off */
+		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+		rmh.cmd_len = 2;
+		rmh.cmd[0] |= IO_NUM_REG_CONFIG_SRC;
+		rmh.cmd[1] = src_config;
+		err = pcxhr_send_msg(chip->mgr, &rmh);
+	} else {
+		int use_src = 0;
+		if (chip->audio_capture_source == 2)
+			use_src = 1;
+		/* set codec SRC on off */
+		pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+		rmh.cmd_len = 3;
+		rmh.cmd[0] |= IO_NUM_UER_CHIP_REG;
+		rmh.cmd[1] = codec;
+		rmh.cmd[2] = ((CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) |
+			      (use_src ? 0x41 : 0x54));
+		err = pcxhr_send_msg(chip->mgr, &rmh);
+		if (err)
+			return err;
+		rmh.cmd[2] = ((CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) |
+			      (use_src ? 0x41 : 0x49));
+		err = pcxhr_send_msg(chip->mgr, &rmh);
+	}
 	return err;
 }
 
 static int pcxhr_audio_src_info(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = {"Analog", "Digital", "Digi+SRC"};
+	static const char *texts[5] = {
+		"Line", "Digital", "Digi+SRC", "Mic", "Line+Mic"
+	};
+	int i;
+	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
 
+	i = 2;			/* no SRC, no Mic available */
+	if (chip->mgr->board_has_aes1) {
+		i = 3;		/* SRC available */
+		if (chip->mgr->board_has_mic)
+			i = 5;	/* Mic and MicroMix available */
+	}
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
+	uinfo->value.enumerated.items = i;
+	if (uinfo->value.enumerated.item > (i-1))
+		uinfo->value.enumerated.item = i-1;
 	strcpy(uinfo->value.enumerated.name,
 		texts[uinfo->value.enumerated.item]);
 	return 0;
@@ -601,13 +681,21 @@ static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
 	int ret = 0;
-
-	if (ucontrol->value.enumerated.item[0] >= 3)
+	int i = 2;		/* no SRC, no Mic available */
+	if (chip->mgr->board_has_aes1) {
+		i = 3;		/* SRC available */
+		if (chip->mgr->board_has_mic)
+			i = 5;	/* Mic and MicroMix available */
+	}
+	if (ucontrol->value.enumerated.item[0] >= i)
 		return -EINVAL;
 	mutex_lock(&chip->mgr->mixer_mutex);
 	if (chip->audio_capture_source != ucontrol->value.enumerated.item[0]) {
 		chip->audio_capture_source = ucontrol->value.enumerated.item[0];
-		pcxhr_set_audio_source(chip);
+		if (chip->mgr->is_hr_stereo)
+			hr222_set_audio_source(chip);
+		else
+			pcxhr_set_audio_source(chip);
 		ret = 1;
 	}
 	mutex_unlock(&chip->mgr->mixer_mutex);
@@ -626,25 +714,46 @@ static struct snd_kcontrol_new pcxhr_control_audio_src = {
 /*
  * clock type selection
  * enum pcxhr_clock_type {
- *		PCXHR_CLOCK_TYPE_INTERNAL = 0,
- *		PCXHR_CLOCK_TYPE_WORD_CLOCK,
- *		PCXHR_CLOCK_TYPE_AES_SYNC,
- *		PCXHR_CLOCK_TYPE_AES_1,
- *		PCXHR_CLOCK_TYPE_AES_2,
- *		PCXHR_CLOCK_TYPE_AES_3,
- *		PCXHR_CLOCK_TYPE_AES_4,
- *	};
+ *	PCXHR_CLOCK_TYPE_INTERNAL = 0,
+ *	PCXHR_CLOCK_TYPE_WORD_CLOCK,
+ *	PCXHR_CLOCK_TYPE_AES_SYNC,
+ *	PCXHR_CLOCK_TYPE_AES_1,
+ *	PCXHR_CLOCK_TYPE_AES_2,
+ *	PCXHR_CLOCK_TYPE_AES_3,
+ *	PCXHR_CLOCK_TYPE_AES_4,
+ *	PCXHR_CLOCK_TYPE_MAX = PCXHR_CLOCK_TYPE_AES_4,
+ *	HR22_CLOCK_TYPE_INTERNAL = PCXHR_CLOCK_TYPE_INTERNAL,
+ *	HR22_CLOCK_TYPE_AES_SYNC,
+ *	HR22_CLOCK_TYPE_AES_1,
+ *	HR22_CLOCK_TYPE_MAX = HR22_CLOCK_TYPE_AES_1,
+ * };
  */
 
 static int pcxhr_clock_type_info(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[7] = {
-		"Internal", "WordClock", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4"
+	static const char *textsPCXHR[7] = {
+		"Internal", "WordClock", "AES Sync",
+		"AES 1", "AES 2", "AES 3", "AES 4"
 	};
+	static const char *textsHR22[3] = {
+		"Internal", "AES Sync", "AES 1"
+	};
+	const char **texts;
 	struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
-	int clock_items = 3 + mgr->capture_chips;
-
+	int clock_items = 2;	/* at least Internal and AES Sync clock */
+	if (mgr->board_has_aes1) {
+		clock_items += mgr->capture_chips;	/* add AES x */
+		if (!mgr->is_hr_stereo)
+			clock_items += 1;		/* add word clock */
+	}
+	if (mgr->is_hr_stereo) {
+		texts = textsHR22;
+		snd_BUG_ON(clock_items > (HR22_CLOCK_TYPE_MAX+1));
+	} else {
+		texts = textsPCXHR;
+		snd_BUG_ON(clock_items > (PCXHR_CLOCK_TYPE_MAX+1));
+	}
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = clock_items;
@@ -667,9 +776,13 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
 	struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
-	unsigned int clock_items = 3 + mgr->capture_chips;
 	int rate, ret = 0;
-
+	unsigned int clock_items = 2; /* at least Internal and AES Sync clock */
+	if (mgr->board_has_aes1) {
+		clock_items += mgr->capture_chips;	/* add AES x */
+		if (!mgr->is_hr_stereo)
+			clock_items += 1;		/* add word clock */
+	}
 	if (ucontrol->value.enumerated.item[0] >= clock_items)
 		return -EINVAL;
 	mutex_lock(&mgr->mixer_mutex);
@@ -677,7 +790,8 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
 		mutex_lock(&mgr->setup_mutex);
 		mgr->use_clock_type = ucontrol->value.enumerated.item[0];
 		if (mgr->use_clock_type)
-			pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate);
+			pcxhr_get_external_clock(mgr, mgr->use_clock_type,
+						 &rate);
 		else
 			rate = mgr->sample_rate;
 		if (rate) {
@@ -686,7 +800,7 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
 				mgr->sample_rate = rate;
 		}
 		mutex_unlock(&mgr->setup_mutex);
-		ret = 1;	/* return 1 even if the set was not done. ok ? */
+		ret = 1; /* return 1 even if the set was not done. ok ? */
 	}
 	mutex_unlock(&mgr->mixer_mutex);
 	return ret;
@@ -747,14 +861,16 @@ static struct snd_kcontrol_new pcxhr_control_clock_rate = {
 /*
  * IEC958 status bits
  */
-static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_info *uinfo)
 {
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
 	uinfo->count = 1;
 	return 0;
 }
 
-static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char* aes_bits)
+static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip,
+				     int aes_idx, unsigned char *aes_bits)
 {
 	int i, err;
 	unsigned char temp;
@@ -763,39 +879,61 @@ static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsign
 	pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
 	rmh.cmd[0] |= IO_NUM_UER_CHIP_REG;
 	switch (chip->chip_idx) {
-	case 0:	rmh.cmd[1] = CS8420_01_CS; break;	/* use CS8416_01_CS for AES SYNC plug */
+	  /* instead of CS8420_01_CS use CS8416_01_CS for AES SYNC plug */
+	case 0:	rmh.cmd[1] = CS8420_01_CS; break;
 	case 1:	rmh.cmd[1] = CS8420_23_CS; break;
 	case 2:	rmh.cmd[1] = CS8420_45_CS; break;
 	case 3:	rmh.cmd[1] = CS8420_67_CS; break;
 	default: return -EINVAL;
 	}
-	switch (aes_idx) {
-	case 0:	rmh.cmd[2] = CS8420_CSB0; break;	/* use CS8416_CSBx for AES SYNC plug */
-	case 1:	rmh.cmd[2] = CS8420_CSB1; break;
-	case 2:	rmh.cmd[2] = CS8420_CSB2; break;
-	case 3:	rmh.cmd[2] = CS8420_CSB3; break;
-	case 4:	rmh.cmd[2] = CS8420_CSB4; break;
-	default: return -EINVAL;
+	if (chip->mgr->board_aes_in_192k) {
+		switch (aes_idx) {
+		case 0:	rmh.cmd[2] = CS8416_CSB0; break;
+		case 1:	rmh.cmd[2] = CS8416_CSB1; break;
+		case 2:	rmh.cmd[2] = CS8416_CSB2; break;
+		case 3:	rmh.cmd[2] = CS8416_CSB3; break;
+		case 4:	rmh.cmd[2] = CS8416_CSB4; break;
+		default: return -EINVAL;
+		}
+	} else {
+		switch (aes_idx) {
+		  /* instead of CS8420_CSB0 use CS8416_CSBx for AES SYNC plug */
+		case 0:	rmh.cmd[2] = CS8420_CSB0; break;
+		case 1:	rmh.cmd[2] = CS8420_CSB1; break;
+		case 2:	rmh.cmd[2] = CS8420_CSB2; break;
+		case 3:	rmh.cmd[2] = CS8420_CSB3; break;
+		case 4:	rmh.cmd[2] = CS8420_CSB4; break;
+		default: return -EINVAL;
+		}
 	}
-	rmh.cmd[1] &= 0x0fffff;			/* size and code the chip id for the fpga */
-	rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI;	/* chip signature + map for spi read */
+	/* size and code the chip id for the fpga */
+	rmh.cmd[1] &= 0x0fffff;
+	/* chip signature + map for spi read */
+	rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI;
 	rmh.cmd_len = 3;
 	err = pcxhr_send_msg(chip->mgr, &rmh);
 	if (err)
 		return err;
-	temp = 0;
-	for (i = 0; i < 8; i++) {
-		/* attention : reversed bit order (not with CS8416_01_CS) */
-		temp <<= 1;
-		if (rmh.stat[1] & (1 << i))
-			temp |= 1;
+
+	if (chip->mgr->board_aes_in_192k) {
+		temp = (unsigned char)rmh.stat[1];
+	} else {
+		temp = 0;
+		/* reversed bit order (not with CS8416_01_CS) */
+		for (i = 0; i < 8; i++) {
+			temp <<= 1;
+			if (rmh.stat[1] & (1 << i))
+				temp |= 1;
+		}
 	}
-	snd_printdd("read iec958 AES %d byte %d = 0x%x\n", chip->chip_idx, aes_idx, temp);
+	snd_printdd("read iec958 AES %d byte %d = 0x%x\n",
+		    chip->chip_idx, aes_idx, temp);
 	*aes_bits = temp;
 	return 0;
 }
 
-static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
 	unsigned char aes_bits;
@@ -806,7 +944,12 @@ static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 		if (kcontrol->private_value == 0)	/* playback */
 			aes_bits = chip->aes_bits[i];
 		else {				/* capture */
-			err = pcxhr_iec958_capture_byte(chip, i, &aes_bits);
+			if (chip->mgr->is_hr_stereo)
+				err = hr222_iec958_capture_byte(chip, i,
+								&aes_bits);
+			else
+				err = pcxhr_iec958_capture_byte(chip, i,
+								&aes_bits);
 			if (err)
 				break;
 		}
@@ -825,7 +968,8 @@ static int pcxhr_iec958_mask_get(struct snd_kcontrol *kcontrol,
         return 0;
 }
 
-static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char aes_bits)
+static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip,
+				    int aes_idx, unsigned char aes_bits)
 {
 	int i, err, cmd;
 	unsigned char new_bits = aes_bits;
@@ -834,12 +978,12 @@ static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigne
 
 	for (i = 0; i < 8; i++) {
 		if ((old_bits & 0x01) != (new_bits & 0x01)) {
-			cmd = chip->chip_idx & 0x03;		/* chip index 0..3 */
-			if(chip->chip_idx > 3)
+			cmd = chip->chip_idx & 0x03;      /* chip index 0..3 */
+			if (chip->chip_idx > 3)
 				/* new bit used if chip_idx>3 (PCX1222HR) */
 				cmd |= 1 << 22;
-			cmd |= ((aes_idx << 3) + i) << 2;	/* add bit offset */
-			cmd |= (new_bits & 0x01) << 23;		/* add bit value */
+			cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */
+			cmd |= (new_bits & 0x01) << 23;   /* add bit value */
 			pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
 			rmh.cmd[0] |= IO_NUM_REG_CUER;
 			rmh.cmd[1] = cmd;
@@ -867,7 +1011,12 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol,
 	mutex_lock(&chip->mgr->mixer_mutex);
 	for (i = 0; i < 5; i++) {
 		if (ucontrol->value.iec958.status[i] != chip->aes_bits[i]) {
-			pcxhr_iec958_update_byte(chip, i, ucontrol->value.iec958.status[i]);
+			if (chip->mgr->is_hr_stereo)
+				hr222_iec958_update_byte(chip, i,
+					ucontrol->value.iec958.status[i]);
+			else
+				pcxhr_iec958_update_byte(chip, i,
+					ucontrol->value.iec958.status[i]);
 			changed = 1;
 		}
 	}
@@ -917,29 +1066,53 @@ static void pcxhr_init_audio_levels(struct snd_pcxhr *chip)
 			/* at boot time the digital volumes are unmuted 0dB */
 			for (j = 0; j < PCXHR_PLAYBACK_STREAMS; j++) {
 				chip->digital_playback_active[j][i] = 1;
-				chip->digital_playback_volume[j][i] = PCXHR_DIGITAL_ZERO_LEVEL;
+				chip->digital_playback_volume[j][i] =
+					PCXHR_DIGITAL_ZERO_LEVEL;
 			}
-			/* after boot, only two bits are set on the uer interface */
-			chip->aes_bits[0] = IEC958_AES0_PROFESSIONAL | IEC958_AES0_PRO_FS_48000;
-/* only for test purpose, remove later */
+			/* after boot, only two bits are set on the uer
+			 * interface
+			 */
+			chip->aes_bits[0] = (IEC958_AES0_PROFESSIONAL |
+					     IEC958_AES0_PRO_FS_48000);
 #ifdef CONFIG_SND_DEBUG
-			/* analog volumes for playback (is LEVEL_MIN after boot) */
+			/* analog volumes for playback
+			 * (is LEVEL_MIN after boot)
+			 */
 			chip->analog_playback_active[i] = 1;
-			chip->analog_playback_volume[i] = PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL;
-			pcxhr_update_analog_audio_level(chip, 0, i);
+			if (chip->mgr->is_hr_stereo)
+				chip->analog_playback_volume[i] =
+					HR222_LINE_PLAYBACK_ZERO_LEVEL;
+			else {
+				chip->analog_playback_volume[i] =
+					PCXHR_LINE_PLAYBACK_ZERO_LEVEL;
+				pcxhr_update_analog_audio_level(chip, 0, i);
+			}
 #endif
-/* test end */
+			/* stereo cards need to be initialised after boot */
+			if (chip->mgr->is_hr_stereo)
+				hr222_update_analog_audio_level(chip, 0, i);
 		}
 		if (chip->nb_streams_capt) {
 			/* at boot time the digital volumes are unmuted 0dB */
-			chip->digital_capture_volume[i] = PCXHR_DIGITAL_ZERO_LEVEL;
-/* only for test purpose, remove later */
+			chip->digital_capture_volume[i] =
+				PCXHR_DIGITAL_ZERO_LEVEL;
+			chip->analog_capture_active = 1;
 #ifdef CONFIG_SND_DEBUG
-			/* analog volumes for playback (is LEVEL_MIN after boot) */
-			chip->analog_capture_volume[i]  = PCXHR_ANALOG_CAPTURE_ZERO_LEVEL;
-			pcxhr_update_analog_audio_level(chip, 1, i);
+			/* analog volumes for playback
+			 * (is LEVEL_MIN after boot)
+			 */
+			if (chip->mgr->is_hr_stereo)
+				chip->analog_capture_volume[i] =
+					HR222_LINE_CAPTURE_ZERO_LEVEL;
+			else {
+				chip->analog_capture_volume[i] =
+					PCXHR_LINE_CAPTURE_ZERO_LEVEL;
+				pcxhr_update_analog_audio_level(chip, 1, i);
+			}
 #endif
-/* test end */
+			/* stereo cards need to be initialised after boot */
+			if (chip->mgr->is_hr_stereo)
+				hr222_update_analog_audio_level(chip, 1, i);
 		}
 	}
 
@@ -963,90 +1136,125 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr)
 			temp = pcxhr_control_analog_level;
 			temp.name = "Master Playback Volume";
 			temp.private_value = 0; /* playback */
-			temp.tlv.p = db_scale_analog_playback;
-			if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+			if (mgr->is_hr_stereo)
+				temp.tlv.p = db_scale_a_hr222_playback;
+			else
+				temp.tlv.p = db_scale_analog_playback;
+			err = snd_ctl_add(chip->card,
+					  snd_ctl_new1(&temp, chip));
+			if (err < 0)
 				return err;
+
 			/* output mute controls */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_output_switch,
-							    chip))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_output_switch,
+					     chip));
+			if (err < 0)
 				return err;
-			
+
 			temp = snd_pcxhr_pcm_vol;
 			temp.name = "PCM Playback Volume";
 			temp.count = PCXHR_PLAYBACK_STREAMS;
 			temp.private_value = 0; /* playback */
-			if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+			err = snd_ctl_add(chip->card,
+					  snd_ctl_new1(&temp, chip));
+			if (err < 0)
 				return err;
 
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_pcm_switch,
-							    chip))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_pcm_switch, chip));
+			if (err < 0)
 				return err;
 
 			/* IEC958 controls */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_playback_iec958_mask,
-							    chip))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_playback_iec958_mask,
+					     chip));
+			if (err < 0)
 				return err;
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_playback_iec958,
-							    chip))) < 0)
+
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_playback_iec958,
+					     chip));
+			if (err < 0)
 				return err;
 		}
 		if (chip->nb_streams_capt) {
-			/* analog input level control only on first two chips !*/
+			/* analog input level control */
 			temp = pcxhr_control_analog_level;
-			temp.name = "Master Capture Volume";
+			temp.name = "Line Capture Volume";
 			temp.private_value = 1; /* capture */
-			temp.tlv.p = db_scale_analog_capture;
-			if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+			if (mgr->is_hr_stereo)
+				temp.tlv.p = db_scale_a_hr222_capture;
+			else
+				temp.tlv.p = db_scale_analog_capture;
+
+			err = snd_ctl_add(chip->card,
+					  snd_ctl_new1(&temp, chip));
+			if (err < 0)
 				return err;
 
 			temp = snd_pcxhr_pcm_vol;
 			temp.name = "PCM Capture Volume";
 			temp.count = 1;
 			temp.private_value = 1; /* capture */
-			if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+
+			err = snd_ctl_add(chip->card,
+					  snd_ctl_new1(&temp, chip));
+			if (err < 0)
 				return err;
+
 			/* Audio source */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_audio_src,
-							    chip))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_audio_src, chip));
+			if (err < 0)
 				return err;
+
 			/* IEC958 controls */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_capture_iec958_mask,
-							    chip))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_capture_iec958_mask,
+					     chip));
+			if (err < 0)
 				return err;
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_capture_iec958,
-							    chip))) < 0)
+
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_capture_iec958,
+					     chip));
+			if (err < 0)
 				return err;
+
+			if (mgr->is_hr_stereo) {
+				err = hr222_add_mic_controls(chip);
+				if (err < 0)
+					return err;
+			}
 		}
 		/* monitoring only if playback and capture device available */
 		if (chip->nb_streams_capt > 0 && chip->nb_streams_play > 0) {
 			/* monitoring */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_monitor_vol,
-							    chip))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_monitor_vol, chip));
+			if (err < 0)
 				return err;
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_monitor_sw,
-							    chip))) < 0)
+
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_monitor_sw, chip));
+			if (err < 0)
 				return err;
 		}
 
 		if (i == 0) {
 			/* clock mode only one control per pcxhr */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_clock_type,
-							    mgr))) < 0)
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_clock_type, mgr));
+			if (err < 0)
 				return err;
-			/* non standard control used to scan the external clock presence/frequencies */
-			if ((err = snd_ctl_add(chip->card,
-					       snd_ctl_new1(&pcxhr_control_clock_rate,
-							    mgr))) < 0)
+			/* non standard control used to scan
+			 * the external clock presence/frequencies
+			 */
+			err = snd_ctl_add(chip->card,
+				snd_ctl_new1(&pcxhr_control_clock_rate, mgr));
+			if (err < 0)
 				return err;
 		}
 
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index e9f0706ed3e4..3caacfb9d8e0 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -172,7 +172,7 @@ MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver.");
 
 #define MAX_WRITE_RETRY  10	/* cmd interface limits */
 #define MAX_ERROR_COUNT  10
-#define CMDIF_TIMEOUT    500000
+#define CMDIF_TIMEOUT    50000
 #define RESET_TRIES      5
 
 #define READ_PORT_ULONG(p)     inl((unsigned long)&(p))
@@ -1754,7 +1754,7 @@ snd_riptide_interrupt(int irq, void *dev_id)
 		if (IS_EOBIRQ(cif->hwport) || IS_EOSIRQ(cif->hwport) ||
 		    IS_EOCIRQ(cif->hwport)) {
 			chip->handled_irqs++;
-			tasklet_hi_schedule(&chip->riptide_tq);
+			tasklet_schedule(&chip->riptide_tq);
 		}
 		if (chip->rmidi && IS_MPUIRQ(cif->hwport)) {
 			chip->handled_irqs++;
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 736246f98acc..44d0c15e2b71 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -1452,7 +1452,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i
 	if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0)
 		return -1;
 
-	sprintf (hdsp->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1);
+	sprintf(hdsp->midi[id].rmidi->name, "HDSP MIDI %d", id+1);
 	hdsp->midi[id].rmidi->private_data = &hdsp->midi[id];
 
 	snd_rawmidi_set_ops (hdsp->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_hdsp_midi_output);
@@ -3750,7 +3750,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 		}
 	}
 	if (hdsp->use_midi_tasklet && schedule)
-		tasklet_hi_schedule(&hdsp->midi_tasklet);
+		tasklet_schedule(&hdsp->midi_tasklet);
 	return IRQ_HANDLED;
 }
 
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 98762f909d64..71231cf1b2b0 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1293,7 +1293,7 @@ static int __devinit snd_hdspm_create_midi (struct snd_card *card,
 	if (err < 0)
 		return err;
 
-	sprintf (hdspm->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1);
+	sprintf(hdspm->midi[id].rmidi->name, "HDSPM MIDI %d", id+1);
 	hdspm->midi[id].rmidi->private_data = &hdspm->midi[id];
 
 	snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
@@ -3476,7 +3476,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
 		schedule = 1;
 	}
 	if (schedule)
-		tasklet_hi_schedule(&hdspm->midi_tasklet);
+		tasklet_schedule(&hdspm->midi_tasklet);
 	return IRQ_HANDLED;
 }
 
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
index fa4b11398b1f..ea903c8e90dd 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
@@ -41,7 +41,7 @@ irqreturn_t pdacf_interrupt(int irq, void *dev)
 		if (stat & PDAUDIOCF_IRQOVR)	/* should never happen */
 			snd_printk(KERN_ERR "PDAUDIOCF SRAM buffer overrun detected!\n");
 		if (chip->pcm_substream)
-			tasklet_hi_schedule(&chip->tq);
+			tasklet_schedule(&chip->tq);
 		if (!(stat & PDAUDIOCF_IRQAKM))
 			stat |= PDAUDIOCF_IRQAKM;	/* check rate */
 	}
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index a38c0c790d2b..af76ee862d27 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -1033,7 +1033,7 @@ static int __init snd_pmac_detect(struct snd_pmac *chip)
 	}
 	if (of_device_is_compatible(sound, "tumbler")) {
 		chip->model = PMAC_TUMBLER;
-		chip->can_capture = 0;  /* no capture */
+		chip->can_capture = machine_is_compatible("PowerMac4,2");
 		chip->can_duplex = 0;
 		// chip->can_byte_swap = 0; /* FIXME: check this */
 		chip->num_freqs = ARRAY_SIZE(tumbler_freqs);
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
index f746e15b8481..3eb223385416 100644
--- a/sound/ppc/tumbler.c
+++ b/sound/ppc/tumbler.c
@@ -875,7 +875,8 @@ static struct snd_kcontrol_new snapper_mixers[] __initdata = {
 	  .put = tumbler_put_master_switch
 	},
 	DEFINE_SNAPPER_MIX("PCM Playback Volume", 0, VOL_IDX_PCM),
-	DEFINE_SNAPPER_MIX("PCM Playback Volume", 1, VOL_IDX_PCM2),
+	/* Alternative PCM is assigned to Mic analog loopback on iBook G4 */
+	DEFINE_SNAPPER_MIX("Mic Playback Volume", 0, VOL_IDX_PCM2),
 	DEFINE_SNAPPER_MIX("Monitor Mix Volume", 0, VOL_IDX_ADC),
 	DEFINE_SNAPPER_MONO("Tone Control - Bass", bass),
 	DEFINE_SNAPPER_MONO("Tone Control - Treble", treble),
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 4dfda6674bec..ef025c66cc66 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -22,17 +22,16 @@ if SND_SOC
 config SND_SOC_AC97_BUS
 	bool
 
-# All the supported Soc's
-source "sound/soc/at32/Kconfig"
-source "sound/soc/at91/Kconfig"
+# All the supported SoCs
+source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
+source "sound/soc/blackfin/Kconfig"
+source "sound/soc/davinci/Kconfig"
+source "sound/soc/fsl/Kconfig"
+source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
-source "sound/soc/fsl/Kconfig"
-source "sound/soc/davinci/Kconfig"
-source "sound/soc/omap/Kconfig"
-source "sound/soc/blackfin/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index d849349f2c66..86a9b1f5b0f3 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,13 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
-obj-$(CONFIG_SND_SOC)	+= omap/ au1x/ blackfin/
+obj-$(CONFIG_SND_SOC)	+= codecs/
+obj-$(CONFIG_SND_SOC)	+= atmel/
+obj-$(CONFIG_SND_SOC)	+= au1x/
+obj-$(CONFIG_SND_SOC)	+= blackfin/
+obj-$(CONFIG_SND_SOC)	+= davinci/
+obj-$(CONFIG_SND_SOC)	+= fsl/
+obj-$(CONFIG_SND_SOC)	+= omap/
+obj-$(CONFIG_SND_SOC)	+= pxa/
+obj-$(CONFIG_SND_SOC)	+= s3c24xx/
+obj-$(CONFIG_SND_SOC)	+= sh/
diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig
deleted file mode 100644
index b0765e86c085..000000000000
--- a/sound/soc/at32/Kconfig
+++ /dev/null
@@ -1,34 +0,0 @@
-config SND_AT32_SOC
-        tristate "SoC Audio for the Atmel AT32 System-on-a-Chip"
-        depends on AVR32 && SND_SOC
-        help
-          Say Y or M if you want to add support for codecs attached to 
-          the AT32 SSC interface.  You will also need to
-          to select the audio interfaces to support below.
-
-
-config SND_AT32_SOC_SSC
-        tristate
-
-
-
-config SND_AT32_SOC_PLAYPAQ
-        tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_AT32_SOC && BOARD_PLAYPAQ
-        select SND_AT32_SOC_SSC
-        select SND_SOC_WM8510
-        help
-          Say Y or M here if you want to add support for SoC audio
-          on the LRS PlayPaq.
-
-
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
-        bool "Run CODEC on PlayPaq in slave mode"
-        depends on SND_AT32_SOC_PLAYPAQ
-        default n
-        help
-          Say Y if you want to run with the AT32 SSC generating the BCLK
-          and FRAME signals on the PlayPaq.  Unless you want to play
-          with the AT32 as the SSC master, you probably want to say N here,
-          as this will give you better sound quality.
diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile
deleted file mode 100644
index c03e55ececeb..000000000000
--- a/sound/soc/at32/Makefile
+++ /dev/null
@@ -1,11 +0,0 @@
-# AT32 Platform Support
-snd-soc-at32-objs := at32-pcm.o
-snd-soc-at32-ssc-objs := at32-ssc.o
-
-obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o
-obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o
-
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c
deleted file mode 100644
index c83584f989a9..000000000000
--- a/sound/soc/at32/at32-pcm.c
+++ /dev/null
@@ -1,492 +0,0 @@
-/* sound/soc/at32/at32-pcm.c
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * Note that this is basically a port of the sound/soc/at91-pcm.c to
- * the AVR32 kernel.  Thanks to Frank Mandarino for that code.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "at32-pcm.h"
-
-
-
-/*--------------------------------------------------------------------------*\
- * Hardware definition
-\*--------------------------------------------------------------------------*/
-/* TODO: These values were taken from the AT91 platform driver, check
- *	 them against real values for AT32
- */
-static const struct snd_pcm_hardware at32_pcm_hardware = {
-	.info = (SNDRV_PCM_INFO_MMAP |
-		 SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_INTERLEAVED |
-		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_PAUSE),
-
-	.formats = SNDRV_PCM_FMTBIT_S16,
-	.period_bytes_min = 32,
-	.period_bytes_max = 8192,	/* 512 frames * 16 bytes / frame */
-	.periods_min = 2,
-	.periods_max = 1024,
-	.buffer_bytes_max = 32 * 1024,
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * Data types
-\*--------------------------------------------------------------------------*/
-struct at32_runtime_data {
-	struct at32_pcm_dma_params *params;
-	dma_addr_t dma_buffer;	/* physical address of DMA buffer */
-	dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
-	size_t period_size;
-
-	dma_addr_t period_ptr;	/* physical address of next period */
-	int periods;		/* period index of period_ptr */
-
-	/* Save PDC registers (for power management) */
-	u32 pdc_xpr_save;
-	u32 pdc_xcr_save;
-	u32 pdc_xnpr_save;
-	u32 pdc_xncr_save;
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * Helper functions
-\*--------------------------------------------------------------------------*/
-static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *dmabuf = &substream->dma_buffer;
-	size_t size = at32_pcm_hardware.buffer_bytes_max;
-
-	dmabuf->dev.type = SNDRV_DMA_TYPE_DEV;
-	dmabuf->dev.dev = pcm->card->dev;
-	dmabuf->private_data = NULL;
-	dmabuf->area = dma_alloc_coherent(pcm->card->dev, size,
-					  &dmabuf->addr, GFP_KERNEL);
-	pr_debug("at32_pcm: preallocate_dma_buffer: "
-		 "area=%p, addr=%p, size=%ld\n",
-		 (void *)dmabuf->area, (void *)dmabuf->addr, size);
-
-	if (!dmabuf->area)
-		return -ENOMEM;
-
-	dmabuf->bytes = size;
-	return 0;
-}
-
-
-
-/*--------------------------------------------------------------------------*\
- * ISR
-\*--------------------------------------------------------------------------*/
-static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *rtd = substream->runtime;
-	struct at32_runtime_data *prtd = rtd->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-	static int count;
-
-	count++;
-	if (ssc_sr & params->mask->ssc_endbuf) {
-		pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
-			   substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			   "underrun" : "overrun", params->name, ssc_sr, count);
-
-		/* re-start the PDC */
-		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-			   params->mask->pdc_disable);
-		prtd->period_ptr += prtd->period_size;
-		if (prtd->period_ptr >= prtd->dma_buffer_end)
-			prtd->period_ptr = prtd->dma_buffer;
-
-
-		ssc_writex(params->ssc->regs, params->pdc->xpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xcr,
-			   prtd->period_size / params->pdc_xfer_size);
-		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-			   params->mask->pdc_enable);
-	}
-
-
-	if (ssc_sr & params->mask->ssc_endx) {
-		/* Load the PDC next pointer and counter registers */
-		prtd->period_ptr += prtd->period_size;
-		if (prtd->period_ptr >= prtd->dma_buffer_end)
-			prtd->period_ptr = prtd->dma_buffer;
-		ssc_writex(params->ssc->regs, params->pdc->xnpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xncr,
-			   prtd->period_size / params->pdc_xfer_size);
-	}
-
-
-	snd_pcm_period_elapsed(substream);
-}
-
-
-
-/*--------------------------------------------------------------------------*\
- * PCM operations
-\*--------------------------------------------------------------------------*/
-static int at32_pcm_hw_params(struct snd_pcm_substream *substream,
-			      struct snd_pcm_hw_params *params)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at32_runtime_data *prtd = runtime->private_data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-
-	/* this may get called several times by oss emulation
-	 * with different params
-	 */
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-	runtime->dma_bytes = params_buffer_bytes(params);
-
-	prtd->params = rtd->dai->cpu_dai->dma_data;
-	prtd->params->dma_intr_handler = at32_pcm_dma_irq;
-
-	prtd->dma_buffer = runtime->dma_addr;
-	prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
-	prtd->period_size = params_period_bytes(params);
-
-	pr_debug("hw_params: DMA for %s initialized "
-		 "(dma_bytes=%ld, period_size=%ld)\n",
-		 prtd->params->name, runtime->dma_bytes, prtd->period_size);
-
-	return 0;
-}
-
-
-
-static int at32_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	struct at32_runtime_data *prtd = substream->runtime->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-
-	if (params != NULL) {
-		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
-			   params->mask->pdc_disable);
-		prtd->params->dma_intr_handler = NULL;
-	}
-
-	return 0;
-}
-
-
-
-static int at32_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct at32_runtime_data *prtd = substream->runtime->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-
-	ssc_writex(params->ssc->regs, SSC_IDR,
-		   params->mask->ssc_endx | params->mask->ssc_endbuf);
-	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-		   params->mask->pdc_disable);
-
-	return 0;
-}
-
-
-static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_pcm_runtime *rtd = substream->runtime;
-	struct at32_runtime_data *prtd = rtd->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-	int ret = 0;
-
-	pr_debug("at32_pcm_trigger: buffer_size = %ld, "
-		 "dma_area = %p, dma_bytes = %ld\n",
-		 rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		prtd->period_ptr = prtd->dma_buffer;
-
-		ssc_writex(params->ssc->regs, params->pdc->xpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xcr,
-			   prtd->period_size / params->pdc_xfer_size);
-
-		prtd->period_ptr += prtd->period_size;
-		ssc_writex(params->ssc->regs, params->pdc->xnpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xncr,
-			   prtd->period_size / params->pdc_xfer_size);
-
-		pr_debug("trigger: period_ptr=%lx, xpr=%x, "
-			 "xcr=%d, xnpr=%x, xncr=%d\n",
-			 (unsigned long)prtd->period_ptr,
-			 ssc_readx(params->ssc->regs, params->pdc->xpr),
-			 ssc_readx(params->ssc->regs, params->pdc->xcr),
-			 ssc_readx(params->ssc->regs, params->pdc->xnpr),
-			 ssc_readx(params->ssc->regs, params->pdc->xncr));
-
-		ssc_writex(params->ssc->regs, SSC_IER,
-			   params->mask->ssc_endx | params->mask->ssc_endbuf);
-		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
-			   params->mask->pdc_enable);
-
-		pr_debug("sr=%x, imr=%x\n",
-			 ssc_readx(params->ssc->regs, SSC_SR),
-			 ssc_readx(params->ssc->regs, SSC_IER));
-		break;		/* SNDRV_PCM_TRIGGER_START */
-
-
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-			   params->mask->pdc_disable);
-		break;
-
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-			   params->mask->pdc_enable);
-		break;
-
-	default:
-		ret = -EINVAL;
-	}
-
-	return ret;
-}
-
-
-
-static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at32_runtime_data *prtd = runtime->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-	dma_addr_t ptr;
-	snd_pcm_uframes_t x;
-
-	ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
-	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
-
-	if (x == runtime->buffer_size)
-		x = 0;
-
-	return x;
-}
-
-
-
-static int at32_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at32_runtime_data *prtd;
-	int ret = 0;
-
-	snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware);
-
-	/* ensure that buffer size is a multiple of period size */
-	ret = snd_pcm_hw_constraint_integer(runtime,
-					    SNDRV_PCM_HW_PARAM_PERIODS);
-	if (ret < 0)
-		goto out;
-
-	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
-	if (prtd == NULL) {
-		ret = -ENOMEM;
-		goto out;
-	}
-	runtime->private_data = prtd;
-
-
-out:
-	return ret;
-}
-
-
-
-static int at32_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct at32_runtime_data *prtd = substream->runtime->private_data;
-
-	kfree(prtd);
-	return 0;
-}
-
-
-static int at32_pcm_mmap(struct snd_pcm_substream *substream,
-			 struct vm_area_struct *vma)
-{
-	return remap_pfn_range(vma, vma->vm_start,
-			       substream->dma_buffer.addr >> PAGE_SHIFT,
-			       vma->vm_end - vma->vm_start, vma->vm_page_prot);
-}
-
-
-
-static struct snd_pcm_ops at32_pcm_ops = {
-	.open = at32_pcm_open,
-	.close = at32_pcm_close,
-	.ioctl = snd_pcm_lib_ioctl,
-	.hw_params = at32_pcm_hw_params,
-	.hw_free = at32_pcm_hw_free,
-	.prepare = at32_pcm_prepare,
-	.trigger = at32_pcm_trigger,
-	.pointer = at32_pcm_pointer,
-	.mmap = at32_pcm_mmap,
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * ASoC platform driver
-\*--------------------------------------------------------------------------*/
-static u64 at32_pcm_dmamask = 0xffffffff;
-
-static int at32_pcm_new(struct snd_card *card,
-			struct snd_soc_dai *dai,
-			struct snd_pcm *pcm)
-{
-	int ret = 0;
-
-	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &at32_pcm_dmamask;
-	if (!card->dev->coherent_dma_mask)
-		card->dev->coherent_dma_mask = 0xffffffff;
-
-	if (dai->playback.channels_min) {
-		ret = at32_pcm_preallocate_dma_buffer(
-			  pcm, SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto out;
-	}
-
-	if (dai->capture.channels_min) {
-		pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n");
-		ret = at32_pcm_preallocate_dma_buffer(
-			  pcm, SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto out;
-	}
-
-
-out:
-	return ret;
-}
-
-
-
-static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
-	struct snd_pcm_substream *substream;
-	struct snd_dma_buffer *buf;
-	int stream;
-
-	for (stream = 0; stream < 2; stream++) {
-		substream = pcm->streams[stream].substream;
-		if (substream == NULL)
-			continue;
-
-		buf = &substream->dma_buffer;
-		if (!buf->area)
-			continue;
-		dma_free_coherent(pcm->card->dev, buf->bytes,
-				  buf->area, buf->addr);
-		buf->area = NULL;
-	}
-}
-
-
-
-#ifdef CONFIG_PM
-static int at32_pcm_suspend(struct platform_device *pdev,
-			    struct snd_soc_dai *dai)
-{
-	struct snd_pcm_runtime *runtime = dai->runtime;
-	struct at32_runtime_data *prtd;
-	struct at32_pcm_dma_params *params;
-
-	if (runtime == NULL)
-		return 0;
-	prtd = runtime->private_data;
-	params = prtd->params;
-
-	/* Disable the PDC and save the PDC registers */
-	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-		   params->mask->pdc_disable);
-
-	prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
-	prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
-	prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
-	prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
-
-	return 0;
-}
-
-
-
-static int at32_pcm_resume(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_pcm_runtime *runtime = dai->runtime;
-	struct at32_runtime_data *prtd;
-	struct at32_pcm_dma_params *params;
-
-	if (runtime == NULL)
-		return 0;
-	prtd = runtime->private_data;
-	params = prtd->params;
-
-	/* Restore the PDC registers and enable the PDC */
-	ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
-	ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
-	ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
-	ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
-
-	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable);
-	return 0;
-}
-#else /* CONFIG_PM */
-#  define at32_pcm_suspend	NULL
-#  define at32_pcm_resume	NULL
-#endif /* CONFIG_PM */
-
-
-
-struct snd_soc_platform at32_soc_platform = {
-	.name = "at32-audio",
-	.pcm_ops = &at32_pcm_ops,
-	.pcm_new = at32_pcm_new,
-	.pcm_free = at32_pcm_free_dma_buffers,
-	.suspend = at32_pcm_suspend,
-	.resume = at32_pcm_resume,
-};
-EXPORT_SYMBOL_GPL(at32_soc_platform);
-
-
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("Atmel AT32 PCM module");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h
deleted file mode 100644
index 2a52430417da..000000000000
--- a/sound/soc/at32/at32-pcm.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/* sound/soc/at32/at32-pcm.h
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef __SOUND_SOC_AT32_AT32_PCM_H
-#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__
-
-#include <linux/atmel-ssc.h>
-
-
-/*
- * Registers and status bits that are required by the PCM driver
- * TODO: Is ptcr really used?
- */
-struct at32_pdc_regs {
-	u32 xpr;		/* PDC RX/TX pointer */
-	u32 xcr;		/* PDC RX/TX counter */
-	u32 xnpr;		/* PDC next RX/TX pointer */
-	u32 xncr;		/* PDC next RX/TX counter */
-	u32 ptcr;		/* PDC transfer control */
-};
-
-
-
-/*
- * SSC mask info
- */
-struct at32_ssc_mask {
-	u32 ssc_enable;		/* SSC RX/TX enable */
-	u32 ssc_disable;	/* SSC RX/TX disable */
-	u32 ssc_endx;		/* SSC ENDTX or ENDRX */
-	u32 ssc_endbuf;		/* SSC TXBUFF or RXBUFF */
-	u32 pdc_enable;		/* PDC RX/TX enable */
-	u32 pdc_disable;	/* PDC RX/TX disable */
-};
-
-
-
-/*
- * This structure, shared between the PCM driver and the interface,
- * contains all information required by the PCM driver to perform the
- * PDC DMA operation.  All fields except dma_intr_handler() are initialized
- * by the interface.  The dms_intr_handler() pointer is set by the PCM
- * driver and called by the interface SSC interrupt handler if it is
- * non-NULL.
- */
-struct at32_pcm_dma_params {
-	char *name;		/* stream identifier */
-	int pdc_xfer_size;	/* PDC counter increment in bytes */
-	struct ssc_device *ssc;	/* SSC device for stream */
-	struct at32_pdc_regs *pdc;	/* PDC register info */
-	struct at32_ssc_mask *mask;	/* SSC mask info */
-	struct snd_pcm_substream *substream;
-	void (*dma_intr_handler) (u32, struct snd_pcm_substream *);
-};
-
-
-
-/*
- * The AT32 ASoC platform driver
- */
-extern struct snd_soc_platform at32_soc_platform;
-
-
-
-/*
- * SSC register access (since ssc_writel() / ssc_readl() require literal name)
- */
-#define ssc_readx(base, reg)            (__raw_readl((base) + (reg)))
-#define ssc_writex(base, reg, value)    __raw_writel((value), (base) + (reg))
-
-#endif /* __SOUND_SOC_AT32_AT32_PCM_H */
diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c
deleted file mode 100644
index 4ef6492c902e..000000000000
--- a/sound/soc/at32/at32-ssc.c
+++ /dev/null
@@ -1,849 +0,0 @@
-/* sound/soc/at32/at32-ssc.c
- * ASoC platform driver for AT32 using SSC as DAI
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * Note that this is basically a port of the sound/soc/at91-ssc.c to
- * the AVR32 kernel.  Thanks to Frank Mandarino for that code.
- */
-
-/* #define DEBUG */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
-#include <linux/clk.h>
-#include <linux/io.h>
-#include <linux/atmel_pdc.h>
-#include <linux/atmel-ssc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include "at32-pcm.h"
-#include "at32-ssc.h"
-
-
-
-/*-------------------------------------------------------------------------*\
- * Constants
-\*-------------------------------------------------------------------------*/
-#define NUM_SSC_DEVICES		3
-
-/*
- * SSC direction masks
- */
-#define SSC_DIR_MASK_UNUSED	0
-#define SSC_DIR_MASK_PLAYBACK	1
-#define SSC_DIR_MASK_CAPTURE	2
-
-/*
- * SSC register values that Atmel left out of <linux/atmel-ssc.h>.  These
- * are expected to be used with SSC_BF
- */
-/* START bit field values */
-#define SSC_START_CONTINUOUS	0
-#define SSC_START_TX_RX		1
-#define SSC_START_LOW_RF	2
-#define SSC_START_HIGH_RF	3
-#define SSC_START_FALLING_RF	4
-#define SSC_START_RISING_RF	5
-#define SSC_START_LEVEL_RF	6
-#define SSC_START_EDGE_RF	7
-#define SSS_START_COMPARE_0	8
-
-/* CKI bit field values */
-#define SSC_CKI_FALLING		0
-#define SSC_CKI_RISING		1
-
-/* CKO bit field values */
-#define SSC_CKO_NONE		0
-#define SSC_CKO_CONTINUOUS	1
-#define SSC_CKO_TRANSFER	2
-
-/* CKS bit field values */
-#define SSC_CKS_DIV		0
-#define SSC_CKS_CLOCK		1
-#define SSC_CKS_PIN		2
-
-/* FSEDGE bit field values */
-#define SSC_FSEDGE_POSITIVE	0
-#define SSC_FSEDGE_NEGATIVE	1
-
-/* FSOS bit field values */
-#define SSC_FSOS_NONE		0
-#define SSC_FSOS_NEGATIVE	1
-#define SSC_FSOS_POSITIVE	2
-#define SSC_FSOS_LOW		3
-#define SSC_FSOS_HIGH		4
-#define SSC_FSOS_TOGGLE		5
-
-#define START_DELAY		1
-
-
-
-/*-------------------------------------------------------------------------*\
- * Module data
-\*-------------------------------------------------------------------------*/
-/*
- * SSC PDC registered required by the PCM DMA engine
- */
-static struct at32_pdc_regs pdc_tx_reg = {
-	.xpr = SSC_PDC_TPR,
-	.xcr = SSC_PDC_TCR,
-	.xnpr = SSC_PDC_TNPR,
-	.xncr = SSC_PDC_TNCR,
-};
-
-
-
-static struct at32_pdc_regs pdc_rx_reg = {
-	.xpr = SSC_PDC_RPR,
-	.xcr = SSC_PDC_RCR,
-	.xnpr = SSC_PDC_RNPR,
-	.xncr = SSC_PDC_RNCR,
-};
-
-
-
-/*
- * SSC and PDC status bits for transmit and receive
- */
-static struct at32_ssc_mask ssc_tx_mask = {
-	.ssc_enable = SSC_BIT(CR_TXEN),
-	.ssc_disable = SSC_BIT(CR_TXDIS),
-	.ssc_endx = SSC_BIT(SR_ENDTX),
-	.ssc_endbuf = SSC_BIT(SR_TXBUFE),
-	.pdc_enable = SSC_BIT(PDC_PTCR_TXTEN),
-	.pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS),
-};
-
-
-
-static struct at32_ssc_mask ssc_rx_mask = {
-	.ssc_enable = SSC_BIT(CR_RXEN),
-	.ssc_disable = SSC_BIT(CR_RXDIS),
-	.ssc_endx = SSC_BIT(SR_ENDRX),
-	.ssc_endbuf = SSC_BIT(SR_RXBUFF),
-	.pdc_enable = SSC_BIT(PDC_PTCR_RXTEN),
-	.pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS),
-};
-
-
-
-/*
- * DMA parameters for each SSC
- */
-static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
-	{
-	 {
-	  .name = "SSC0 PCM out",
-	  .pdc = &pdc_tx_reg,
-	  .mask = &ssc_tx_mask,
-	  },
-	 {
-	  .name = "SSC0 PCM in",
-	  .pdc = &pdc_rx_reg,
-	  .mask = &ssc_rx_mask,
-	  },
-	 },
-	{
-	 {
-	  .name = "SSC1 PCM out",
-	  .pdc = &pdc_tx_reg,
-	  .mask = &ssc_tx_mask,
-	  },
-	 {
-	  .name = "SSC1 PCM in",
-	  .pdc = &pdc_rx_reg,
-	  .mask = &ssc_rx_mask,
-	  },
-	 },
-	{
-	 {
-	  .name = "SSC2 PCM out",
-	  .pdc = &pdc_tx_reg,
-	  .mask = &ssc_tx_mask,
-	  },
-	 {
-	  .name = "SSC2 PCM in",
-	  .pdc = &pdc_rx_reg,
-	  .mask = &ssc_rx_mask,
-	  },
-	 },
-};
-
-
-
-static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = {
-	{
-	 .name = "ssc0",
-	 .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
-	 .dir_mask = SSC_DIR_MASK_UNUSED,
-	 .initialized = 0,
-	 },
-	{
-	 .name = "ssc1",
-	 .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
-	 .dir_mask = SSC_DIR_MASK_UNUSED,
-	 .initialized = 0,
-	 },
-	{
-	 .name = "ssc2",
-	 .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
-	 .dir_mask = SSC_DIR_MASK_UNUSED,
-	 .initialized = 0,
-	 },
-};
-
-
-
-
-/*-------------------------------------------------------------------------*\
- * ISR
-\*-------------------------------------------------------------------------*/
-/*
- * SSC interrupt handler.  Passes PDC interrupts to the DMA interrupt
- * handler in the PCM driver.
- */
-static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id)
-{
-	struct at32_ssc_info *ssc_p = dev_id;
-	struct at32_pcm_dma_params *dma_params;
-	u32 ssc_sr;
-	u32 ssc_substream_mask;
-	int i;
-
-	ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) &
-		  ssc_readl(ssc_p->ssc->regs, IMR));
-
-	/*
-	 * Loop through substreams attached to this SSC.  If a DMA-related
-	 * interrupt occured on that substream, call the DMA interrupt
-	 * handler function, if one has been registered in the dma_param
-	 * structure by the PCM driver.
-	 */
-	for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
-		dma_params = ssc_p->dma_params[i];
-
-		if ((dma_params != NULL) &&
-		    (dma_params->dma_intr_handler != NULL)) {
-			ssc_substream_mask = (dma_params->mask->ssc_endx |
-					      dma_params->mask->ssc_endbuf);
-			if (ssc_sr & ssc_substream_mask) {
-				dma_params->dma_intr_handler(ssc_sr,
-							     dma_params->
-							     substream);
-			}
-		}
-	}
-
-
-	return IRQ_HANDLED;
-}
-
-/*-------------------------------------------------------------------------*\
- * DAI functions
-\*-------------------------------------------------------------------------*/
-/*
- * Startup.  Only that one substream allowed in each direction.
- */
-static int at32_ssc_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	int dir_mask;
-
-	dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		    SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE);
-
-	spin_lock_irq(&ssc_p->lock);
-	if (ssc_p->dir_mask & dir_mask) {
-		spin_unlock_irq(&ssc_p->lock);
-		return -EBUSY;
-	}
-	ssc_p->dir_mask |= dir_mask;
-	spin_unlock_irq(&ssc_p->lock);
-
-	return 0;
-}
-
-
-
-/*
- * Shutdown.  Clear DMA parameters and shutdown the SSC if there
- * are no other substreams open.
- */
-static void at32_ssc_shutdown(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	struct at32_pcm_dma_params *dma_params;
-	int dir_mask;
-
-	dma_params = ssc_p->dma_params[substream->stream];
-
-	if (dma_params != NULL) {
-		ssc_writel(dma_params->ssc->regs, CR,
-			   dma_params->mask->ssc_disable);
-		pr_debug("%s disabled SSC_SR=0x%08x\n",
-			 (substream->stream ? "receiver" : "transmit"),
-			 ssc_readl(ssc_p->ssc->regs, SR));
-
-		dma_params->ssc = NULL;
-		dma_params->substream = NULL;
-		ssc_p->dma_params[substream->stream] = NULL;
-	}
-
-
-	dir_mask = 1 << substream->stream;
-	spin_lock_irq(&ssc_p->lock);
-	ssc_p->dir_mask &= ~dir_mask;
-	if (!ssc_p->dir_mask) {
-		/* Shutdown the SSC clock */
-		pr_debug("at32-ssc: Stopping user %d clock\n",
-			 ssc_p->ssc->user);
-		clk_disable(ssc_p->ssc->clk);
-
-		if (ssc_p->initialized) {
-			free_irq(ssc_p->ssc->irq, ssc_p);
-			ssc_p->initialized = 0;
-		}
-
-		/* Reset the SSC */
-		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
-		/* clear the SSC dividers */
-		ssc_p->cmr_div = 0;
-		ssc_p->tcmr_period = 0;
-		ssc_p->rcmr_period = 0;
-	}
-	spin_unlock_irq(&ssc_p->lock);
-}
-
-
-
-/*
- * Set the SSC system clock rate
- */
-static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
-				   int clk_id, unsigned int freq, int dir)
-{
-	/* TODO: What the heck do I do here? */
-	return 0;
-}
-
-
-
-/*
- * Record DAI format for use by hw_params()
- */
-static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
-				unsigned int fmt)
-{
-	struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	ssc_p->daifmt = fmt;
-	return 0;
-}
-
-
-
-/*
- * Record SSC clock dividers for use in hw_params()
- */
-static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
-				   int div_id, int div)
-{
-	struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	switch (div_id) {
-	case AT32_SSC_CMR_DIV:
-		/*
-		 * The same master clock divider is used for both
-		 * transmit and receive, so if a value has already
-		 * been set, it must match this value
-		 */
-		if (ssc_p->cmr_div == 0)
-			ssc_p->cmr_div = div;
-		else if (div != ssc_p->cmr_div)
-			return -EBUSY;
-		break;
-
-	case AT32_SSC_TCMR_PERIOD:
-		ssc_p->tcmr_period = div;
-		break;
-
-	case AT32_SSC_RCMR_PERIOD:
-		ssc_p->rcmr_period = div;
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-
-
-/*
- * Configure the SSC
- */
-static int at32_ssc_hw_params(struct snd_pcm_substream *substream,
-			      struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int id = rtd->dai->cpu_dai->id;
-	struct at32_ssc_info *ssc_p = &ssc_info[id];
-	struct at32_pcm_dma_params *dma_params;
-	int channels, bits;
-	u32 tfmr, rfmr, tcmr, rcmr;
-	int start_event;
-	int ret;
-
-
-	/*
-	 * Currently, there is only one set of dma_params for each direction.
-	 * If more are added, this code will have to be changed to select
-	 * the proper set
-	 */
-	dma_params = &ssc_dma_params[id][substream->stream];
-	dma_params->ssc = ssc_p->ssc;
-	dma_params->substream = substream;
-
-	ssc_p->dma_params[substream->stream] = dma_params;
-
-
-	/*
-	 * The cpu_dai->dma_data field is only used to communicate the
-	 * appropriate DMA parameters to the PCM driver's hw_params()
-	 * function.  It should not be used for other purposes as it
-	 * is common to all substreams.
-	 */
-	rtd->dai->cpu_dai->dma_data = dma_params;
-
-	channels = params_channels(params);
-
-
-	/*
-	 * Determine sample size in bits and the PDC increment
-	 */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		bits = 8;
-		dma_params->pdc_xfer_size = 1;
-		break;
-
-	case SNDRV_PCM_FORMAT_S16:
-		bits = 16;
-		dma_params->pdc_xfer_size = 2;
-		break;
-
-	case SNDRV_PCM_FORMAT_S24:
-		bits = 24;
-		dma_params->pdc_xfer_size = 4;
-		break;
-
-	case SNDRV_PCM_FORMAT_S32:
-		bits = 32;
-		dma_params->pdc_xfer_size = 4;
-		break;
-
-	default:
-		pr_warning("at32-ssc: Unsupported PCM format %d",
-			   params_format(params));
-		return -EINVAL;
-	}
-	pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n",
-		 bits, dma_params->pdc_xfer_size, channels);
-
-
-	/*
-	 * The SSC only supports up to 16-bit samples in I2S format, due
-	 * to the size of the Frame Mode Register FSLEN field.
-	 */
-	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S)
-		if (bits > 16) {
-			pr_warning("at32-ssc: "
-				   "sample size %d is too large for I2S\n",
-				   bits);
-			return -EINVAL;
-		}
-
-
-	/*
-	 * Compute the SSC register settings
-	 */
-	switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK |
-				 SND_SOC_DAIFMT_MASTER_MASK)) {
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * I2S format, SSC provides BCLK and LRS clocks.
-		 *
-		 * The SSC transmit and receive clocks are generated from the
-		 * MCK divider, and the BCLK signal is output on the SSC TK line
-		 */
-		pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n");
-		rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
-			SSC_BF(RCMR_STTDLY, START_DELAY) |
-			SSC_BF(RCMR_START, SSC_START_FALLING_RF) |
-			SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(RCMR_CKS, SSC_CKS_DIV));
-
-		rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) |
-			SSC_BF(RFMR_FSLEN, bits - 1) |
-			SSC_BF(RFMR_DATNB, channels - 1) |
-			SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
-		tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
-			SSC_BF(TCMR_STTDLY, START_DELAY) |
-			SSC_BF(TCMR_START, SSC_START_FALLING_RF) |
-			SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
-			SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
-			SSC_BF(TCMR_CKS, SSC_CKS_DIV));
-
-		tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) |
-			SSC_BF(TFMR_FSLEN, bits - 1) |
-			SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) |
-			SSC_BF(TFMR_DATLEN, bits - 1));
-		break;
-
-
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
-		/*
-		 * I2S format, CODEC supplies BCLK and LRC clock.
-		 *
-		 * The SSC transmit clock is obtained from the BCLK signal
-		 * on the TK line, and the SSC receive clock is generated from
-		 * the transmit clock.
-		 *
-		 * For single channel data, one sample is transferred on the
-		 * falling edge of the LRC clock.  For two channel data, one
-		 * sample is transferred on both edges of the LRC clock.
-		 */
-		pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n");
-		start_event = ((channels == 1) ?
-			       SSC_START_FALLING_RF : SSC_START_EDGE_RF);
-
-		rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) |
-			SSC_BF(RCMR_START, start_event) |
-			SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(RCMR_CKS, SSC_CKS_CLOCK));
-
-		rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) |
-			SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
-		tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) |
-			SSC_BF(TCMR_START, start_event) |
-			SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
-			SSC_BF(TCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(TCMR_CKS, SSC_CKS_PIN));
-
-		tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) |
-			SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
-		break;
-
-
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
-		 *
-		 * The SSC transmit and receive clocks are generated from the
-		 * MCK divider, and the BCLK signal is output on the SSC TK line
-		 */
-		pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n");
-		rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
-			SSC_BF(RCMR_STTDLY, 1) |
-			SSC_BF(RCMR_START, SSC_START_RISING_RF) |
-			SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(RCMR_CKS, SSC_CKS_DIV));
-
-		rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) |
-			SSC_BF(RFMR_DATNB, channels - 1) |
-			SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
-		tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
-			SSC_BF(TCMR_STTDLY, 1) |
-			SSC_BF(TCMR_START, SSC_START_RISING_RF) |
-			SSC_BF(TCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
-			SSC_BF(TCMR_CKS, SSC_CKS_DIV));
-
-		tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) |
-			SSC_BF(TFMR_DATNB, channels - 1) |
-			SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
-		break;
-
-
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
-	default:
-		pr_warning("at32-ssc: unsupported DAI format 0x%x\n",
-			   ssc_p->daifmt);
-		return -EINVAL;
-		break;
-	}
-	pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
-		 rcmr, rfmr, tcmr, tfmr);
-
-
-	if (!ssc_p->initialized) {
-		/* enable peripheral clock */
-		pr_debug("at32-ssc: Starting clock\n");
-		clk_enable(ssc_p->ssc->clk);
-
-		/* Reset the SSC and its PDC registers */
-		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
-		ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
-
-		ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
-
-		ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0,
-				  ssc_p->name, ssc_p);
-		if (ret < 0) {
-			pr_warning("at32-ssc: request irq failed (%d)\n", ret);
-			pr_debug("at32-ssc: Stopping clock\n");
-			clk_disable(ssc_p->ssc->clk);
-			return ret;
-		}
-
-		ssc_p->initialized = 1;
-	}
-
-	/* Set SSC clock mode register */
-	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
-
-	/* set receive clock mode and format */
-	ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
-	ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
-
-	/* set transmit clock mode and format */
-	ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
-	ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
-
-	pr_debug("at32-ssc: SSC initialized\n");
-	return 0;
-}
-
-
-
-static int at32_ssc_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	struct at32_pcm_dma_params *dma_params;
-
-	dma_params = ssc_p->dma_params[substream->stream];
-
-	ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable);
-
-	return 0;
-}
-
-
-
-#ifdef CONFIG_PM
-static int at32_ssc_suspend(struct platform_device *pdev,
-			    struct snd_soc_dai *cpu_dai)
-{
-	struct at32_ssc_info *ssc_p;
-
-	if (!cpu_dai->active)
-		return 0;
-
-	ssc_p = &ssc_info[cpu_dai->id];
-
-	/* Save the status register before disabling transmit and receive */
-	ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR);
-	ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
-
-	/* Save the current interrupt mask, then disable unmasked interrupts */
-	ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
-	ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
-
-	ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
-	ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
-	ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
-	ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
-	ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
-
-	return 0;
-}
-
-
-
-static int at32_ssc_resume(struct platform_device *pdev,
-			   struct snd_soc_dai *cpu_dai)
-{
-	struct at32_ssc_info *ssc_p;
-	u32 cr;
-
-	if (!cpu_dai->active)
-		return 0;
-
-	ssc_p = &ssc_info[cpu_dai->id];
-
-	/* restore SSC register settings */
-	ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
-	ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
-	ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
-	ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
-	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
-
-	/* re-enable interrupts */
-	ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
-
-	/* Re-enable recieve and transmit as appropriate */
-	cr = 0;
-	cr |=
-	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
-	cr |=
-	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
-	ssc_writel(ssc_p->ssc->regs, CR, cr);
-
-	return 0;
-}
-#else /* CONFIG_PM */
-#  define at32_ssc_suspend	NULL
-#  define at32_ssc_resume	NULL
-#endif /* CONFIG_PM */
-
-
-#define AT32_SSC_RATES \
-    (SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-     SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-     SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-
-
-#define AT32_SSC_FORMATS \
-    (SNDRV_PCM_FMTBIT_S8  | SNDRV_PCM_FMTBIT_S16 | \
-     SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32)
-
-
-struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = {
-	{
-	 .name = "at32-ssc0",
-	 .id = 0,
-	 .type = SND_SOC_DAI_PCM,
-	 .suspend = at32_ssc_suspend,
-	 .resume = at32_ssc_resume,
-	 .playback = {
-		      .channels_min = 1,
-		      .channels_max = 2,
-		      .rates = AT32_SSC_RATES,
-		      .formats = AT32_SSC_FORMATS,
-		      },
-	 .capture = {
-		     .channels_min = 1,
-		     .channels_max = 2,
-		     .rates = AT32_SSC_RATES,
-		     .formats = AT32_SSC_FORMATS,
-		     },
-	 .ops = {
-		 .startup = at32_ssc_startup,
-		 .shutdown = at32_ssc_shutdown,
-		 .prepare = at32_ssc_prepare,
-		 .hw_params = at32_ssc_hw_params,
-		 },
-	 .dai_ops = {
-		     .set_sysclk = at32_ssc_set_dai_sysclk,
-		     .set_fmt = at32_ssc_set_dai_fmt,
-		     .set_clkdiv = at32_ssc_set_dai_clkdiv,
-		     },
-	 .private_data = &ssc_info[0],
-	 },
-	{
-	 .name = "at32-ssc1",
-	 .id = 1,
-	 .type = SND_SOC_DAI_PCM,
-	 .suspend = at32_ssc_suspend,
-	 .resume = at32_ssc_resume,
-	 .playback = {
-		      .channels_min = 1,
-		      .channels_max = 2,
-		      .rates = AT32_SSC_RATES,
-		      .formats = AT32_SSC_FORMATS,
-		      },
-	 .capture = {
-		     .channels_min = 1,
-		     .channels_max = 2,
-		     .rates = AT32_SSC_RATES,
-		     .formats = AT32_SSC_FORMATS,
-		     },
-	 .ops = {
-		 .startup = at32_ssc_startup,
-		 .shutdown = at32_ssc_shutdown,
-		 .prepare = at32_ssc_prepare,
-		 .hw_params = at32_ssc_hw_params,
-		 },
-	 .dai_ops = {
-		     .set_sysclk = at32_ssc_set_dai_sysclk,
-		     .set_fmt = at32_ssc_set_dai_fmt,
-		     .set_clkdiv = at32_ssc_set_dai_clkdiv,
-		     },
-	 .private_data = &ssc_info[1],
-	 },
-	{
-	 .name = "at32-ssc2",
-	 .id = 2,
-	 .type = SND_SOC_DAI_PCM,
-	 .suspend = at32_ssc_suspend,
-	 .resume = at32_ssc_resume,
-	 .playback = {
-		      .channels_min = 1,
-		      .channels_max = 2,
-		      .rates = AT32_SSC_RATES,
-		      .formats = AT32_SSC_FORMATS,
-		      },
-	 .capture = {
-		     .channels_min = 1,
-		     .channels_max = 2,
-		     .rates = AT32_SSC_RATES,
-		     .formats = AT32_SSC_FORMATS,
-		     },
-	 .ops = {
-		 .startup = at32_ssc_startup,
-		 .shutdown = at32_ssc_shutdown,
-		 .prepare = at32_ssc_prepare,
-		 .hw_params = at32_ssc_hw_params,
-		 },
-	 .dai_ops = {
-		     .set_sysclk = at32_ssc_set_dai_sysclk,
-		     .set_fmt = at32_ssc_set_dai_fmt,
-		     .set_clkdiv = at32_ssc_set_dai_clkdiv,
-		     },
-	 .private_data = &ssc_info[2],
-	 },
-};
-EXPORT_SYMBOL_GPL(at32_ssc_dai);
-
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("AT32 SSC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h
deleted file mode 100644
index 3c052dbbe460..000000000000
--- a/sound/soc/at32/at32-ssc.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/* sound/soc/at32/at32-ssc.h
- * ASoC SSC interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef __SOUND_SOC_AT32_AT32_SSC_H
-#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__
-
-#include <linux/types.h>
-#include <linux/atmel-ssc.h>
-
-#include "at32-pcm.h"
-
-
-
-struct at32_ssc_state {
-	u32 ssc_cmr;
-	u32 ssc_rcmr;
-	u32 ssc_rfmr;
-	u32 ssc_tcmr;
-	u32 ssc_tfmr;
-	u32 ssc_sr;
-	u32 ssc_imr;
-};
-
-
-
-struct at32_ssc_info {
-	char *name;
-	struct ssc_device *ssc;
-	spinlock_t lock;	/* lock for dir_mask */
-	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
-	unsigned short initialized;	/* true if SSC has been initialized */
-	unsigned short daifmt;
-	unsigned short cmr_div;
-	unsigned short tcmr_period;
-	unsigned short rcmr_period;
-	struct at32_pcm_dma_params *dma_params[2];
-	struct at32_ssc_state ssc_state;
-};
-
-
-/* SSC divider ids */
-#define AT32_SSC_CMR_DIV        0	/* MCK divider for BCLK */
-#define AT32_SSC_TCMR_PERIOD    1	/* BCLK divider for transmit FS */
-#define AT32_SSC_RCMR_PERIOD    2	/* BCLK divider for receive FS */
-
-
-extern struct snd_soc_dai at32_ssc_dai[];
-
-
-
-#endif /* __SOUND_SOC_AT32_AT32_SSC_H */
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
deleted file mode 100644
index 85a883299c2e..000000000000
--- a/sound/soc/at91/Kconfig
+++ /dev/null
@@ -1,10 +0,0 @@
-config SND_AT91_SOC
-	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
-	depends on ARCH_AT91
-	help
-	  Say Y or M if you want to add support for codecs attached to
-	  the AT91 SSC interface. You will also need
-	  to select the audio interfaces to support below.
-
-config SND_AT91_SOC_SSC
-	tristate
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
deleted file mode 100644
index b817f11df286..000000000000
--- a/sound/soc/at91/Makefile
+++ /dev/null
@@ -1,6 +0,0 @@
-# AT91 Platform Support
-snd-soc-at91-objs := at91-pcm.o
-snd-soc-at91-ssc-objs := at91-ssc.o
-
-obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
-obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
deleted file mode 100644
index 7ab48bd25e4c..000000000000
--- a/sound/soc/at91/at91-pcm.c
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
- *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Mar 3, 2006
- *
- * Based on pxa2xx-pcm.c by:
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	(C) 2004 MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-#include <mach/at91_ssc.h>
-
-#include "at91-pcm.h"
-
-#if 0
-#define	DBG(x...)	printk(KERN_INFO "at91-pcm: " x)
-#else
-#define	DBG(x...)
-#endif
-
-static const struct snd_pcm_hardware at91_pcm_hardware = {
-	.info			= SNDRV_PCM_INFO_MMAP |
-				  SNDRV_PCM_INFO_MMAP_VALID |
-				  SNDRV_PCM_INFO_INTERLEAVED |
-				  SNDRV_PCM_INFO_PAUSE,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
-	.period_bytes_min	= 32,
-	.period_bytes_max	= 8192,
-	.periods_min		= 2,
-	.periods_max		= 1024,
-	.buffer_bytes_max	= 32 * 1024,
-};
-
-struct at91_runtime_data {
-	struct at91_pcm_dma_params *params;
-	dma_addr_t dma_buffer;			/* physical address of dma buffer */
-	dma_addr_t dma_buffer_end;		/* first address beyond DMA buffer */
-	size_t period_size;
-	dma_addr_t period_ptr;			/* physical address of next period */
-	u32 pdc_xpr_save;			/* PDC register save */
-	u32 pdc_xcr_save;
-	u32 pdc_xnpr_save;
-	u32 pdc_xncr_save;
-};
-
-static void at91_pcm_dma_irq(u32 ssc_sr,
-	struct snd_pcm_substream *substream)
-{
-	struct at91_runtime_data *prtd = substream->runtime->private_data;
-	struct at91_pcm_dma_params *params = prtd->params;
-	static int count = 0;
-
-	count++;
-
-	if (ssc_sr & params->mask->ssc_endbuf) {
-
-		printk(KERN_WARNING
-			"at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
-			substream->stream == SNDRV_PCM_STREAM_PLAYBACK
-				? "underrun" : "overrun",
-			params->name, ssc_sr, count);
-
-		/* re-start the PDC */
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-
-		prtd->period_ptr += prtd->period_size;
-		if (prtd->period_ptr >= prtd->dma_buffer_end) {
-			prtd->period_ptr = prtd->dma_buffer;
-		}
-
-		at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
-		at91_ssc_write(params->ssc_base + params->pdc->xcr,
-				prtd->period_size / params->pdc_xfer_size);
-
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
-	}
-
-	if (ssc_sr & params->mask->ssc_endx) {
-
-		/* Load the PDC next pointer and counter registers */
-		prtd->period_ptr += prtd->period_size;
-		if (prtd->period_ptr >= prtd->dma_buffer_end) {
-			prtd->period_ptr = prtd->dma_buffer;
-		}
-		at91_ssc_write(params->ssc_base + params->pdc->xnpr,
-			       prtd->period_ptr);
-		at91_ssc_write(params->ssc_base + params->pdc->xncr,
-				prtd->period_size / params->pdc_xfer_size);
-	}
-
-	snd_pcm_period_elapsed(substream);
-}
-
-static int at91_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at91_runtime_data *prtd = runtime->private_data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-
-	/* this may get called several times by oss emulation
-	 * with different params */
-
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-	runtime->dma_bytes = params_buffer_bytes(params);
-
-	prtd->params = rtd->dai->cpu_dai->dma_data;
-	prtd->params->dma_intr_handler = at91_pcm_dma_irq;
-
-	prtd->dma_buffer = runtime->dma_addr;
-	prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
-	prtd->period_size = params_period_bytes(params);
-
-	DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n",
-		prtd->params->name, runtime->dma_bytes, prtd->period_size);
-	return 0;
-}
-
-static int at91_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	struct at91_runtime_data *prtd = substream->runtime->private_data;
-	struct at91_pcm_dma_params *params = prtd->params;
-
-	if (params != NULL) {
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-		prtd->params->dma_intr_handler = NULL;
-	}
-
-	return 0;
-}
-
-static int at91_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct at91_runtime_data *prtd = substream->runtime->private_data;
-	struct at91_pcm_dma_params *params = prtd->params;
-
-	at91_ssc_write(params->ssc_base + AT91_SSC_IDR,
-			params->mask->ssc_endx | params->mask->ssc_endbuf);
-
-	at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-	return 0;
-}
-
-static int at91_pcm_trigger(struct snd_pcm_substream *substream,
-	int cmd)
-{
-	struct at91_runtime_data *prtd = substream->runtime->private_data;
-	struct at91_pcm_dma_params *params = prtd->params;
-	int ret = 0;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		prtd->period_ptr = prtd->dma_buffer;
-
-		at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
-		at91_ssc_write(params->ssc_base + params->pdc->xcr,
-				prtd->period_size / params->pdc_xfer_size);
-
-		prtd->period_ptr += prtd->period_size;
-		at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
-		at91_ssc_write(params->ssc_base + params->pdc->xncr,
-				prtd->period_size / params->pdc_xfer_size);
-
-		DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n",
-			(unsigned long) prtd->period_ptr,
-			at91_ssc_read(params->ssc_base + params->pdc->xpr),
-			at91_ssc_read(params->ssc_base + params->pdc->xcr),
-			at91_ssc_read(params->ssc_base + params->pdc->xnpr),
-			at91_ssc_read(params->ssc_base + params->pdc->xncr));
-
-		at91_ssc_write(params->ssc_base + AT91_SSC_IER,
-			params->mask->ssc_endx | params->mask->ssc_endbuf);
-
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR,
-			params->mask->pdc_enable);
-
-		DBG("sr=%lx imr=%lx\n",
-		    at91_ssc_read(params->ssc_base + AT91_SSC_SR),
-		    at91_ssc_read(params->ssc_base + AT91_SSC_IMR));
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-		break;
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
-		break;
-
-	default:
-		ret = -EINVAL;
-	}
-
-	return ret;
-}
-
-static snd_pcm_uframes_t at91_pcm_pointer(
-	struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at91_runtime_data *prtd = runtime->private_data;
-	struct at91_pcm_dma_params *params = prtd->params;
-	dma_addr_t ptr;
-	snd_pcm_uframes_t x;
-
-	ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr);
-	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
-
-	if (x == runtime->buffer_size)
-		x = 0;
-	return x;
-}
-
-static int at91_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at91_runtime_data *prtd;
-	int ret = 0;
-
-	snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware);
-
-	/* ensure that buffer size is a multiple of period size */
-	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
-	if (ret < 0)
-		goto out;
-
-	prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL);
-	if (prtd == NULL) {
-		ret = -ENOMEM;
-		goto out;
-	}
-	runtime->private_data = prtd;
-
- out:
-	return ret;
-}
-
-static int at91_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct at91_runtime_data *prtd = substream->runtime->private_data;
-
-	kfree(prtd);
-	return 0;
-}
-
-static int at91_pcm_mmap(struct snd_pcm_substream *substream,
-	struct vm_area_struct *vma)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-				     runtime->dma_area,
-				     runtime->dma_addr,
-				     runtime->dma_bytes);
-}
-
-struct snd_pcm_ops at91_pcm_ops = {
-	.open		= at91_pcm_open,
-	.close		= at91_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= at91_pcm_hw_params,
-	.hw_free	= at91_pcm_hw_free,
-	.prepare	= at91_pcm_prepare,
-	.trigger	= at91_pcm_trigger,
-	.pointer	= at91_pcm_pointer,
-	.mmap		= at91_pcm_mmap,
-};
-
-static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
-	int stream)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	size_t size = at91_pcm_hardware.buffer_bytes_max;
-
-	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = pcm->card->dev;
-	buf->private_data = NULL;
-	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
-					   &buf->addr, GFP_KERNEL);
-
-	DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
-		(void *) buf->area,
-		(void *) buf->addr,
-		size);
-
-	if (!buf->area)
-		return -ENOMEM;
-
-	buf->bytes = size;
-	return 0;
-}
-
-static u64 at91_pcm_dmamask = 0xffffffff;
-
-static int at91_pcm_new(struct snd_card *card,
-	struct snd_soc_dai *dai, struct snd_pcm *pcm)
-{
-	int ret = 0;
-
-	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &at91_pcm_dmamask;
-	if (!card->dev->coherent_dma_mask)
-		card->dev->coherent_dma_mask = 0xffffffff;
-
-	if (dai->playback.channels_min) {
-		ret = at91_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto out;
-	}
-
-	if (dai->capture.channels_min) {
-		ret = at91_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto out;
-	}
- out:
-	return ret;
-}
-
-static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
-	struct snd_pcm_substream *substream;
-	struct snd_dma_buffer *buf;
-	int stream;
-
-	for (stream = 0; stream < 2; stream++) {
-		substream = pcm->streams[stream].substream;
-		if (!substream)
-			continue;
-
-		buf = &substream->dma_buffer;
-		if (!buf->area)
-			continue;
-
-		dma_free_writecombine(pcm->card->dev, buf->bytes,
-				      buf->area, buf->addr);
-		buf->area = NULL;
-	}
-}
-
-#ifdef CONFIG_PM
-static int at91_pcm_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
-{
-	struct snd_pcm_runtime *runtime = dai->runtime;
-	struct at91_runtime_data *prtd;
-	struct at91_pcm_dma_params *params;
-
-	if (!runtime)
-		return 0;
-
-	prtd = runtime->private_data;
-	params = prtd->params;
-
-	/* disable the PDC and save the PDC registers */
-
-	at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-
-	prtd->pdc_xpr_save  = at91_ssc_read(params->ssc_base + params->pdc->xpr);
-	prtd->pdc_xcr_save  = at91_ssc_read(params->ssc_base + params->pdc->xcr);
-	prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr);
-	prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr);
-
-	return 0;
-}
-
-static int at91_pcm_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
-{
-	struct snd_pcm_runtime *runtime = dai->runtime;
-	struct at91_runtime_data *prtd;
-	struct at91_pcm_dma_params *params;
-
-	if (!runtime)
-		return 0;
-
-	prtd = runtime->private_data;
-	params = prtd->params;
-
-	/* restore the PDC registers and enable the PDC */
-	at91_ssc_write(params->ssc_base + params->pdc->xpr,  prtd->pdc_xpr_save);
-	at91_ssc_write(params->ssc_base + params->pdc->xcr,  prtd->pdc_xcr_save);
-	at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save);
-	at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save);
-
-	at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
-	return 0;
-}
-#else
-#define at91_pcm_suspend	NULL
-#define at91_pcm_resume		NULL
-#endif
-
-struct snd_soc_platform at91_soc_platform = {
-	.name		= "at91-audio",
-	.pcm_ops 	= &at91_pcm_ops,
-	.pcm_new	= at91_pcm_new,
-	.pcm_free	= at91_pcm_free_dma_buffers,
-	.suspend	= at91_pcm_suspend,
-	.resume		= at91_pcm_resume,
-};
-
-EXPORT_SYMBOL_GPL(at91_soc_platform);
-
-MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
-MODULE_DESCRIPTION("Atmel AT91 PCM module");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
deleted file mode 100644
index e5aada2cb102..000000000000
--- a/sound/soc/at91/at91-pcm.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
- *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Mar 3, 2006
- *
- * Based on pxa2xx-pcm.h by:
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _AT91_PCM_H
-#define _AT91_PCM_H
-
-#include <mach/hardware.h>
-
-struct at91_ssc_periph {
-	void __iomem	*base;
-	u32		pid;
-};
-
-/*
- * Registers and status bits that are required by the PCM driver.
- */
-struct at91_pdc_regs {
-	unsigned int	xpr;		/* PDC recv/trans pointer */
-	unsigned int	xcr;		/* PDC recv/trans counter */
-	unsigned int	xnpr;		/* PDC next recv/trans pointer */
-	unsigned int	xncr;		/* PDC next recv/trans counter */
-	unsigned int	ptcr;		/* PDC transfer control */
-};
-
-struct at91_ssc_mask {
-	u32	ssc_enable;		/* SSC recv/trans enable */
-	u32	ssc_disable;		/* SSC recv/trans disable */
-	u32	ssc_endx;		/* SSC ENDTX or ENDRX */
-	u32	ssc_endbuf;		/* SSC TXBUFE or RXBUFF */
-	u32	pdc_enable;		/* PDC recv/trans enable */
-	u32	pdc_disable;		/* PDC recv/trans disable */
-};
-
-/*
- * This structure, shared between the PCM driver and the interface,
- * contains all information required by the PCM driver to perform the
- * PDC DMA operation.  All fields except dma_intr_handler() are initialized
- * by the interface.  The dms_intr_handler() pointer is set by the PCM
- * driver and called by the interface SSC interrupt handler if it is
- * non-NULL.
- */
-struct at91_pcm_dma_params {
-	char *name;			/* stream identifier */
-	int pdc_xfer_size;		/* PDC counter increment in bytes */
-	void __iomem *ssc_base;		/* SSC base address */
-	struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
-	struct at91_ssc_mask *mask;/* SSC & PDC status bits */
-	struct snd_pcm_substream *substream;
-	void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
-};
-
-extern struct snd_soc_platform at91_soc_platform;
-
-#define at91_ssc_read(a)	((unsigned long) __raw_readl(a))
-#define at91_ssc_write(a,v)	__raw_writel((v),(a))
-
-#endif /* _AT91_PCM_H */
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
deleted file mode 100644
index 1b61cc461261..000000000000
--- a/sound/soc/at91/at91-ssc.c
+++ /dev/null
@@ -1,791 +0,0 @@
-/*
- * at91-ssc.c  --  ALSA SoC AT91 SSC Audio Layer Platform driver
- *
- * Author: Frank Mandarino <fmandarino@endrelia.com>
- *         Endrelia Technologies Inc.
- *
- * Based on pxa2xx Platform drivers by
- * Liam Girdwood <lrg@slimlogic.co.uk>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
-#include <linux/clk.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-#include <mach/at91_pmc.h>
-#include <mach/at91_ssc.h>
-
-#include "at91-pcm.h"
-#include "at91-ssc.h"
-
-#if 0
-#define	DBG(x...)	printk(KERN_DEBUG "at91-ssc:" x)
-#else
-#define	DBG(x...)
-#endif
-
-#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
-#define NUM_SSC_DEVICES		1
-#else
-#define NUM_SSC_DEVICES		3
-#endif
-
-
-/*
- * SSC PDC registers required by the PCM DMA engine.
- */
-static struct at91_pdc_regs pdc_tx_reg = {
-	.xpr		= ATMEL_PDC_TPR,
-	.xcr		= ATMEL_PDC_TCR,
-	.xnpr		= ATMEL_PDC_TNPR,
-	.xncr		= ATMEL_PDC_TNCR,
-};
-
-static struct at91_pdc_regs pdc_rx_reg = {
-	.xpr		= ATMEL_PDC_RPR,
-	.xcr		= ATMEL_PDC_RCR,
-	.xnpr		= ATMEL_PDC_RNPR,
-	.xncr		= ATMEL_PDC_RNCR,
-};
-
-/*
- * SSC & PDC status bits for transmit and receive.
- */
-static struct at91_ssc_mask ssc_tx_mask = {
-	.ssc_enable	= AT91_SSC_TXEN,
-	.ssc_disable	= AT91_SSC_TXDIS,
-	.ssc_endx	= AT91_SSC_ENDTX,
-	.ssc_endbuf	= AT91_SSC_TXBUFE,
-	.pdc_enable	= ATMEL_PDC_TXTEN,
-	.pdc_disable	= ATMEL_PDC_TXTDIS,
-};
-
-static struct at91_ssc_mask ssc_rx_mask = {
-	.ssc_enable	= AT91_SSC_RXEN,
-	.ssc_disable	= AT91_SSC_RXDIS,
-	.ssc_endx	= AT91_SSC_ENDRX,
-	.ssc_endbuf	= AT91_SSC_RXBUFF,
-	.pdc_enable	= ATMEL_PDC_RXTEN,
-	.pdc_disable	= ATMEL_PDC_RXTDIS,
-};
-
-
-/*
- * DMA parameters.
- */
-static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
-	{{
-	.name		= "SSC0 PCM out",
-	.pdc		= &pdc_tx_reg,
-	.mask		= &ssc_tx_mask,
-	},
-	{
-	.name		= "SSC0 PCM in",
-	.pdc		= &pdc_rx_reg,
-	.mask		= &ssc_rx_mask,
-	}},
-#if NUM_SSC_DEVICES == 3
-	{{
-	.name		= "SSC1 PCM out",
-	.pdc		= &pdc_tx_reg,
-	.mask		= &ssc_tx_mask,
-	},
-	{
-	.name		= "SSC1 PCM in",
-	.pdc		= &pdc_rx_reg,
-	.mask		= &ssc_rx_mask,
-	}},
-	{{
-	.name		= "SSC2 PCM out",
-	.pdc		= &pdc_tx_reg,
-	.mask		= &ssc_tx_mask,
-	},
-	{
-	.name		= "SSC2 PCM in",
-	.pdc		= &pdc_rx_reg,
-	.mask		= &ssc_rx_mask,
-	}},
-#endif
-};
-
-struct at91_ssc_state {
-	u32	ssc_cmr;
-	u32	ssc_rcmr;
-	u32	ssc_rfmr;
-	u32	ssc_tcmr;
-	u32	ssc_tfmr;
-	u32	ssc_sr;
-	u32	ssc_imr;
-};
-
-static struct at91_ssc_info {
-	char		*name;
-	struct at91_ssc_periph ssc;
-	spinlock_t 	lock;		/* lock for dir_mask */
-	unsigned short	dir_mask;	/* 0=unused, 1=playback, 2=capture */
-	unsigned short	initialized;	/* 1=SSC has been initialized */
-	unsigned short	daifmt;
-	unsigned short	cmr_div;
-	unsigned short	tcmr_period;
-	unsigned short	rcmr_period;
-	struct at91_pcm_dma_params *dma_params[2];
-	struct at91_ssc_state ssc_state;
-
-} ssc_info[NUM_SSC_DEVICES] = {
-	{
-	.name		= "ssc0",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
-	.dir_mask	= 0,
-	.initialized	= 0,
-	},
-#if NUM_SSC_DEVICES == 3
-	{
-	.name		= "ssc1",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
-	.dir_mask	= 0,
-	.initialized	= 0,
-	},
-	{
-	.name		= "ssc2",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
-	.dir_mask	= 0,
-	.initialized	= 0,
-	},
-#endif
-};
-
-static unsigned int at91_ssc_sysclk;
-
-/*
- * SSC interrupt handler.  Passes PDC interrupts to the DMA
- * interrupt handler in the PCM driver.
- */
-static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id)
-{
-	struct at91_ssc_info *ssc_p = dev_id;
-	struct at91_pcm_dma_params *dma_params;
-	u32 ssc_sr;
-	int i;
-
-	ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)
-			& at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
-
-	/*
-	 * Loop through the substreams attached to this SSC.  If
-	 * a DMA-related interrupt occurred on that substream, call
-	 * the DMA interrupt handler function, if one has been
-	 * registered in the dma_params structure by the PCM driver.
-	 */
-	for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
-		dma_params = ssc_p->dma_params[i];
-
-		if (dma_params != NULL && dma_params->dma_intr_handler != NULL &&
-			(ssc_sr &
-			(dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf)))
-
-			dma_params->dma_intr_handler(ssc_sr, dma_params->substream);
-	}
-
-	return IRQ_HANDLED;
-}
-
-/*
- * Startup.  Only that one substream allowed in each direction.
- */
-static int at91_ssc_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	int dir_mask;
-
-	DBG("ssc_startup: SSC_SR=0x%08lx\n",
-			at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
-	dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2;
-
-	spin_lock_irq(&ssc_p->lock);
-	if (ssc_p->dir_mask & dir_mask) {
-		spin_unlock_irq(&ssc_p->lock);
-		return -EBUSY;
-	}
-	ssc_p->dir_mask |= dir_mask;
-	spin_unlock_irq(&ssc_p->lock);
-
-	return 0;
-}
-
-/*
- * Shutdown.  Clear DMA parameters and shutdown the SSC if there
- * are no other substreams open.
- */
-static void at91_ssc_shutdown(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	struct at91_pcm_dma_params *dma_params;
-	int dir, dir_mask;
-
-	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
-	dma_params = ssc_p->dma_params[dir];
-
-	if (dma_params != NULL) {
-		at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
-				dma_params->mask->ssc_disable);
-		DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"),
-			at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
-
-		dma_params->ssc_base = NULL;
-		dma_params->substream = NULL;
-		ssc_p->dma_params[dir] = NULL;
-	}
-
-	dir_mask = 1 << dir;
-
-	spin_lock_irq(&ssc_p->lock);
-	ssc_p->dir_mask &= ~dir_mask;
-	if (!ssc_p->dir_mask) {
-		/* Shutdown the SSC clock. */
-		DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
-		at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
-
-		if (ssc_p->initialized) {
-			free_irq(ssc_p->ssc.pid, ssc_p);
-			ssc_p->initialized = 0;
-		}
-
-		/* Reset the SSC */
-		at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
-
-		/* Clear the SSC dividers */
-		ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
-	}
-	spin_unlock_irq(&ssc_p->lock);
-}
-
-/*
- * Record the SSC system clock rate.
- */
-static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
-		int clk_id, unsigned int freq, int dir)
-{
-	/*
-	 * The only clock supplied to the SSC is the AT91 master clock,
-	 * which is only used if the SSC is generating BCLK and/or
-	 * LRC clocks.
-	 */
-	switch (clk_id) {
-	case AT91_SYSCLK_MCK:
-		at91_ssc_sysclk = freq;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-/*
- * Record the DAI format for use in hw_params().
- */
-static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
-		unsigned int fmt)
-{
-	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	ssc_p->daifmt = fmt;
-	return 0;
-}
-
-/*
- * Record SSC clock dividers for use in hw_params().
- */
-static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
-	int div_id, int div)
-{
-	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	switch (div_id) {
-	case AT91SSC_CMR_DIV:
-		/*
-		 * The same master clock divider is used for both
-		 * transmit and receive, so if a value has already
-		 * been set, it must match this value.
-		 */
-		if (ssc_p->cmr_div == 0)
-			ssc_p->cmr_div = div;
-		else
-			if (div != ssc_p->cmr_div)
-				return -EBUSY;
-		break;
-
-	case AT91SSC_TCMR_PERIOD:
-		ssc_p->tcmr_period = div;
-		break;
-
-	case AT91SSC_RCMR_PERIOD:
-		ssc_p->rcmr_period = div;
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-/*
- * Configure the SSC.
- */
-static int at91_ssc_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int id = rtd->dai->cpu_dai->id;
-	struct at91_ssc_info *ssc_p = &ssc_info[id];
-	struct at91_pcm_dma_params *dma_params;
-	int dir, channels, bits;
-	u32 tfmr, rfmr, tcmr, rcmr;
-	int start_event;
-	int ret;
-
-	/*
-	 * Currently, there is only one set of dma params for
-	 * each direction.  If more are added, this code will
-	 * have to be changed to select the proper set.
-	 */
-	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
-
-	dma_params = &ssc_dma_params[id][dir];
-	dma_params->ssc_base = ssc_p->ssc.base;
-	dma_params->substream = substream;
-
-	ssc_p->dma_params[dir] = dma_params;
-
-	/*
-	 * The cpu_dai->dma_data field is only used to communicate the
-	 * appropriate DMA parameters to the pcm driver hw_params()
-	 * function.  It should not be used for other purposes
-	 * as it is common to all substreams.
-	 */
-	rtd->dai->cpu_dai->dma_data = dma_params;
-
-	channels = params_channels(params);
-
-	/*
-	 * Determine sample size in bits and the PDC increment.
-	 */
-	switch(params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		bits = 8;
-		dma_params->pdc_xfer_size = 1;
-		break;
-	case SNDRV_PCM_FORMAT_S16_LE:
-		bits = 16;
-		dma_params->pdc_xfer_size = 2;
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		bits = 24;
-		dma_params->pdc_xfer_size = 4;
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		bits = 32;
-		dma_params->pdc_xfer_size = 4;
-		break;
-	default:
-		printk(KERN_WARNING "at91-ssc: unsupported PCM format\n");
-		return -EINVAL;
-	}
-
-	/*
-	 * The SSC only supports up to 16-bit samples in I2S format, due
-	 * to the size of the Frame Mode Register FSLEN field.
-	 */
-	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
-		&& bits > 16) {
-		printk(KERN_WARNING
-			"at91-ssc: sample size %d is too large for I2S\n", bits);
-		return -EINVAL;
-	}
-
-	/*
-	 * Compute SSC register settings.
-	 */
-	switch (ssc_p->daifmt
-		& (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
-
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * I2S format, SSC provides BCLK and LRC clocks.
-		 *
-		 * The SSC transmit and receive clocks are generated from the
-		 * MCK divider, and the BCLK signal is output on the SSC TK line.
-		 */
-		rcmr =	  (( ssc_p->rcmr_period		<< 24) & AT91_SSC_PERIOD)
-			| (( 1				<< 16) & AT91_SSC_STTDLY)
-			| (( AT91_SSC_START_FALLING_RF	     ) & AT91_SSC_START)
-			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
-			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
-			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
-
-		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
-			| (( AT91_SSC_FSOS_NEGATIVE	     ) & AT91_SSC_FSOS)
-			| (((bits - 1)			<< 16) & AT91_SSC_FSLEN)
-			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
-			| (( 1				<<  7) & AT91_SSC_MSBF)
-			| (( 0				<<  5) & AT91_SSC_LOOP)
-			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
-
-		tcmr =	  (( ssc_p->tcmr_period		<< 24) & AT91_SSC_PERIOD)
-			| (( 1				<< 16) & AT91_SSC_STTDLY)
-			| (( AT91_SSC_START_FALLING_RF       ) & AT91_SSC_START)
-			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
-			| (( AT91_SSC_CKO_CONTINUOUS	     ) & AT91_SSC_CKO)
-			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
-
-		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
-			| (( 0				<< 23) & AT91_SSC_FSDEN)
-			| (( AT91_SSC_FSOS_NEGATIVE	     ) & AT91_SSC_FSOS)
-			| (((bits - 1)			<< 16) & AT91_SSC_FSLEN)
-			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
-			| (( 1				<<  7) & AT91_SSC_MSBF)
-			| (( 0				<<  5) & AT91_SSC_DATDEF)
-			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
-		break;
-
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
-		/*
-		 * I2S format, CODEC supplies BCLK and LRC clocks.
-		 *
-		 * The SSC transmit clock is obtained from the BCLK signal on
-		 * on the TK line, and the SSC receive clock is generated from the
-		 * transmit clock.
-		 *
-		 * For single channel data, one sample is transferred on the falling
-		 * edge of the LRC clock.  For two channel data, one sample is
-		 * transferred on both edges of the LRC clock.
-		 */
-		start_event = channels == 1
-				? AT91_SSC_START_FALLING_RF
-				: AT91_SSC_START_EDGE_RF;
-
-		rcmr =	  (( 0				<< 24) & AT91_SSC_PERIOD)
-			| (( 1				<< 16) & AT91_SSC_STTDLY)
-			| (( start_event		     ) & AT91_SSC_START)
-			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
-			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
-			| (( AT91_SSC_CKS_CLOCK		     ) & AT91_SSC_CKS);
-
-		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
-			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
-			| (( 0				<< 16) & AT91_SSC_FSLEN)
-			| (( 0				<<  8) & AT91_SSC_DATNB)
-			| (( 1				<<  7) & AT91_SSC_MSBF)
-			| (( 0				<<  5) & AT91_SSC_LOOP)
-			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
-
-		tcmr =	  (( 0				<< 24) & AT91_SSC_PERIOD)
-			| (( 1				<< 16) & AT91_SSC_STTDLY)
-			| (( start_event		     ) & AT91_SSC_START)
-			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
-			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
-			| (( AT91_SSC_CKS_PIN		     ) & AT91_SSC_CKS);
-
-		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
-			| (( 0				<< 23) & AT91_SSC_FSDEN)
-			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
-			| (( 0				<< 16) & AT91_SSC_FSLEN)
-			| (( 0				<<  8) & AT91_SSC_DATNB)
-			| (( 1				<<  7) & AT91_SSC_MSBF)
-			| (( 0				<<  5) & AT91_SSC_DATDEF)
-			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
-		break;
-
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
-		 *
-		 * The SSC transmit and receive clocks are generated from the
-		 * MCK divider, and the BCLK signal is output on the SSC TK line.
-		 */
-		rcmr =	  (( ssc_p->rcmr_period		<< 24) & AT91_SSC_PERIOD)
-			| (( 1				<< 16) & AT91_SSC_STTDLY)
-			| (( AT91_SSC_START_RISING_RF	     ) & AT91_SSC_START)
-			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
-			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
-			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
-
-		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
-			| (( AT91_SSC_FSOS_POSITIVE	     ) & AT91_SSC_FSOS)
-			| (( 0				<< 16) & AT91_SSC_FSLEN)
-			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
-			| (( 1				<<  7) & AT91_SSC_MSBF)
-			| (( 0				<<  5) & AT91_SSC_LOOP)
-			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
-
-		tcmr =	  (( ssc_p->tcmr_period		<< 24) & AT91_SSC_PERIOD)
-			| (( 1				<< 16) & AT91_SSC_STTDLY)
-			| (( AT91_SSC_START_RISING_RF        ) & AT91_SSC_START)
-			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
-			| (( AT91_SSC_CKO_CONTINUOUS	     ) & AT91_SSC_CKO)
-			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
-
-		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
-			| (( 0				<< 23) & AT91_SSC_FSDEN)
-			| (( AT91_SSC_FSOS_POSITIVE	     ) & AT91_SSC_FSOS)
-			| (( 0				<< 16) & AT91_SSC_FSLEN)
-			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
-			| (( 1				<<  7) & AT91_SSC_MSBF)
-			| (( 0				<<  5) & AT91_SSC_DATDEF)
-			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
-
-
-
-			break;
-
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
-	default:
-		printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n",
-			ssc_p->daifmt);
-		return -EINVAL;
-		break;
-	}
-	DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr);
-
-	if (!ssc_p->initialized) {
-
-		/* Enable PMC peripheral clock for this SSC */
-		DBG("Starting pid %d clock\n", ssc_p->ssc.pid);
-		at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
-
-		/* Reset the SSC and its PDC registers */
-		at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
-
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0);
-
-		if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt,
-					0, ssc_p->name, ssc_p)) < 0) {
-			printk(KERN_WARNING "at91-ssc: request_irq failure\n");
-
-			DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
-			at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
-			return ret;
-		}
-
-		ssc_p->initialized = 1;
-	}
-
-	/* set SSC clock mode register */
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div);
-
-	/* set receive clock mode and format */
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr);
-
-	/* set transmit clock mode and format */
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr);
-
-	DBG("hw_params: SSC initialized\n");
-	return 0;
-}
-
-
-static int at91_ssc_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	struct at91_pcm_dma_params *dma_params;
-	int dir;
-
-	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
-	dma_params = ssc_p->dma_params[dir];
-
-	at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
-			dma_params->mask->ssc_enable);
-
-	DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit",
-		at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR));
-	return 0;
-}
-
-
-#ifdef CONFIG_PM
-static int at91_ssc_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *cpu_dai)
-{
-	struct at91_ssc_info *ssc_p;
-
-	if(!cpu_dai->active)
-		return 0;
-
-	ssc_p = &ssc_info[cpu_dai->id];
-
-	/* Save the status register before disabling transmit and receive. */
-	ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
-			AT91_SSC_TXDIS | AT91_SSC_RXDIS);
-
-	/* Save the current interrupt mask, then disable unmasked interrupts. */
-	ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr);
-
-	ssc_p->ssc_state.ssc_cmr  = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR);
-	ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR);
-	ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR);
-	ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR);
-	ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR);
-
-	return 0;
-}
-
-static int at91_ssc_resume(struct platform_device *pdev,
-	struct snd_soc_dai *cpu_dai)
-{
-	struct at91_ssc_info *ssc_p;
-
-	if(!cpu_dai->active)
-		return 0;
-
-	ssc_p = &ssc_info[cpu_dai->id];
-
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr);
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR,  ssc_p->ssc_state.ssc_cmr);
-
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER,  ssc_p->ssc_state.ssc_imr);
-
-	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
-		((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) |
-		((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0));
-
-	return 0;
-}
-
-#else
-#define at91_ssc_suspend	NULL
-#define at91_ssc_resume		NULL
-#endif
-
-#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
-			SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-			SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-			SNDRV_PCM_RATE_96000)
-
-#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
-			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-
-struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = {
-	{	.name = "at91-ssc0",
-		.id = 0,
-		.type = SND_SOC_DAI_PCM,
-		.suspend = at91_ssc_suspend,
-		.resume = at91_ssc_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.ops = {
-			.startup = at91_ssc_startup,
-			.shutdown = at91_ssc_shutdown,
-			.prepare = at91_ssc_prepare,
-			.hw_params = at91_ssc_hw_params,},
-		.dai_ops = {
-			.set_sysclk = at91_ssc_set_dai_sysclk,
-			.set_fmt = at91_ssc_set_dai_fmt,
-			.set_clkdiv = at91_ssc_set_dai_clkdiv,},
-		.private_data = &ssc_info[0].ssc,
-	},
-#if NUM_SSC_DEVICES == 3
-	{	.name = "at91-ssc1",
-		.id = 1,
-		.type = SND_SOC_DAI_PCM,
-		.suspend = at91_ssc_suspend,
-		.resume = at91_ssc_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.ops = {
-			.startup = at91_ssc_startup,
-			.shutdown = at91_ssc_shutdown,
-			.prepare = at91_ssc_prepare,
-			.hw_params = at91_ssc_hw_params,},
-		.dai_ops = {
-			.set_sysclk = at91_ssc_set_dai_sysclk,
-			.set_fmt = at91_ssc_set_dai_fmt,
-			.set_clkdiv = at91_ssc_set_dai_clkdiv,},
-		.private_data = &ssc_info[1].ssc,
-	},
-	{	.name = "at91-ssc2",
-		.id = 2,
-		.type = SND_SOC_DAI_PCM,
-		.suspend = at91_ssc_suspend,
-		.resume = at91_ssc_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.ops = {
-			.startup = at91_ssc_startup,
-			.shutdown = at91_ssc_shutdown,
-			.prepare = at91_ssc_prepare,
-			.hw_params = at91_ssc_hw_params,},
-		.dai_ops = {
-			.set_sysclk = at91_ssc_set_dai_sysclk,
-			.set_fmt = at91_ssc_set_dai_fmt,
-			.set_clkdiv = at91_ssc_set_dai_clkdiv,},
-		.private_data = &ssc_info[2].ssc,
-	},
-#endif
-};
-
-EXPORT_SYMBOL_GPL(at91_ssc_dai);
-
-/* Module information */
-MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com");
-MODULE_DESCRIPTION("AT91 SSC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h
deleted file mode 100644
index 6b7bf382d06f..000000000000
--- a/sound/soc/at91/at91-ssc.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC
- *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Jan 9, 2007
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _AT91_SSC_H
-#define _AT91_SSC_H
-
-/* SSC system clock ids */
-#define AT91_SYSCLK_MCK		0 /* SSC uses AT91 MCK as system clock */
-
-/* SSC divider ids */
-#define AT91SSC_CMR_DIV		0 /* MCK divider for BCLK */
-#define AT91SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
-#define AT91SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
-
-extern struct snd_soc_dai at91_ssc_dai[];
-
-#endif /* _AT91_SSC_H */
-
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
new file mode 100644
index 000000000000..a608d7009dbd
--- /dev/null
+++ b/sound/soc/atmel/Kconfig
@@ -0,0 +1,43 @@
+config SND_ATMEL_SOC
+	tristate "SoC Audio for the Atmel System-on-Chip"
+	depends on ARCH_AT91 || AVR32
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the ATMEL SSC interface. You will also need
+	  to select the audio interfaces to support below.
+
+config SND_ATMEL_SOC_SSC
+	tristate
+	depends on SND_ATMEL_SOC
+	help
+	  Say Y or M if you want to add support for codecs the
+	  ATMEL SSC interface. You will also needs to select the individual
+	  machine drivers to support below.
+
+config SND_AT91_SOC_SAM9G20_WM8731
+	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
+	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on WM8731-based
+	  AT91sam9g20 evaluation board.
+
+config SND_AT32_SOC_PLAYPAQ
+        tristate "SoC Audio support for PlayPaq with WM8510"
+        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+        select SND_ATMEL_SOC_SSC
+        select SND_SOC_WM8510
+        help
+          Say Y or M here if you want to add support for SoC audio
+          on the LRS PlayPaq.
+
+config SND_AT32_SOC_PLAYPAQ_SLAVE
+        bool "Run CODEC on PlayPaq in slave mode"
+        depends on SND_AT32_SOC_PLAYPAQ
+        default n
+        help
+          Say Y if you want to run with the AT32 SSC generating the BCLK
+          and FRAME signals on the PlayPaq.  Unless you want to play
+          with the AT32 as the SSC master, you probably want to say N here,
+          as this will give you better sound quality.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
new file mode 100644
index 000000000000..f54a7cc68e66
--- /dev/null
+++ b/sound/soc/atmel/Makefile
@@ -0,0 +1,15 @@
+# AT91 Platform Support
+snd-soc-atmel-pcm-objs := atmel-pcm.o
+snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
+
+obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o
+obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
+
+# AT91 Machine Support
+snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+
+# AT32 Machine Support
+snd-soc-playpaq-objs := playpaq_wm8510.o
+
+obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
new file mode 100644
index 000000000000..1fac5efd285b
--- /dev/null
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -0,0 +1,494 @@
+/*
+ * atmel-pcm.c  --  ALSA PCM interface for the Atmel atmel SoC.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	(C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/atmel_pdc.h>
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+
+#include "atmel-pcm.h"
+
+
+/*--------------------------------------------------------------------------*\
+ * Hardware definition
+\*--------------------------------------------------------------------------*/
+/* TODO: These values were taken from the AT91 platform driver, check
+ *	 them against real values for AT32
+ */
+static const struct snd_pcm_hardware atmel_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_PAUSE,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 2,
+	.periods_max		= 1024,
+	.buffer_bytes_max	= 32 * 1024,
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * Data types
+\*--------------------------------------------------------------------------*/
+struct atmel_runtime_data {
+	struct atmel_pcm_dma_params *params;
+	dma_addr_t dma_buffer;		/* physical address of dma buffer */
+	dma_addr_t dma_buffer_end;	/* first address beyond DMA buffer */
+	size_t period_size;
+
+	dma_addr_t period_ptr;		/* physical address of next period */
+	int periods;			/* period index of period_ptr */
+
+	/* PDC register save */
+	u32 pdc_xpr_save;
+	u32 pdc_xcr_save;
+	u32 pdc_xnpr_save;
+	u32 pdc_xncr_save;
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * Helper functions
+\*--------------------------------------------------------------------------*/
+static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+	int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = atmel_pcm_hardware.buffer_bytes_max;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->area = dma_alloc_coherent(pcm->card->dev, size,
+					  &buf->addr, GFP_KERNEL);
+	pr_debug("atmel-pcm:"
+		"preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+		(void *) buf->area,
+		(void *) buf->addr,
+		size);
+
+	if (!buf->area)
+		return -ENOMEM;
+
+	buf->bytes = size;
+	return 0;
+}
+/*--------------------------------------------------------------------------*\
+ * ISR
+\*--------------------------------------------------------------------------*/
+static void atmel_pcm_dma_irq(u32 ssc_sr,
+	struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+	static int count;
+
+	count++;
+
+	if (ssc_sr & params->mask->ssc_endbuf) {
+		pr_warning("atmel-pcm: buffer %s on %s"
+				" (SSC_SR=%#x, count=%d)\n",
+				substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+				? "underrun" : "overrun",
+				params->name, ssc_sr, count);
+
+		/* re-start the PDC */
+		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+			   params->mask->pdc_disable);
+		prtd->period_ptr += prtd->period_size;
+		if (prtd->period_ptr >= prtd->dma_buffer_end)
+			prtd->period_ptr = prtd->dma_buffer;
+
+		ssc_writex(params->ssc->regs, params->pdc->xpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xcr,
+			   prtd->period_size / params->pdc_xfer_size);
+		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+			   params->mask->pdc_enable);
+	}
+
+	if (ssc_sr & params->mask->ssc_endx) {
+		/* Load the PDC next pointer and counter registers */
+		prtd->period_ptr += prtd->period_size;
+		if (prtd->period_ptr >= prtd->dma_buffer_end)
+			prtd->period_ptr = prtd->dma_buffer;
+
+		ssc_writex(params->ssc->regs, params->pdc->xnpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xncr,
+			   prtd->period_size / params->pdc_xfer_size);
+	}
+
+	snd_pcm_period_elapsed(substream);
+}
+
+
+/*--------------------------------------------------------------------------*\
+ * PCM operations
+\*--------------------------------------------------------------------------*/
+static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct atmel_runtime_data *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* this may get called several times by oss emulation
+	 * with different params */
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = params_buffer_bytes(params);
+
+	prtd->params = rtd->dai->cpu_dai->dma_data;
+	prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
+
+	prtd->dma_buffer = runtime->dma_addr;
+	prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
+	prtd->period_size = params_period_bytes(params);
+
+	pr_debug("atmel-pcm: "
+		"hw_params: DMA for %s initialized "
+		"(dma_bytes=%u, period_size=%u)\n",
+		prtd->params->name,
+		runtime->dma_bytes,
+		prtd->period_size);
+	return 0;
+}
+
+static int atmel_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+
+	if (params != NULL) {
+		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+			   params->mask->pdc_disable);
+		prtd->params->dma_intr_handler = NULL;
+	}
+
+	return 0;
+}
+
+static int atmel_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+
+	ssc_writex(params->ssc->regs, SSC_IDR,
+		   params->mask->ssc_endx | params->mask->ssc_endbuf);
+	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+		   params->mask->pdc_disable);
+	return 0;
+}
+
+static int atmel_pcm_trigger(struct snd_pcm_substream *substream,
+	int cmd)
+{
+	struct snd_pcm_runtime *rtd = substream->runtime;
+	struct atmel_runtime_data *prtd = rtd->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+	int ret = 0;
+
+	pr_debug("atmel-pcm:buffer_size = %ld,"
+		"dma_area = %p, dma_bytes = %u\n",
+		rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		prtd->period_ptr = prtd->dma_buffer;
+
+		ssc_writex(params->ssc->regs, params->pdc->xpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xcr,
+			   prtd->period_size / params->pdc_xfer_size);
+
+		prtd->period_ptr += prtd->period_size;
+		ssc_writex(params->ssc->regs, params->pdc->xnpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xncr,
+			   prtd->period_size / params->pdc_xfer_size);
+
+		pr_debug("atmel-pcm: trigger: "
+			"period_ptr=%lx, xpr=%u, "
+			"xcr=%u, xnpr=%u, xncr=%u\n",
+			(unsigned long)prtd->period_ptr,
+			ssc_readx(params->ssc->regs, params->pdc->xpr),
+			ssc_readx(params->ssc->regs, params->pdc->xcr),
+			ssc_readx(params->ssc->regs, params->pdc->xnpr),
+			ssc_readx(params->ssc->regs, params->pdc->xncr));
+
+		ssc_writex(params->ssc->regs, SSC_IER,
+			   params->mask->ssc_endx | params->mask->ssc_endbuf);
+		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+			   params->mask->pdc_enable);
+
+		pr_debug("sr=%u imr=%u\n",
+			ssc_readx(params->ssc->regs, SSC_SR),
+			ssc_readx(params->ssc->regs, SSC_IER));
+		break;		/* SNDRV_PCM_TRIGGER_START */
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+			   params->mask->pdc_disable);
+		break;
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+			   params->mask->pdc_enable);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static snd_pcm_uframes_t atmel_pcm_pointer(
+	struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct atmel_runtime_data *prtd = runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+	dma_addr_t ptr;
+	snd_pcm_uframes_t x;
+
+	ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
+	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+	if (x == runtime->buffer_size)
+		x = 0;
+
+	return x;
+}
+
+static int atmel_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct atmel_runtime_data *prtd;
+	int ret = 0;
+
+	snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware);
+
+	/* ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+						SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto out;
+
+	prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL);
+	if (prtd == NULL) {
+		ret = -ENOMEM;
+		goto out;
+	}
+	runtime->private_data = prtd;
+
+ out:
+	return ret;
+}
+
+static int atmel_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+
+	kfree(prtd);
+	return 0;
+}
+
+static int atmel_pcm_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	return remap_pfn_range(vma, vma->vm_start,
+		       substream->dma_buffer.addr >> PAGE_SHIFT,
+		       vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops atmel_pcm_ops = {
+	.open		= atmel_pcm_open,
+	.close		= atmel_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= atmel_pcm_hw_params,
+	.hw_free	= atmel_pcm_hw_free,
+	.prepare	= atmel_pcm_prepare,
+	.trigger	= atmel_pcm_trigger,
+	.pointer	= atmel_pcm_pointer,
+	.mmap		= atmel_pcm_mmap,
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * ASoC platform driver
+\*--------------------------------------------------------------------------*/
+static u64 atmel_pcm_dmamask = 0xffffffff;
+
+static int atmel_pcm_new(struct snd_card *card,
+	struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &atmel_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	if (dai->playback.channels_min) {
+		ret = atmel_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (dai->capture.channels_min) {
+		pr_debug("at32-pcm:"
+				"Allocating PCM capture DMA buffer\n");
+		ret = atmel_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+
+static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+		dma_free_coherent(pcm->card->dev, buf->bytes,
+				  buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+#ifdef CONFIG_PM
+static int atmel_pcm_suspend(struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct atmel_runtime_data *prtd;
+	struct atmel_pcm_dma_params *params;
+
+	if (!runtime)
+		return 0;
+
+	prtd = runtime->private_data;
+	params = prtd->params;
+
+	/* disable the PDC and save the PDC registers */
+
+	ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable);
+
+	prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
+	prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
+	prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
+	prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
+
+	return 0;
+}
+
+static int atmel_pcm_resume(struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct atmel_runtime_data *prtd;
+	struct atmel_pcm_dma_params *params;
+
+	if (!runtime)
+		return 0;
+
+	prtd = runtime->private_data;
+	params = prtd->params;
+
+	/* restore the PDC registers and enable the PDC */
+	ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
+	ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
+	ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
+	ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
+
+	ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable);
+	return 0;
+}
+#else
+#define atmel_pcm_suspend	NULL
+#define atmel_pcm_resume	NULL
+#endif
+
+struct snd_soc_platform atmel_soc_platform = {
+	.name		= "atmel-audio",
+	.pcm_ops 	= &atmel_pcm_ops,
+	.pcm_new	= atmel_pcm_new,
+	.pcm_free	= atmel_pcm_free_dma_buffers,
+	.suspend	= atmel_pcm_suspend,
+	.resume		= atmel_pcm_resume,
+};
+EXPORT_SYMBOL_GPL(atmel_soc_platform);
+
+static int __init atmel_pcm_modinit(void)
+{
+	return snd_soc_register_platform(&atmel_soc_platform);
+}
+module_init(atmel_pcm_modinit);
+
+static void __exit atmel_pcm_modexit(void)
+{
+	snd_soc_unregister_platform(&atmel_soc_platform);
+}
+module_exit(atmel_pcm_modexit);
+
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("Atmel PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
new file mode 100644
index 000000000000..ec9b2824b663
--- /dev/null
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -0,0 +1,86 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	(C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#ifndef _ATMEL_PCM_H
+#define _ATMEL_PCM_H
+
+#include <linux/atmel-ssc.h>
+
+/*
+ * Registers and status bits that are required by the PCM driver.
+ */
+struct atmel_pdc_regs {
+	unsigned int	xpr;		/* PDC recv/trans pointer */
+	unsigned int	xcr;		/* PDC recv/trans counter */
+	unsigned int	xnpr;		/* PDC next recv/trans pointer */
+	unsigned int	xncr;		/* PDC next recv/trans counter */
+	unsigned int	ptcr;		/* PDC transfer control */
+};
+
+struct atmel_ssc_mask {
+	u32	ssc_enable;		/* SSC recv/trans enable */
+	u32	ssc_disable;		/* SSC recv/trans disable */
+	u32	ssc_endx;		/* SSC ENDTX or ENDRX */
+	u32	ssc_endbuf;		/* SSC TXBUFE or RXBUFF */
+	u32	pdc_enable;		/* PDC recv/trans enable */
+	u32	pdc_disable;		/* PDC recv/trans disable */
+};
+
+/*
+ * This structure, shared between the PCM driver and the interface,
+ * contains all information required by the PCM driver to perform the
+ * PDC DMA operation.  All fields except dma_intr_handler() are initialized
+ * by the interface.  The dms_intr_handler() pointer is set by the PCM
+ * driver and called by the interface SSC interrupt handler if it is
+ * non-NULL.
+ */
+struct atmel_pcm_dma_params {
+	char *name;			/* stream identifier */
+	int pdc_xfer_size;		/* PDC counter increment in bytes */
+	struct ssc_device *ssc;		/* SSC device for stream */
+	struct atmel_pdc_regs *pdc;	/* PDC receive or transmit registers */
+	struct atmel_ssc_mask *mask;	/* SSC & PDC status bits */
+	struct snd_pcm_substream *substream;
+	void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
+};
+
+extern struct snd_soc_platform atmel_soc_platform;
+
+
+/*
+ * SSC register access (since ssc_writel() / ssc_readl() require literal name)
+ */
+#define ssc_readx(base, reg)            (__raw_readl((base) + (reg)))
+#define ssc_writex(base, reg, value)    __raw_writel((value), (base) + (reg))
+
+#endif /* _ATMEL_PCM_H */
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
new file mode 100644
index 000000000000..c5d67900d666
--- /dev/null
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -0,0 +1,790 @@
+/*
+ * atmel_ssc_dai.c  --  ALSA SoC ATMEL SSC Audio Layer Platform driver
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *         ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/atmel_pdc.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
+#define NUM_SSC_DEVICES		1
+#else
+#define NUM_SSC_DEVICES		3
+#endif
+
+/*
+ * SSC PDC registers required by the PCM DMA engine.
+ */
+static struct atmel_pdc_regs pdc_tx_reg = {
+	.xpr		= ATMEL_PDC_TPR,
+	.xcr		= ATMEL_PDC_TCR,
+	.xnpr		= ATMEL_PDC_TNPR,
+	.xncr		= ATMEL_PDC_TNCR,
+};
+
+static struct atmel_pdc_regs pdc_rx_reg = {
+	.xpr		= ATMEL_PDC_RPR,
+	.xcr		= ATMEL_PDC_RCR,
+	.xnpr		= ATMEL_PDC_RNPR,
+	.xncr		= ATMEL_PDC_RNCR,
+};
+
+/*
+ * SSC & PDC status bits for transmit and receive.
+ */
+static struct atmel_ssc_mask ssc_tx_mask = {
+	.ssc_enable	= SSC_BIT(CR_TXEN),
+	.ssc_disable	= SSC_BIT(CR_TXDIS),
+	.ssc_endx	= SSC_BIT(SR_ENDTX),
+	.ssc_endbuf	= SSC_BIT(SR_TXBUFE),
+	.pdc_enable	= ATMEL_PDC_TXTEN,
+	.pdc_disable	= ATMEL_PDC_TXTDIS,
+};
+
+static struct atmel_ssc_mask ssc_rx_mask = {
+	.ssc_enable	= SSC_BIT(CR_RXEN),
+	.ssc_disable	= SSC_BIT(CR_RXDIS),
+	.ssc_endx	= SSC_BIT(SR_ENDRX),
+	.ssc_endbuf	= SSC_BIT(SR_RXBUFF),
+	.pdc_enable	= ATMEL_PDC_RXTEN,
+	.pdc_disable	= ATMEL_PDC_RXTDIS,
+};
+
+
+/*
+ * DMA parameters.
+ */
+static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+	{{
+	.name		= "SSC0 PCM out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC0 PCM in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	} },
+#if NUM_SSC_DEVICES == 3
+	{{
+	.name		= "SSC1 PCM out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC1 PCM in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	} },
+	{{
+	.name		= "SSC2 PCM out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC2 PCM in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	} },
+#endif
+};
+
+
+static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
+	{
+	.name		= "ssc0",
+	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
+	.dir_mask	= SSC_DIR_MASK_UNUSED,
+	.initialized	= 0,
+	},
+#if NUM_SSC_DEVICES == 3
+	{
+	.name		= "ssc1",
+	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
+	.dir_mask	= SSC_DIR_MASK_UNUSED,
+	.initialized	= 0,
+	},
+	{
+	.name		= "ssc2",
+	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
+	.dir_mask	= SSC_DIR_MASK_UNUSED,
+	.initialized	= 0,
+	},
+#endif
+};
+
+
+/*
+ * SSC interrupt handler.  Passes PDC interrupts to the DMA
+ * interrupt handler in the PCM driver.
+ */
+static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id)
+{
+	struct atmel_ssc_info *ssc_p = dev_id;
+	struct atmel_pcm_dma_params *dma_params;
+	u32 ssc_sr;
+	u32 ssc_substream_mask;
+	int i;
+
+	ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR)
+			& (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR);
+
+	/*
+	 * Loop through the substreams attached to this SSC.  If
+	 * a DMA-related interrupt occurred on that substream, call
+	 * the DMA interrupt handler function, if one has been
+	 * registered in the dma_params structure by the PCM driver.
+	 */
+	for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
+		dma_params = ssc_p->dma_params[i];
+
+		if ((dma_params != NULL) &&
+			(dma_params->dma_intr_handler != NULL)) {
+			ssc_substream_mask = (dma_params->mask->ssc_endx |
+					dma_params->mask->ssc_endbuf);
+			if (ssc_sr & ssc_substream_mask) {
+				dma_params->dma_intr_handler(ssc_sr,
+						dma_params->
+						substream);
+			}
+		}
+	}
+
+	return IRQ_HANDLED;
+}
+
+
+/*-------------------------------------------------------------------------*\
+ * DAI functions
+\*-------------------------------------------------------------------------*/
+/*
+ * Startup.  Only that one substream allowed in each direction.
+ */
+static int atmel_ssc_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	int dir_mask;
+
+	pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
+		ssc_readl(ssc_p->ssc->regs, SR));
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir_mask = SSC_DIR_MASK_PLAYBACK;
+	else
+		dir_mask = SSC_DIR_MASK_CAPTURE;
+
+	spin_lock_irq(&ssc_p->lock);
+	if (ssc_p->dir_mask & dir_mask) {
+		spin_unlock_irq(&ssc_p->lock);
+		return -EBUSY;
+	}
+	ssc_p->dir_mask |= dir_mask;
+	spin_unlock_irq(&ssc_p->lock);
+
+	return 0;
+}
+
+/*
+ * Shutdown.  Clear DMA parameters and shutdown the SSC if there
+ * are no other substreams open.
+ */
+static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir, dir_mask;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	if (dma_params != NULL) {
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+		pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n",
+			(dir ? "receive" : "transmit"),
+			ssc_readl(ssc_p->ssc->regs, SR));
+
+		dma_params->ssc = NULL;
+		dma_params->substream = NULL;
+		ssc_p->dma_params[dir] = NULL;
+	}
+
+	dir_mask = 1 << dir;
+
+	spin_lock_irq(&ssc_p->lock);
+	ssc_p->dir_mask &= ~dir_mask;
+	if (!ssc_p->dir_mask) {
+		if (ssc_p->initialized) {
+			/* Shutdown the SSC clock. */
+			pr_debug("atmel_ssc_dau: Stopping clock\n");
+			clk_disable(ssc_p->ssc->clk);
+
+			free_irq(ssc_p->ssc->irq, ssc_p);
+			ssc_p->initialized = 0;
+		}
+
+		/* Reset the SSC */
+		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+		/* Clear the SSC dividers */
+		ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
+	}
+	spin_unlock_irq(&ssc_p->lock);
+}
+
+
+/*
+ * Record the DAI format for use in hw_params().
+ */
+static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+	ssc_p->daifmt = fmt;
+	return 0;
+}
+
+/*
+ * Record SSC clock dividers for use in hw_params().
+ */
+static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+	switch (div_id) {
+	case ATMEL_SSC_CMR_DIV:
+		/*
+		 * The same master clock divider is used for both
+		 * transmit and receive, so if a value has already
+		 * been set, it must match this value.
+		 */
+		if (ssc_p->cmr_div == 0)
+			ssc_p->cmr_div = div;
+		else
+			if (div != ssc_p->cmr_div)
+				return -EBUSY;
+		break;
+
+	case ATMEL_SSC_TCMR_PERIOD:
+		ssc_p->tcmr_period = div;
+		break;
+
+	case ATMEL_SSC_RCMR_PERIOD:
+		ssc_p->rcmr_period = div;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the SSC.
+ */
+static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	int id = rtd->dai->cpu_dai->id;
+	struct atmel_ssc_info *ssc_p = &ssc_info[id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir, channels, bits;
+	u32 tfmr, rfmr, tcmr, rcmr;
+	int start_event;
+	int ret;
+
+	/*
+	 * Currently, there is only one set of dma params for
+	 * each direction.  If more are added, this code will
+	 * have to be changed to select the proper set.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = &ssc_dma_params[id][dir];
+	dma_params->ssc = ssc_p->ssc;
+	dma_params->substream = substream;
+
+	ssc_p->dma_params[dir] = dma_params;
+
+	/*
+	 * The cpu_dai->dma_data field is only used to communicate the
+	 * appropriate DMA parameters to the pcm driver hw_params()
+	 * function.  It should not be used for other purposes
+	 * as it is common to all substreams.
+	 */
+	rtd->dai->cpu_dai->dma_data = dma_params;
+
+	channels = params_channels(params);
+
+	/*
+	 * Determine sample size in bits and the PDC increment.
+	 */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		bits = 8;
+		dma_params->pdc_xfer_size = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		bits = 16;
+		dma_params->pdc_xfer_size = 2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		bits = 24;
+		dma_params->pdc_xfer_size = 4;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		bits = 32;
+		dma_params->pdc_xfer_size = 4;
+		break;
+	default:
+		printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format");
+		return -EINVAL;
+	}
+
+	/*
+	 * The SSC only supports up to 16-bit samples in I2S format, due
+	 * to the size of the Frame Mode Register FSLEN field.
+	 */
+	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
+		&& bits > 16) {
+		printk(KERN_WARNING
+				"atmel_ssc_dai: sample size %d"
+				"is too large for I2S\n", bits);
+		return -EINVAL;
+	}
+
+	/*
+	 * Compute SSC register settings.
+	 */
+	switch (ssc_p->daifmt
+		& (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
+
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+		/*
+		 * I2S format, SSC provides BCLK and LRC clocks.
+		 *
+		 * The SSC transmit and receive clocks are generated
+		 * from the MCK divider, and the BCLK signal
+		 * is output on the SSC TK line.
+		 */
+		rcmr =	  SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+			| SSC_BF(RCMR_STTDLY, START_DELAY)
+			| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
+			| SSC_BF(RFMR_FSLEN, (bits - 1))
+			| SSC_BF(RFMR_DATNB, (channels - 1))
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+			| SSC_BF(TCMR_STTDLY, START_DELAY)
+			| SSC_BF(TCMR_START, SSC_START_FALLING_RF)
+			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
+			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
+			| SSC_BF(TFMR_FSLEN, (bits - 1))
+			| SSC_BF(TFMR_DATNB, (channels - 1))
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+		/*
+		 * I2S format, CODEC supplies BCLK and LRC clocks.
+		 *
+		 * The SSC transmit clock is obtained from the BCLK signal on
+		 * on the TK line, and the SSC receive clock is
+		 * generated from the transmit clock.
+		 *
+		 *  For single channel data, one sample is transferred
+		 * on the falling edge of the LRC clock.
+		 * For two channel data, one sample is
+		 * transferred on both edges of the LRC clock.
+		 */
+		start_event = ((channels == 1)
+				? SSC_START_FALLING_RF
+				: SSC_START_EDGE_RF);
+
+		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
+			| SSC_BF(RCMR_STTDLY, START_DELAY)
+			| SSC_BF(RCMR_START, start_event)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_CLOCK);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
+			| SSC_BF(RFMR_FSLEN, 0)
+			| SSC_BF(RFMR_DATNB, 0)
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, 0)
+			| SSC_BF(TCMR_STTDLY, START_DELAY)
+			| SSC_BF(TCMR_START, start_event)
+			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
+			| SSC_BF(TFMR_FSLEN, 0)
+			| SSC_BF(TFMR_DATNB, 0)
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+		/*
+		 * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
+		 *
+		 * The SSC transmit and receive clocks are generated from the
+		 * MCK divider, and the BCLK signal is output
+		 * on the SSC TK line.
+		 */
+		rcmr =	  SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+			| SSC_BF(RCMR_STTDLY, 1)
+			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE)
+			| SSC_BF(RFMR_FSLEN, 0)
+			| SSC_BF(RFMR_DATNB, (channels - 1))
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+			| SSC_BF(TCMR_STTDLY, 1)
+			| SSC_BF(TCMR_START, SSC_START_RISING_RF)
+			| SSC_BF(TCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
+			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE)
+			| SSC_BF(TFMR_FSLEN, 0)
+			| SSC_BF(TFMR_DATNB, (channels - 1))
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+	default:
+		printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
+			ssc_p->daifmt);
+		return -EINVAL;
+		break;
+	}
+	pr_debug("atmel_ssc_hw_params: "
+			"RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
+			rcmr, rfmr, tcmr, tfmr);
+
+	if (!ssc_p->initialized) {
+
+		/* Enable PMC peripheral clock for this SSC */
+		pr_debug("atmel_ssc_dai: Starting clock\n");
+		clk_enable(ssc_p->ssc->clk);
+
+		/* Reset the SSC and its PDC registers */
+		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+
+		ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
+
+		ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
+
+		ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0,
+				ssc_p->name, ssc_p);
+		if (ret < 0) {
+			printk(KERN_WARNING
+					"atmel_ssc_dai: request_irq failure\n");
+			pr_debug("Atmel_ssc_dai: Stoping clock\n");
+			clk_disable(ssc_p->ssc->clk);
+			return ret;
+		}
+
+		ssc_p->initialized = 1;
+	}
+
+	/* set SSC clock mode register */
+	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
+
+	/* set receive clock mode and format */
+	ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
+	ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
+
+	/* set transmit clock mode and format */
+	ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
+	ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
+
+	pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n");
+	return 0;
+}
+
+
+static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+
+	pr_debug("%s enabled SSC_SR=0x%08x\n",
+			dir ? "receive" : "transmit",
+			ssc_readl(ssc_p->ssc->regs, SR));
+	return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
+{
+	struct atmel_ssc_info *ssc_p;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssc_p = &ssc_info[cpu_dai->id];
+
+	/* Save the status register before disabling transmit and receive */
+	ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR);
+	ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
+
+	/* Save the current interrupt mask, then disable unmasked interrupts */
+	ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
+	ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
+
+	ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
+	ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
+	ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
+	ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
+	ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
+
+	return 0;
+}
+
+
+
+static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
+{
+	struct atmel_ssc_info *ssc_p;
+	u32 cr;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssc_p = &ssc_info[cpu_dai->id];
+
+	/* restore SSC register settings */
+	ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
+	ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
+	ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
+	ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
+	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
+
+	/* re-enable interrupts */
+	ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
+
+	/* Re-enable recieve and transmit as appropriate */
+	cr = 0;
+	cr |=
+	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
+	cr |=
+	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
+	ssc_writel(ssc_p->ssc->regs, CR, cr);
+
+	return 0;
+}
+#else /* CONFIG_PM */
+#  define atmel_ssc_suspend	NULL
+#  define atmel_ssc_resume	NULL
+#endif /* CONFIG_PM */
+
+
+#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
+			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
+	{	.name = "atmel-ssc0",
+		.id = 0,
+		.suspend = atmel_ssc_suspend,
+		.resume = atmel_ssc_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.ops = {
+			.startup = atmel_ssc_startup,
+			.shutdown = atmel_ssc_shutdown,
+			.prepare = atmel_ssc_prepare,
+			.hw_params = atmel_ssc_hw_params,
+			.set_fmt = atmel_ssc_set_dai_fmt,
+			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.private_data = &ssc_info[0],
+	},
+#if NUM_SSC_DEVICES == 3
+	{	.name = "atmel-ssc1",
+		.id = 1,
+		.suspend = atmel_ssc_suspend,
+		.resume = atmel_ssc_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.ops = {
+			.startup = atmel_ssc_startup,
+			.shutdown = atmel_ssc_shutdown,
+			.prepare = atmel_ssc_prepare,
+			.hw_params = atmel_ssc_hw_params,
+			.set_fmt = atmel_ssc_set_dai_fmt,
+			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.private_data = &ssc_info[1],
+	},
+	{	.name = "atmel-ssc2",
+		.id = 2,
+		.suspend = atmel_ssc_suspend,
+		.resume = atmel_ssc_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.ops = {
+			.startup = atmel_ssc_startup,
+			.shutdown = atmel_ssc_shutdown,
+			.prepare = atmel_ssc_prepare,
+			.hw_params = atmel_ssc_hw_params,
+			.set_fmt = atmel_ssc_set_dai_fmt,
+			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.private_data = &ssc_info[2],
+	},
+#endif
+};
+EXPORT_SYMBOL_GPL(atmel_ssc_dai);
+
+static int __init atmel_ssc_modinit(void)
+{
+	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_init(atmel_ssc_modinit);
+
+static void __exit atmel_ssc_modexit(void)
+{
+	snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_exit(atmel_ssc_modexit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com");
+MODULE_DESCRIPTION("ATMEL SSC ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
new file mode 100644
index 000000000000..a828746e8a2f
--- /dev/null
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -0,0 +1,121 @@
+/*
+ * atmel_ssc_dai.h - ALSA SSC interface for the Atmel  SoC
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *         ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#ifndef _ATMEL_SSC_DAI_H
+#define _ATMEL_SSC_DAI_H
+
+#include <linux/types.h>
+#include <linux/atmel-ssc.h>
+
+#include "atmel-pcm.h"
+
+/* SSC system clock ids */
+#define ATMEL_SYSCLK_MCK	0 /* SSC uses AT91 MCK as system clock */
+
+/* SSC divider ids */
+#define ATMEL_SSC_CMR_DIV	0 /* MCK divider for BCLK */
+#define ATMEL_SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
+#define ATMEL_SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
+/*
+ * SSC direction masks
+ */
+#define SSC_DIR_MASK_UNUSED	0
+#define SSC_DIR_MASK_PLAYBACK	1
+#define SSC_DIR_MASK_CAPTURE	2
+
+/*
+ * SSC register values that Atmel left out of <linux/atmel-ssc.h>.  These
+ * are expected to be used with SSC_BF
+ */
+/* START bit field values */
+#define SSC_START_CONTINUOUS	0
+#define SSC_START_TX_RX		1
+#define SSC_START_LOW_RF	2
+#define SSC_START_HIGH_RF	3
+#define SSC_START_FALLING_RF	4
+#define SSC_START_RISING_RF	5
+#define SSC_START_LEVEL_RF	6
+#define SSC_START_EDGE_RF	7
+#define SSS_START_COMPARE_0	8
+
+/* CKI bit field values */
+#define SSC_CKI_FALLING		0
+#define SSC_CKI_RISING		1
+
+/* CKO bit field values */
+#define SSC_CKO_NONE		0
+#define SSC_CKO_CONTINUOUS	1
+#define SSC_CKO_TRANSFER	2
+
+/* CKS bit field values */
+#define SSC_CKS_DIV		0
+#define SSC_CKS_CLOCK		1
+#define SSC_CKS_PIN		2
+
+/* FSEDGE bit field values */
+#define SSC_FSEDGE_POSITIVE	0
+#define SSC_FSEDGE_NEGATIVE	1
+
+/* FSOS bit field values */
+#define SSC_FSOS_NONE		0
+#define SSC_FSOS_NEGATIVE	1
+#define SSC_FSOS_POSITIVE	2
+#define SSC_FSOS_LOW		3
+#define SSC_FSOS_HIGH		4
+#define SSC_FSOS_TOGGLE		5
+
+#define START_DELAY		1
+
+struct atmel_ssc_state {
+	u32 ssc_cmr;
+	u32 ssc_rcmr;
+	u32 ssc_rfmr;
+	u32 ssc_tcmr;
+	u32 ssc_tfmr;
+	u32 ssc_sr;
+	u32 ssc_imr;
+};
+
+
+struct atmel_ssc_info {
+	char *name;
+	struct ssc_device *ssc;
+	spinlock_t lock;	/* lock for dir_mask */
+	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
+	unsigned short initialized;	/* true if SSC has been initialized */
+	unsigned short daifmt;
+	unsigned short cmr_div;
+	unsigned short tcmr_period;
+	unsigned short rcmr_period;
+	struct atmel_pcm_dma_params *dma_params[2];
+	struct atmel_ssc_state ssc_state;
+};
+extern struct snd_soc_dai atmel_ssc_dai[];
+
+#endif /* _AT91_SSC_DAI_H */
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
similarity index 98%
rename from sound/soc/at32/playpaq_wm8510.c
rename to sound/soc/atmel/playpaq_wm8510.c
index b1966e4dfcd3..43dd8cee83c6 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -22,7 +22,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/errno.h>
 #include <linux/clk.h>
@@ -40,8 +39,8 @@
 #include <mach/portmux.h>
 
 #include "../codecs/wm8510.h"
-#include "at32-pcm.h"
-#include "at32-ssc.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
 
 
 /*-------------------------------------------------------------------------*\
@@ -362,8 +361,9 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = {
 
 
 
-static struct snd_soc_machine snd_soc_machine_playpaq = {
+static struct snd_soc_card snd_soc_playpaq = {
 	.name = "LRS_PlayPaq_WM8510",
+	.platform = &at32_soc_platform,
 	.dai_link = &playpaq_wm8510_dai,
 	.num_links = 1,
 };
@@ -378,8 +378,7 @@ static struct wm8510_setup_data playpaq_wm8510_setup = {
 
 
 static struct snd_soc_device playpaq_wm8510_snd_devdata = {
-	.machine = &snd_soc_machine_playpaq,
-	.platform = &at32_soc_platform,
+	.card = &snd_soc_playpaq,
 	.codec_dev = &soc_codec_dev_wm8510,
 	.codec_data = &playpaq_wm8510_setup,
 };
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
new file mode 100644
index 000000000000..1fb59a9d3719
--- /dev/null
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -0,0 +1,328 @@
+/*
+ * sam9g20_wm8731  --  SoC audio for AT91SAM9G20-based
+ * 			ATMEL AT91SAM9G20ek board.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on ati_b1_wm8731.c by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ * Based on corgi.c by:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	int ret;
+
+	/* codec system clock is supplied by PCK0, set to 12MHz */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+		12000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+	dev_dbg(rtd->socdev->dev, "shutdown");
+}
+
+static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct atmel_ssc_info *ssc_p = cpu_dai->private_data;
+	struct ssc_device *ssc = ssc_p->ssc;
+	int ret;
+
+	unsigned int rate;
+	int cmr_div, period;
+
+	if (ssc == NULL) {
+		printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n");
+		return -EINVAL;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
+	 * field divides the system master clock MCK to drive the SSC TK
+	 * signal which provides the codec BCLK.  The TCMR.PERIOD and
+	 * RCMR.PERIOD fields further divide the BCLK signal to drive
+	 * the SSC TF and RF signals which provide the codec DACLRC and
+	 * ADCLRC clocks.
+	 *
+	 * The dividers were determined through trial and error, where a
+	 * CMR.DIV value is chosen such that the resulting BCLK value is
+	 * divisible, or almost divisible, by (2 * sample rate), and then
+	 * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
+	 */
+	rate = params_rate(params);
+
+	switch (rate) {
+	case 8000:
+		cmr_div = 55;	/* BCLK = 133MHz/(2*55) = 1.209MHz */
+		period = 74;	/* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */
+		break;
+	case 11025:
+		cmr_div = 67;	/* BCLK = 133MHz/(2*60) = 1.108MHz */
+		period = 45;	/* LRC = BCLK/(2*(49+1)) = 11083,3Hz */
+		break;
+	case 16000:
+		cmr_div = 63;	/* BCLK = 133MHz/(2*63) = 1.055MHz */
+		period = 32;	/* LRC = BCLK/(2*(32+1)) = 15993,2Hz */
+		break;
+	case 22050:
+		cmr_div = 52;	/* BCLK = 133MHz/(2*52) = 1.278MHz */
+		period = 28;	/* LRC = BCLK/(2*(28+1)) = 22049Hz */
+		break;
+	case 32000:
+		cmr_div = 66;	/* BCLK = 133MHz/(2*66) = 1.007MHz */
+		period = 15;	/* LRC = BCLK/(2*(15+1)) = 31486,742Hz */
+		break;
+	case 44100:
+		cmr_div = 29;	/* BCLK = 133MHz/(2*29) = 2.293MHz */
+		period = 25;	/* LRC = BCLK/(2*(25+1)) = 44098Hz */
+		break;
+	case 48000:
+		cmr_div = 33;	/* BCLK = 133MHz/(2*33) = 2.015MHz */
+		period = 20;	/* LRC = BCLK/(2*(20+1)) = 47979,79Hz */
+		break;
+	case 88200:
+		cmr_div = 29;	/* BCLK = 133MHz/(2*29) = 2.293MHz */
+		period = 12;	/* LRC = BCLK/(2*(12+1)) = 88196Hz */
+		break;
+	case 96000:
+		cmr_div = 23;	/* BCLK = 133MHz/(2*23) = 2.891MHz */
+		period = 14;	/* LRC = BCLK/(2*(14+1)) = 96376Hz */
+		break;
+	default:
+		printk(KERN_WARNING "unsupported rate %d"
+				" on at91sam9g20ek board\n", rate);
+		return -EINVAL;
+	}
+
+	/* set the MCK divider for BCLK */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div);
+	if (ret < 0)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* set the BCLK divider for DACLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						ATMEL_SSC_TCMR_PERIOD, period);
+	} else {
+		/* set the BCLK divider for ADCLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						ATMEL_SSC_RCMR_PERIOD, period);
+	}
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops at91sam9g20ek_ops = {
+	.startup = at91sam9g20ek_startup,
+	.hw_params = at91sam9g20ek_hw_params,
+	.shutdown = at91sam9g20ek_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+	/* speaker connected to LHPOUT */
+	{"Ext Spk", NULL, "LHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Int Mic"},
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9g20ek board.
+ */
+static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec)
+{
+	printk(KERN_DEBUG
+			"at91sam9g20ek_wm8731 "
+			": at91sam9g20ek_wm8731_init() called\n");
+
+	/* Add specific widgets */
+	snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+				  ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
+	/* Set up specific audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	/* not connected */
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link at91sam9g20ek_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731 PCM",
+	.cpu_dai = &atmel_ssc_dai[0],
+	.codec_dai = &wm8731_dai,
+	.init = at91sam9g20ek_wm8731_init,
+	.ops = &at91sam9g20ek_ops,
+};
+
+static struct snd_soc_card snd_soc_at91sam9g20ek = {
+	.name = "WM8731",
+	.platform = &atmel_soc_platform,
+	.dai_link = &at91sam9g20ek_dai,
+	.num_links = 1,
+};
+
+static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
+	.i2c_bus = 0,
+	.i2c_address = 0x1b,
+};
+
+static struct snd_soc_device at91sam9g20ek_snd_devdata = {
+	.card = &snd_soc_at91sam9g20ek,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &at91sam9g20ek_wm8731_setup,
+};
+
+static struct platform_device *at91sam9g20ek_snd_device;
+
+static int __init at91sam9g20ek_init(void)
+{
+	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+	int ret;
+
+	/*
+	 * Request SSC device
+	 */
+	ssc = ssc_request(0);
+	if (IS_ERR(ssc)) {
+		ret = PTR_ERR(ssc);
+		ssc = NULL;
+		goto err_ssc;
+	}
+	ssc_p->ssc = ssc;
+
+	at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!at91sam9g20ek_snd_device) {
+		printk(KERN_DEBUG
+				"platform device allocation failed\n");
+		ret = -ENOMEM;
+	}
+
+	platform_set_drvdata(at91sam9g20ek_snd_device,
+			&at91sam9g20ek_snd_devdata);
+	at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev;
+
+	ret = platform_device_add(at91sam9g20ek_snd_device);
+	if (ret) {
+		printk(KERN_DEBUG
+				"platform device allocation failed\n");
+		platform_device_put(at91sam9g20ek_snd_device);
+	}
+
+	return ret;
+
+err_ssc:
+	return ret;
+}
+
+static void __exit at91sam9g20ek_exit(void)
+{
+	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+	struct ssc_device *ssc;
+
+	if (ssc_p != NULL) {
+		ssc = ssc_p->ssc;
+		if (ssc != NULL)
+			ssc_free(ssc);
+		ssc_p->ssc = NULL;
+	}
+
+	platform_device_unregister(at91sam9g20ek_snd_device);
+	at91sam9g20ek_snd_device = NULL;
+}
+
+module_init(at91sam9g20ek_init);
+module_exit(at91sam9g20ek_exit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 1466d9328800..74c823d60f91 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -406,11 +406,12 @@ static int __init au1xpsc_audio_dbdma_init(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
-	return 0;
+	return snd_soc_register_platform(&au1xpsc_soc_platform);
 }
 
 static void __exit au1xpsc_audio_dbdma_exit(void)
 {
+	snd_soc_unregister_platform(&au1xpsc_soc_platform);
 }
 
 module_init(au1xpsc_audio_dbdma_init);
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 57facbad6825..f0e30aec7f23 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
-				int cmd)
+				int cmd, struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -313,8 +314,7 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev,
 	au1xpsc_ac97_workdata = NULL;
 }
 
-static int au1xpsc_ac97_suspend(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
 {
 	/* save interesting registers and disable PSC */
 	au1xpsc_ac97_workdata->pm[0] =
@@ -328,8 +328,7 @@ static int au1xpsc_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_ac97_resume(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 {
 	/* restore PSC clock config */
 	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
@@ -345,7 +344,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.probe			= au1xpsc_ac97_probe,
 	.remove			= au1xpsc_ac97_remove,
 	.suspend		= au1xpsc_ac97_suspend,
@@ -372,11 +371,12 @@ EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 static int __init au1xpsc_ac97_init(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_ac97_dai);
 }
 
 static void __exit au1xpsc_ac97_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 }
 
 module_init(au1xpsc_ac97_init);
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 9384702c7ebd..f916de4400ed 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -116,7 +116,8 @@ out:
 }
 
 static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 
@@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
 	return 0;
 }
 
-static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 	int ret, stype = SUBSTREAM_TYPE(substream);
@@ -337,8 +339,7 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev,
 	au1xpsc_i2s_workdata = NULL;
 }
 
-static int au1xpsc_i2s_suspend(struct platform_device *pdev,
-			       struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	/* save interesting register and disable PSC */
 	au1xpsc_i2s_workdata->pm[0] =
@@ -352,8 +353,7 @@ static int au1xpsc_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	/* select I2S mode and PSC clock */
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
@@ -369,7 +369,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
-	.type			= SND_SOC_DAI_I2S,
 	.probe			= au1xpsc_i2s_probe,
 	.remove			= au1xpsc_i2s_remove,
 	.suspend		= au1xpsc_i2s_suspend,
@@ -389,8 +388,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 	.ops = {
 		.trigger	= au1xpsc_i2s_trigger,
 		.hw_params	= au1xpsc_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= au1xpsc_i2s_set_fmt,
 	},
 };
@@ -399,11 +396,12 @@ EXPORT_SYMBOL(au1xpsc_i2s_dai);
 static int __init au1xpsc_i2s_init(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_i2s_dai);
 }
 
 static void __exit au1xpsc_i2s_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 }
 
 module_init(au1xpsc_i2s_init);
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
index f75ae7f62c3d..27683eb7905e 100644
--- a/sound/soc/au1x/sample-ac97.c
+++ b/sound/soc/au1x/sample-ac97.c
@@ -42,14 +42,14 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
 	.ops		= NULL,
 };
 
-static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
+static struct snd_soc_card au1xpsc_sample_ac97_machine = {
 	.name		= "Au1xxx PSC AC97 Audio",
 	.dai_link	= &au1xpsc_sample_ac97_dai,
 	.num_links	= 1,
 };
 
 static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
-	.machine	= &au1xpsc_sample_ac97_machine,
+	.card		= &au1xpsc_sample_ac97_machine,
 	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
 	.codec_dev	= &soc_codec_dev_ac97,
 };
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index dc006206f622..0a2f8f9eff53 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,6 +1,6 @@
 config SND_BF5XX_I2S
 	tristate "SoC I2S Audio for the ADI BF5xx chip"
-	depends on BLACKFIN && SND_SOC
+	depends on BLACKFIN
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -13,7 +13,6 @@ config SND_BF5XX_SOC_SSM2602
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_SSM2602
 	select I2C
-	select I2C_BLACKFIN_TWI
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
@@ -35,7 +34,7 @@ config SND_BFIN_AD73311_SE
 
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
-	depends on BLACKFIN && SND_SOC
+	depends on BLACKFIN
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in slot 16
@@ -47,7 +46,7 @@ config SND_BF5XX_AC97
 	  properly with this driver. This driver is known to work with the
 	  Analog Devices line of AC97 codecs.
 
-config SND_MMAP_SUPPORT
+config SND_BF5XX_MMAP_SUPPORT
 	bool "Enable MMAP Support"
 	depends on SND_BF5XX_AC97
 	default y
@@ -55,9 +54,17 @@ config SND_MMAP_SUPPORT
 	  Say y if you want AC97 driver to support mmap mode.
 	  We introduce an intermediate buffer to simulate mmap.
 
+config SND_BF5XX_MULTICHAN_SUPPORT
+	bool "Enable Multichannel Support"
+	depends on SND_BF5XX_AC97
+	default n
+	help
+	  Say y if you want AC97 driver to support up to 5.1 channel audio.
+	  this mode will consume much more memory for DMA.
+
 config SND_BF5XX_SOC_SPORT
 	tristate
-	
+
 config SND_BF5XX_SOC_I2S
 	tristate
 	select SND_BF5XX_SOC_SPORT
@@ -80,7 +87,7 @@ config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
 	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
 	range 0 3 if BF54x
-	range 0 1 if (BF53x || BF561)
+	range 0 1 if !BF54x
 	default 0
 	help
 	  Set the correct SPORT for sound chip.
@@ -90,12 +97,13 @@ config SND_BF5XX_HAVE_COLD_RESET
 	depends on SND_BF5XX_AC97
 	default y if BFIN548_EZKIT
 	default n if !BFIN548_EZKIT
-	
+
 config SND_BF5XX_RESET_GPIO_NUM
 	int "Set a GPIO for cold reset"
 	depends on SND_BF5XX_HAVE_COLD_RESET
 	range 0 159
 	default 19 if BFIN548_EZKIT
 	default 5 if BFIN537_STAMP
+	default 0
 	help
 	  Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 25e50d2ea1ec..8067cfafa3a7 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -43,24 +43,34 @@
 #include "bf5xx-ac97.h"
 #include "bf5xx-sport.h"
 
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+static unsigned int ac97_chan_mask[] = {
+	SP_FL, /* Mono */
+	SP_STEREO, /* Stereo */
+	SP_2DOT1, /* 2.1*/
+	SP_QUAD,/*Quadraquic*/
+	SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */
+	SP_5DOT1, /* 5.1 */
+};
+
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 	 snd_pcm_uframes_t count)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct sport_device *sport = runtime->private_data;
+	unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		bf5xx_pcm_to_ac97(
-			(struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos,
-			(__u32 *)runtime->dma_area + sport->tx_pos, count);
+		bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf +
+		sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos *
+		runtime->channels, count, chan_mask);
 		sport->tx_pos += runtime->period_size;
 		if (sport->tx_pos >= runtime->buffer_size)
 			sport->tx_pos %= runtime->buffer_size;
 		sport->tx_delay_pos = sport->tx_pos;
 	} else {
-		bf5xx_ac97_to_pcm(
-			(struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
-			(__u32 *)runtime->dma_area + sport->rx_pos, count);
+		bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf +
+		sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos *
+		runtime->channels, count);
 		sport->rx_pos += runtime->period_size;
 		if (sport->rx_pos >= runtime->buffer_size)
 			sport->rx_pos %= runtime->buffer_size;
@@ -71,7 +81,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 static void bf5xx_dma_irq(void *data)
 {
 	struct snd_pcm_substream *pcm = data;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	struct snd_pcm_runtime *runtime = pcm->runtime;
 	struct sport_device *sport = runtime->private_data;
 	bf5xx_mmap_copy(pcm, runtime->period_size);
@@ -90,17 +100,14 @@ static void bf5xx_dma_irq(void *data)
  * The total rx/tx buffer is for ac97 frame to hold all pcm data
  * is  0x20000 * sizeof(struct ac97_frame) / 4.
  */
-#ifdef CONFIG_SND_MMAP_SUPPORT
 static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 				   SNDRV_PCM_INFO_MMAP |
 				   SNDRV_PCM_INFO_MMAP_VALID |
-				   SNDRV_PCM_INFO_BLOCK_TRANSFER,
-#else
-static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
-	.info			= SNDRV_PCM_INFO_INTERLEAVED |
-				  SNDRV_PCM_INFO_BLOCK_TRANSFER,
 #endif
+				   SNDRV_PCM_INFO_BLOCK_TRANSFER,
+
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 0x10000,
@@ -123,10 +130,20 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct sport_device *sport = runtime->private_data;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-	memset(runtime->dma_area, 0, runtime->buffer_size);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		sport->once = 0;
+		if (runtime->dma_area)
+			memset(runtime->dma_area, 0, runtime->buffer_size);
+		memset(sport->tx_dma_buf, 0, runtime->buffer_size *
+			sizeof(struct ac97_frame));
+	} else
+		memset(sport->rx_dma_buf, 0, runtime->buffer_size *
+			sizeof(struct ac97_frame));
+#endif
 	snd_pcm_lib_free_pages(substream);
 	return 0;
 }
@@ -139,7 +156,7 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
 	/* An intermediate buffer is introduced for implementing mmap for
 	 * SPORT working in TMD mode(include AC97).
 	 */
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
 		sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
@@ -173,24 +190,24 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 			bf5xx_mmap_copy(substream, runtime->period_size);
-			snd_pcm_period_elapsed(substream);
 			sport->tx_delay_pos = 0;
+#endif
 			sport_tx_start(sport);
-		}
-		else
+		} else
 			sport_rx_start(sport);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 			sport->tx_pos = 0;
 #endif
 			sport_tx_stop(sport);
 		} else {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 			sport->rx_pos = 0;
 #endif
 			sport_rx_stop(sport);
@@ -208,7 +225,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 	struct sport_device *sport = runtime->private_data;
 	unsigned int curr;
 
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		curr = sport->tx_delay_pos;
 	else
@@ -249,22 +266,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
 	return ret;
 }
 
-static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct sport_device *sport = runtime->private_data;
-
-	pr_debug("%s enter\n", __func__);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		sport->once = 0;
-		memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-	} else
-		memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-
-	return 0;
-}
-
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
 {
@@ -281,32 +283,29 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 		    void __user *buf, snd_pcm_uframes_t count)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-
+	unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
 	pr_debug("%s copy pos:0x%lx count:0x%lx\n",
 			substream->stream ? "Capture" : "Playback", pos, count);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		bf5xx_pcm_to_ac97(
-				(struct ac97_frame *)runtime->dma_area + pos,
-				buf, count);
+		bf5xx_pcm_to_ac97((struct ac97_frame *)runtime->dma_area + pos,
+			(__u16 *)buf, count, chan_mask);
 	else
-		bf5xx_ac97_to_pcm(
-				(struct ac97_frame *)runtime->dma_area + pos,
-				buf, count);
+		bf5xx_ac97_to_pcm((struct ac97_frame *)runtime->dma_area + pos,
+			(__u16 *)buf, count);
 	return 0;
 }
 #endif
 
 struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
-	.close		= bf5xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
 	.hw_free	= bf5xx_pcm_hw_free,
 	.prepare	= bf5xx_pcm_prepare,
 	.trigger	= bf5xx_pcm_trigger,
 	.pointer	= bf5xx_pcm_pointer,
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	.mmap		= bf5xx_pcm_mmap,
 #else
 	.copy		= bf5xx_pcm_copy,
@@ -344,7 +343,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
  * Need to allocate local buffer when enable
  * MMAP for SPORT working in TMD mode (include AC97).
  */
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (!sport_handle->tx_dma_buf) {
 			sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \
@@ -381,7 +380,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	struct snd_pcm_substream *substream;
 	struct snd_dma_buffer *buf;
 	int stream;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	size_t size = bf5xx_pcm_hardware.buffer_bytes_max *
 		sizeof(struct ac97_frame) / 4;
 #endif
@@ -395,7 +394,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 			continue;
 		dma_free_coherent(NULL, buf->bytes, buf->area, 0);
 		buf->area = NULL;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (sport_handle->tx_dma_buf)
 			dma_free_coherent(NULL, size, \
@@ -452,6 +451,18 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
 
+static int __init bfin_ac97_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_ac97_soc_platform);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_ac97_soc_platform);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 5e5aafb6485f..3be2be60576d 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -54,71 +54,103 @@
 static int *cmd_count;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
-#if defined(CONFIG_BF54x)
+static u16 sport_req[][7] = {
+		PIN_REQ_SPORT_0,
+#ifdef PIN_REQ_SPORT_1
+		PIN_REQ_SPORT_1,
+#endif
+#ifdef PIN_REQ_SPORT_2
+		PIN_REQ_SPORT_2,
+#endif
+#ifdef PIN_REQ_SPORT_3
+		PIN_REQ_SPORT_3,
+#endif
+	};
+
 static struct sport_param sport_params[4] = {
-	{
-		.dma_rx_chan	= CH_SPORT0_RX,
-		.dma_tx_chan	= CH_SPORT0_TX,
-		.err_irq	= IRQ_SPORT0_ERR,
-		.regs		= (struct sport_register *)SPORT0_TCR1,
-	},
-	{
-		.dma_rx_chan	= CH_SPORT1_RX,
-		.dma_tx_chan	= CH_SPORT1_TX,
-		.err_irq	= IRQ_SPORT1_ERR,
-		.regs		= (struct sport_register *)SPORT1_TCR1,
-	},
-	{
-		.dma_rx_chan	= CH_SPORT2_RX,
-		.dma_tx_chan	= CH_SPORT2_TX,
-		.err_irq	= IRQ_SPORT2_ERR,
-		.regs		= (struct sport_register *)SPORT2_TCR1,
-	},
-	{
-		.dma_rx_chan	= CH_SPORT3_RX,
-		.dma_tx_chan	= CH_SPORT3_TX,
-		.err_irq	= IRQ_SPORT3_ERR,
-		.regs		= (struct sport_register *)SPORT3_TCR1,
-	}
-};
-#else
-static struct sport_param sport_params[2] = {
 	{
 		.dma_rx_chan	= CH_SPORT0_RX,
 		.dma_tx_chan	= CH_SPORT0_TX,
 		.err_irq	= IRQ_SPORT0_ERROR,
 		.regs		= (struct sport_register *)SPORT0_TCR1,
 	},
+#ifdef PIN_REQ_SPORT_1
 	{
 		.dma_rx_chan	= CH_SPORT1_RX,
 		.dma_tx_chan	= CH_SPORT1_TX,
 		.err_irq	= IRQ_SPORT1_ERROR,
 		.regs		= (struct sport_register *)SPORT1_TCR1,
-	}
-};
+	},
 #endif
+#ifdef PIN_REQ_SPORT_2
+	{
+		.dma_rx_chan	= CH_SPORT2_RX,
+		.dma_tx_chan	= CH_SPORT2_TX,
+		.err_irq	= IRQ_SPORT2_ERROR,
+		.regs		= (struct sport_register *)SPORT2_TCR1,
+	},
+#endif
+#ifdef PIN_REQ_SPORT_3
+	{
+		.dma_rx_chan	= CH_SPORT3_RX,
+		.dma_tx_chan	= CH_SPORT3_TX,
+		.err_irq	= IRQ_SPORT3_ERROR,
+		.regs		= (struct sport_register *)SPORT3_TCR1,
+	}
+#endif
+};
 
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
-		size_t count)
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src,
+		size_t count, unsigned int chan_mask)
 {
 	while (count--) {
-		dst->ac97_tag = TAG_VALID | TAG_PCM;
-		(dst++)->ac97_pcm = *src++;
+		dst->ac97_tag = TAG_VALID;
+		if (chan_mask & SP_FL) {
+			dst->ac97_pcm_r = *src++;
+			dst->ac97_tag |= TAG_PCM_RIGHT;
+		}
+		if (chan_mask & SP_FR) {
+			dst->ac97_pcm_l = *src++;
+			dst->ac97_tag |= TAG_PCM_LEFT;
+
+		}
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+		if (chan_mask & SP_SR) {
+			dst->ac97_sl = *src++;
+			dst->ac97_tag |= TAG_PCM_SL;
+		}
+		if (chan_mask & SP_SL) {
+			dst->ac97_sr = *src++;
+			dst->ac97_tag |= TAG_PCM_SR;
+		}
+		if (chan_mask & SP_LFE) {
+			dst->ac97_lfe = *src++;
+			dst->ac97_tag |= TAG_PCM_LFE;
+		}
+		if (chan_mask & SP_FC) {
+			dst->ac97_center = *src++;
+			dst->ac97_tag |= TAG_PCM_CENTER;
+		}
+#endif
+		dst++;
 	}
 }
 EXPORT_SYMBOL(bf5xx_pcm_to_ac97);
 
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst,
 		size_t count)
 {
-	while (count--)
-		*(dst++) = (src++)->ac97_pcm;
+	while (count--) {
+		*(dst++) = src->ac97_pcm_l;
+		*(dst++) = src->ac97_pcm_r;
+		src++;
+	}
 }
 EXPORT_SYMBOL(bf5xx_ac97_to_pcm);
 
 static unsigned int sport_tx_curr_frag(struct sport_device *sport)
 {
-	return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \
+	return sport->tx_curr_frag = sport_curr_offset_tx(sport) /
 			sport->tx_fragsize;
 }
 
@@ -130,7 +162,7 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
 
 	sport_incfrag(sport, &nextfrag, 1);
 
-	nextwrite = (struct ac97_frame *)(sport->tx_buf + \
+	nextwrite = (struct ac97_frame *)(sport->tx_buf +
 			nextfrag * sport->tx_fragsize);
 	pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n",
 		sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]);
@@ -237,8 +269,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 #ifdef CONFIG_PM
-static int bf5xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
@@ -253,8 +284,7 @@ static int bf5xx_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int bf5xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	int ret;
 	struct sport_device *sport =
@@ -297,20 +327,15 @@ static int bf5xx_ac97_resume(struct platform_device *pdev,
 static int bf5xx_ac97_probe(struct platform_device *pdev,
 			    struct snd_soc_dai *dai)
 {
-	int ret;
-#if defined(CONFIG_BF54x)
-	u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1,
-				 PIN_REQ_SPORT_2, PIN_REQ_SPORT_3};
-#else
-	u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1};
-#endif
+	int ret = 0;
 	cmd_count = (int *)get_zeroed_page(GFP_KERNEL);
 	if (cmd_count == NULL)
 		return -ENOMEM;
 
 	if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
 		pr_err("Requesting Peripherals failed\n");
-		return -EFAULT;
+		ret =  -EFAULT;
+		goto peripheral_err;
 		}
 
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
@@ -318,54 +343,54 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
 	if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) {
 		pr_err("Failed to request GPIO_%d for reset\n",
 				CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-		peripheral_free_list(&sport_req[sport_num][0]);
-		return -1;
+		ret =  -1;
+		goto gpio_err;
 	}
 	gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1);
 #endif
 	sport_handle = sport_init(&sport_params[sport_num], 2, \
 			sizeof(struct ac97_frame), NULL);
 	if (!sport_handle) {
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -ENODEV;
+		ret = -ENODEV;
+		goto sport_err;
 	}
 	/*SPORT works in TDM mode to simulate AC97 transfers*/
 	ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
-		kfree(sport_handle);
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -EBUSY;
+		ret = -EBUSY;
+		goto sport_config_err;
 	}
 
 	ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
 	if (ret) {
 		pr_err("SPORT is busy!\n");
-		kfree(sport_handle);
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -EBUSY;
+		ret = -EBUSY;
+		goto sport_config_err;
 	}
 
 	ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
 	if (ret) {
 		pr_err("SPORT is busy!\n");
-		kfree(sport_handle);
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -EBUSY;
+		ret = -EBUSY;
+		goto sport_config_err;
 	}
+
 	return 0;
+
+sport_config_err:
+	kfree(sport_handle);
+sport_err:
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+gpio_err:
+	peripheral_free_list(&sport_req[sport_num][0]);
+peripheral_err:
+	free_page((unsigned long)cmd_count);
+	cmd_count = NULL;
+
+	return ret;
 }
 
 static void bf5xx_ac97_remove(struct platform_device *pdev,
@@ -373,6 +398,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 {
 	free_page((unsigned long)cmd_count);
 	cmd_count = NULL;
+	peripheral_free_list(&sport_req[sport_num][0]);
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
 #endif
@@ -381,7 +407,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 struct snd_soc_dai bfin_ac97_dai = {
 	.name = "bf5xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = bf5xx_ac97_probe,
 	.remove = bf5xx_ac97_remove,
 	.suspend = bf5xx_ac97_suspend,
@@ -389,7 +415,11 @@ struct snd_soc_dai bfin_ac97_dai = {
 	.playback = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 2,
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+		.channels_max = 6,
+#else
 		.channels_max = 2,
+#endif
 		.rates = SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 	.capture = {
@@ -401,6 +431,18 @@ struct snd_soc_dai bfin_ac97_dai = {
 };
 EXPORT_SYMBOL_GPL(bfin_ac97_dai);
 
+static int __init bfin_ac97_init(void)
+{
+	return snd_soc_register_dai(&bfin_ac97_dai);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_dai(&bfin_ac97_dai);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Roy Huang");
 MODULE_DESCRIPTION("AC97 driver for ADI Blackfin");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f77cc558dc0..3f2a911fe0cb 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -16,21 +16,46 @@ struct ac97_frame {
 	u16 ac97_tag;		/* slot 0 */
 	u16 ac97_addr;		/* slot 1 */
 	u16 ac97_data;		/* slot 2 */
-	u32 ac97_pcm;		/* slot 3 and 4: left and right pcm data */
+	u16 ac97_pcm_l;		/*slot 3:front left*/
+	u16 ac97_pcm_r;		/*slot 4:front left*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	u16 ac97_mdm_l1;
+	u16 ac97_center;	/*slot 6:center*/
+	u16 ac97_sl;		/*slot 7:surround left*/
+	u16 ac97_sr;		/*slot 8:surround right*/
+	u16 ac97_lfe;		/*slot 9:lfe*/
+#endif
 } __attribute__ ((packed));
 
+/* Speaker location */
+#define SP_FL		0x0001
+#define SP_FR		0x0010
+#define SP_FC		0x0002
+#define SP_LFE		0x0020
+#define SP_SL		0x0004
+#define SP_SR		0x0040
+
+#define SP_STEREO	(SP_FL | SP_FR)
+#define SP_2DOT1	(SP_FL | SP_FR | SP_LFE)
+#define SP_QUAD		(SP_FL | SP_FR | SP_SL | SP_SR)
+#define SP_5DOT1	(SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR)
+
 #define TAG_VALID		0x8000
 #define TAG_CMD			0x6000
 #define TAG_PCM_LEFT		0x1000
 #define TAG_PCM_RIGHT		0x0800
-#define TAG_PCM			(TAG_PCM_LEFT | TAG_PCM_RIGHT)
+#define TAG_PCM_MDM_L1		0x0400
+#define TAG_PCM_CENTER		0x0200
+#define TAG_PCM_SL		0x0100
+#define TAG_PCM_SR		0x0080
+#define TAG_PCM_LFE		0x0040
 
 extern struct snd_soc_dai bfin_ac97_dai;
 
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
-		size_t count);
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \
+		size_t count, unsigned int chan_mask);
 
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \
 		size_t count);
 
 #endif
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
index 124425d22320..d8f591273778 100644
--- a/sound/soc/blackfin/bf5xx-ad1980.c
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -43,7 +43,7 @@
 #include "bf5xx-ac97-pcm.h"
 #include "bf5xx-ac97.h"
 
-static struct snd_soc_machine bf5xx_board;
+static struct snd_soc_card bf5xx_board;
 
 static int bf5xx_board_startup(struct snd_pcm_substream *substream)
 {
@@ -67,15 +67,15 @@ static struct snd_soc_dai_link bf5xx_board_dai = {
 	.ops = &bf5xx_board_ops,
 };
 
-static struct snd_soc_machine bf5xx_board = {
+static struct snd_soc_card bf5xx_board = {
 	.name = "bf5xx-board",
+	.platform = &bf5xx_ac97_soc_platform,
 	.dai_link = &bf5xx_board_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_board_snd_devdata = {
-	.machine = &bf5xx_board,
-	.platform = &bf5xx_ac97_soc_platform,
+	.card = &bf5xx_board,
 	.codec_dev = &soc_codec_dev_ad1980,
 };
 
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 622c9b909532..7f2a5e199075 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -65,7 +65,7 @@
 
 #define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
 
-static struct snd_soc_machine bf5xx_ad73311;
+static struct snd_soc_card bf5xx_ad73311;
 
 static int snd_ad73311_startup(void)
 {
@@ -168,7 +168,7 @@ static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
 		params_format(params));
 
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
@@ -190,16 +190,16 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = {
 	.ops = &bf5xx_ad73311_ops,
 };
 
-static struct snd_soc_machine bf5xx_ad73311 = {
+static struct snd_soc_card bf5xx_ad73311 = {
 	.name = "bf5xx_ad73311",
+	.platform = &bf5xx_i2s_soc_platform,
 	.probe = bf5xx_probe,
 	.dai_link = &bf5xx_ad73311_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
-	.machine = &bf5xx_ad73311,
-	.platform = &bf5xx_i2s_soc_platform,
+	.card = &bf5xx_ad73311,
 	.codec_dev = &soc_codec_dev_ad73311,
 };
 
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 61fccf925192..53d290b3ea47 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -283,6 +283,18 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
 
+static int __init bfin_i2s_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_i2s_soc_platform);
+}
+module_exit(bfin_i2s_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index e020c160ee44..d1d95d2393fe 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -132,7 +132,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	return ret;
 }
 
-static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
+static int bf5xx_i2s_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 
@@ -142,7 +143,8 @@ static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -193,7 +195,8 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	bf5xx_i2s.counter--;
@@ -219,16 +222,14 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	peripheral_free_list(&sport_req[sport_num][0]);
 }
 
 #ifdef CONFIG_PM
-static int bf5xx_i2s_suspend(struct platform_device *dev,
-			     struct snd_soc_dai *dai)
+static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
@@ -289,7 +290,6 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = bf5xx_i2s_probe,
 	.remove = bf5xx_i2s_remove,
 	.suspend = bf5xx_i2s_suspend,
@@ -307,13 +307,24 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 	.ops = {
 		.startup   = bf5xx_i2s_startup,
 		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = bf5xx_i2s_hw_params,
 		.set_fmt = bf5xx_i2s_set_dai_fmt,
 	},
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
+static int __init bfin_i2s_init(void)
+{
+	return snd_soc_register_dai(&bf5xx_i2s_dai);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&bf5xx_i2s_dai);
+}
+module_exit(bfin_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("I2S driver for ADI Blackfin");
diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
index fcadcc081f7f..2e63dea73e9c 100644
--- a/sound/soc/blackfin/bf5xx-sport.h
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -116,7 +116,7 @@ struct sport_device {
 	void *err_data;
 	unsigned char *tx_dma_buf;
 	unsigned char *rx_dma_buf;
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT
 	dma_addr_t tx_dma_phy;
 	dma_addr_t rx_dma_phy;
 	int tx_pos;/*pcm sample count*/
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e15f67fd7769..bc0cdded7116 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,7 +44,7 @@
 #include "bf5xx-i2s-pcm.h"
 #include "bf5xx-i2s.h"
 
-static struct snd_soc_machine bf5xx_ssm2602;
+static struct snd_soc_card bf5xx_ssm2602;
 
 static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream)
 {
@@ -92,17 +92,17 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
 	 */
 
 	/* set codec DAI configuration */
-	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
 
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -135,15 +135,15 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
 	.i2c_address = 0x1b,
 };
 
-static struct snd_soc_machine bf5xx_ssm2602 = {
+static struct snd_soc_card bf5xx_ssm2602 = {
 	.name = "bf5xx_ssm2602",
+	.platform = &bf5xx_i2s_soc_platform,
 	.dai_link = &bf5xx_ssm2602_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
-	.machine = &bf5xx_ssm2602,
-	.platform = &bf5xx_i2s_soc_platform,
+	.card = &bf5xx_ssm2602,
 	.codec_dev = &soc_codec_dev_ssm2602,
 	.codec_data = &bf5xx_ssm2602_setup,
 };
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 38a0e3b620a7..c41289b5f586 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,40 @@
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
-	depends on I2C
-	select SPI
-	select SPI_MASTER
-	select SND_SOC_AD73311
-	select SND_SOC_AK4535
-	select SND_SOC_CS4270
-	select SND_SOC_SSM2602
-	select SND_SOC_TLV320AIC23
-	select SND_SOC_TLV320AIC26
-	select SND_SOC_TLV320AIC3X
-	select SND_SOC_UDA1380
-	select SND_SOC_WM8510
-	select SND_SOC_WM8580
-	select SND_SOC_WM8731
-	select SND_SOC_WM8750
-	select SND_SOC_WM8753
-	select SND_SOC_WM8900
-	select SND_SOC_WM8903
-	select SND_SOC_WM8971
-	select SND_SOC_WM8990
+	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
+	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
+	select SND_SOC_AD73311 if I2C
+	select SND_SOC_AK4535 if I2C
+	select SND_SOC_CS4270 if I2C
+	select SND_SOC_PCM3008
+	select SND_SOC_SSM2602 if I2C
+	select SND_SOC_TLV320AIC23 if I2C
+	select SND_SOC_TLV320AIC26 if SPI_MASTER
+	select SND_SOC_TLV320AIC3X if I2C
+	select SND_SOC_TWL4030 if TWL4030_CORE
+	select SND_SOC_UDA134X
+	select SND_SOC_UDA1380 if I2C
+	select SND_SOC_WM8350 if MFD_WM8350
+	select SND_SOC_WM8510 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8580 if I2C
+	select SND_SOC_WM8728 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8731 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8750 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8753 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8900 if I2C
+	select SND_SOC_WM8903 if I2C
+	select SND_SOC_WM8971 if I2C
+	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
+	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
         help
           Normally ASoC codec drivers are only built if a machine driver which
           uses them is also built since they are only usable with a machine
           driver.  Selecting this option will allow these drivers to be built
           without an explicit machine driver for test and development purposes.
 
+	  Support for the bus types used to access the codecs to be built must
+	  be selected separately.
+
           If unsure select "N".
 
 
@@ -60,6 +69,12 @@ config SND_SOC_CS4270_VD33_ERRATA
 	bool
 	depends on SND_SOC_CS4270
 
+config SND_SOC_L3
+       tristate
+
+config SND_SOC_PCM3008
+       tristate
+
 config SND_SOC_SSM2602
 	tristate
 
@@ -75,15 +90,29 @@ config SND_SOC_TLV320AIC3X
 	tristate
 	depends on I2C
 
+config SND_SOC_TWL4030
+	tristate
+	depends on TWL4030_CORE
+
+config SND_SOC_UDA134X
+       tristate
+       select SND_SOC_L3
+
 config SND_SOC_UDA1380
         tristate
 
+config SND_SOC_WM8350
+	tristate
+
 config SND_SOC_WM8510
 	tristate
 
 config SND_SOC_WM8580
 	tristate
 
+config SND_SOC_WM8728
+	tristate
+
 config SND_SOC_WM8731
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 90f0a585fc70..c4ddc9aa2bbd 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,13 +3,19 @@ snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
+snd-soc-l3-objs := l3.o
+snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-twl4030-objs := twl4030.o
+snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8728-objs := wm8728.o
 snd-soc-wm8731-objs := wm8731.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
@@ -25,13 +31,19 @@ obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index bd1ebdc6c86c..fb53e6511af2 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -24,7 +24,8 @@
 
 #define AC97_VERSION "0.6"
 
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -42,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 1,
@@ -113,7 +114,7 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto bus_err;
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
 		goto bus_err;
 	return 0;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 1397b8e06c0b..73fdbb4d4a3d 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -85,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
 SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
 SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
 
+SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
+
 SOC_ENUM("Capture Source", ad1980_cap_src),
 
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
@@ -142,10 +145,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 struct snd_soc_dai ad1980_dai = {
 	.name = "AC97",
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 2,
-		.channels_max = 2,
+		.channels_max = 6,
 		.rates = SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 	.capture = {
@@ -192,6 +196,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret = 0;
 	u16 vendor_id2;
+	u16 ext_status;
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
@@ -234,7 +239,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 
 	ret = ad1980_reset(codec, 0);
 	if (ret < 0) {
-		printk(KERN_ERR "AC97 link error\n");
+		printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
 		goto reset_err;
 	}
 
@@ -253,12 +258,19 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 				"supported\n");
 	}
 
-	ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
-	ac97_write(codec, AC97_PCM, 0x0000);	/* unmute PCM out volume */
-	ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+	/* unmute captures and playbacks volume */
+	ac97_write(codec, AC97_MASTER, 0x0000);
+	ac97_write(codec, AC97_PCM, 0x0000);
+	ac97_write(codec, AC97_REC_GAIN, 0x0000);
+	ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
+	ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
+
+	/*power on LFE/CENTER/Surround DACs*/
+	ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
 
 	ad1980_add_controls(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register card\n");
 		goto reset_err;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index 37af8607b00a..b09289a1e55a 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -8,14 +8,10 @@
  *  under  the terms of  the GNU General  Public License as published by the
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
- *
- *  Revision history
- *    25th Sep 2008   Initial version.
  */
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
@@ -68,7 +64,7 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad73311: failed to register card\n");
 		goto register_err;
@@ -102,6 +98,18 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
 
+static int __init ad73311_init(void)
+{
+	return snd_soc_register_dai(&ad73311_dai);
+}
+module_init(ad73311_init);
+
+static void __exit ad73311_exit(void)
+{
+	snd_soc_unregister_dai(&ad73311_dai);
+}
+module_exit(ad73311_exit);
+
 MODULE_DESCRIPTION("ASoC ad73311 driver");
 MODULE_AUTHOR("Cliff Cai ");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 2a89b5888e11..81300d8d42ca 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -339,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 }
 
 static int ak4535_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -451,8 +452,6 @@ struct snd_soc_dai ak4535_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
 		.hw_params = ak4535_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt = ak4535_set_dai_fmt,
 		.digital_mute = ak4535_mute,
 		.set_sysclk = ak4535_set_dai_sysclk,
@@ -513,7 +512,7 @@ static int ak4535_init(struct snd_soc_device *socdev)
 
 	ak4535_add_controls(codec);
 	ak4535_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ak4535: failed to register card\n");
 		goto card_err;
@@ -689,6 +688,18 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
 
+static int __init ak4535_modinit(void)
+{
+	return snd_soc_register_dai(&ak4535_dai);
+}
+module_init(ak4535_modinit);
+
+static void __exit ak4535_exit(void)
+{
+	snd_soc_unregister_dai(&ak4535_dai);
+}
+module_exit(ak4535_exit);
+
 MODULE_DESCRIPTION("Soc AK4535 driver");
 MODULE_AUTHOR("Richard Purdie");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0bbd94501d7e..f1aa0c34421c 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 /*
  * Program the CS4270 with the given hardware parameters.
  *
- * The .dai_ops functions are used to provide board-specific data, like
+ * The .ops functions are used to provide board-specific data, like
  * input frequencies, to this driver.  This function takes that information,
  * combines it with the hardware parameters provided, and programs the
  * hardware accordingly.
  */
 static int cs4270_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -450,6 +451,19 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	/* Disable automatic volume control.  It's enabled by default, and
+	 * it causes volume change commands to be delayed, sometimes until
+	 * after playback has started.
+	 */
+
+	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+	if (ret < 0) {
+		printk(KERN_ERR "I2C write failed\n");
+		return ret;
+	}
+
 	/* Thaw and power-up the codec */
 
 	ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
@@ -697,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev)
 	if (codec->control_data) {
 		/* Initialize codec ops */
 		cs4270_dai.ops.hw_params = cs4270_hw_params;
-		cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk;
-		cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt;
+		cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
+		cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
 #ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-		cs4270_dai.dai_ops.digital_mute = cs4270_mute;
+		cs4270_dai.ops.digital_mute = cs4270_mute;
 #endif
 	} else
 		printk(KERN_INFO "cs4270: no I2C device found, "
@@ -709,7 +723,7 @@ static int cs4270_probe(struct platform_device *pdev)
 	printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
 #endif
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "cs4270: failed to register card\n");
 		goto error_del_driver;
@@ -760,6 +774,18 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
+static int __init cs4270_init(void)
+{
+	return snd_soc_register_dai(&cs4270_dai);
+}
+module_init(cs4270_init);
+
+static void __exit cs4270_exit(void)
+{
+	snd_soc_unregister_dai(&cs4270_dai);
+}
+module_exit(cs4270_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c
new file mode 100644
index 000000000000..5353af58862c
--- /dev/null
+++ b/sound/soc/codecs/l3.c
@@ -0,0 +1,91 @@
+/*
+ * L3 code
+ *
+ *  Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ *
+ * based on:
+ *
+ * L3 bus algorithm module.
+ *
+ *  Copyright (C) 2001 Russell King, All Rights Reserved.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <sound/l3.h>
+
+/*
+ * Send one byte of data to the chip.  Data is latched into the chip on
+ * the rising edge of the clock.
+ */
+static void sendbyte(struct l3_pins *adap, unsigned int byte)
+{
+	int i;
+
+	for (i = 0; i < 8; i++) {
+		adap->setclk(0);
+		udelay(adap->data_hold);
+		adap->setdat(byte & 1);
+		udelay(adap->data_setup);
+		adap->setclk(1);
+		udelay(adap->clock_high);
+		byte >>= 1;
+	}
+}
+
+/*
+ * Send a set of bytes to the chip.  We need to pulse the MODE line
+ * between each byte, but never at the start nor at the end of the
+ * transfer.
+ */
+static void sendbytes(struct l3_pins *adap, const u8 *buf,
+		      int len)
+{
+	int i;
+
+	for (i = 0; i < len; i++) {
+		if (i) {
+			udelay(adap->mode_hold);
+			adap->setmode(0);
+			udelay(adap->mode);
+		}
+		adap->setmode(1);
+		udelay(adap->mode_setup);
+		sendbyte(adap, buf[i]);
+	}
+}
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len)
+{
+	adap->setclk(1);
+	adap->setdat(1);
+	adap->setmode(1);
+	udelay(adap->mode);
+
+	adap->setmode(0);
+	udelay(adap->mode_setup);
+	sendbyte(adap, addr);
+	udelay(adap->mode_hold);
+
+	sendbytes(adap, data, len);
+
+	adap->setclk(1);
+	adap->setdat(1);
+	adap->setmode(0);
+
+	return len;
+}
+EXPORT_SYMBOL_GPL(l3_write);
+
+MODULE_DESCRIPTION("L3 bit-banging driver");
+MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
new file mode 100644
index 000000000000..9a3e67e5319c
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.c
@@ -0,0 +1,212 @@
+/*
+ * ALSA Soc PCM3008 codec support
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on AC97 Soc codec, original copyright follow:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ * Generic PCM3008 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "pcm3008.h"
+
+#define PCM3008_VERSION "0.2"
+
+#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |	\
+		       SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai pcm3008_dai = {
+	.name = "PCM3008 HiFi",
+	.playback = {
+		.stream_name = "PCM3008 Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = PCM3008_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "PCM3008 Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = PCM3008_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+};
+EXPORT_SYMBOL_GPL(pcm3008_dai);
+
+static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
+{
+	gpio_free(setup->dem0_pin);
+	gpio_free(setup->dem1_pin);
+	gpio_free(setup->pdad_pin);
+	gpio_free(setup->pdda_pin);
+}
+
+static int pcm3008_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+	int ret = 0;
+
+	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->codec)
+		return -ENOMEM;
+
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->name = "PCM3008";
+	codec->owner = THIS_MODULE;
+	codec->dai = &pcm3008_dai;
+	codec->num_dai = 1;
+	codec->write = NULL;
+	codec->read = NULL;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* Register PCMs. */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "pcm3008: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* Register Card. */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "pcm3008: failed to register card\n");
+		goto card_err;
+	}
+
+	/* DEM1  DEM0  DE-EMPHASIS_MODE
+	 * Low   Low   De-emphasis 44.1 kHz ON
+	 * Low   High  De-emphasis OFF
+	 * High  Low   De-emphasis 48 kHz ON
+	 * High  High  De-emphasis 32 kHz ON
+	 */
+
+	/* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
+	ret = gpio_request(setup->dem0_pin, "codec_dem0");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->dem0_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
+	ret = gpio_request(setup->dem1_pin, "codec_dem1");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->dem1_pin, 0);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure PDAD GPIO. */
+	ret = gpio_request(setup->pdad_pin, "codec_pdad");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->pdad_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure PDDA GPIO. */
+	ret = gpio_request(setup->pdda_pin, "codec_pdda");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->pdda_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	return ret;
+
+gpio_err:
+	pcm3008_gpio_free(setup);
+card_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	kfree(socdev->codec);
+
+	return ret;
+}
+
+static int pcm3008_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	if (!codec)
+		return 0;
+
+	pcm3008_gpio_free(setup);
+	snd_soc_free_pcms(socdev);
+	kfree(socdev->codec);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	gpio_set_value(setup->pdad_pin, 0);
+	gpio_set_value(setup->pdda_pin, 0);
+
+	return 0;
+}
+
+static int pcm3008_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	gpio_set_value(setup->pdad_pin, 1);
+	gpio_set_value(setup->pdda_pin, 1);
+
+	return 0;
+}
+#else
+#define pcm3008_soc_suspend NULL
+#define pcm3008_soc_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
+	.probe = 	pcm3008_soc_probe,
+	.remove = 	pcm3008_soc_remove,
+	.suspend =	pcm3008_soc_suspend,
+	.resume =	pcm3008_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
+
+static int __init pcm3008_init(void)
+{
+	return snd_soc_register_dai(&pcm3008_dai);
+}
+module_init(pcm3008_init);
+
+static void __exit pcm3008_exit(void)
+{
+	snd_soc_unregister_dai(&pcm3008_dai);
+}
+module_exit(pcm3008_exit);
+
+MODULE_DESCRIPTION("Soc PCM3008 driver");
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h
new file mode 100644
index 000000000000..d04e87d3c060
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.h
@@ -0,0 +1,25 @@
+/*
+ * PCM3008 ALSA SoC Layer
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_PCM3008_H
+#define __LINUX_SND_SOC_PCM3008_H
+
+struct pcm3008_setup_data {
+	unsigned dem0_pin;
+	unsigned dem1_pin;
+	unsigned pdad_pin;
+	unsigned pdda_pin;
+};
+
+extern struct snd_soc_codec_device soc_codec_dev_pcm3008;
+extern struct snd_soc_dai pcm3008_dai;
+
+#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 44ef0dacd564..cac373616768 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -285,16 +285,23 @@ static inline int get_coeff(int mclk, int rate)
 }
 
 static int ssm2602_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	u16 srate;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct i2c_client *i2c = codec->control_data;
 	u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
 	int i = get_coeff(ssm2602->sysclk, params_rate(params));
 
+	if (substream == ssm2602->slave_substream) {
+		dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
+		return 0;
+	}
+
 	/*no match is found*/
 	if (i == ARRAY_SIZE(coeff_div))
 		return -EINVAL;
@@ -324,19 +331,26 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ssm2602_startup(struct snd_pcm_substream *substream)
+static int ssm2602_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
 
 	/* The DAI has shared clocks so if we already have a playback or
 	 * capture going then constrain this substream to match it.
+	 * TODO: the ssm2602 allows pairs of non-matching PB/REC rates
 	 */
 	if (ssm2602->master_substream) {
 		master_runtime = ssm2602->master_substream->runtime;
+		dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+			master_runtime->sample_bits,
+			master_runtime->rate);
+
 		snd_pcm_hw_constraint_minmax(substream->runtime,
 					     SNDRV_PCM_HW_PARAM_RATE,
 					     master_runtime->rate,
@@ -354,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
+static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -365,14 +380,21 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void ssm2602_shutdown(struct snd_pcm_substream *substream)
+static void ssm2602_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
+	struct ssm2602_priv *ssm2602 = codec->private_data;
 	/* deactivate */
 	if (!codec->active)
 		ssm2602_write(codec, SSM2602_ACTIVE, 0);
+
+	if (ssm2602->master_substream == substream)
+		ssm2602->master_substream = ssm2602->slave_substream;
+
+	ssm2602->slave_substream = NULL;
 }
 
 static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
@@ -432,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface |= 0x0001;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
-		iface |= 0x0003;
+		iface |= 0x0013;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x0013;
+		iface |= 0x0003;
 		break;
 	default:
 		return -EINVAL;
@@ -496,6 +518,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
 		SNDRV_PCM_RATE_96000)
 
+#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
 struct snd_soc_dai ssm2602_dai = {
 	.name = "SSM2602",
 	.playback = {
@@ -503,20 +528,18 @@ struct snd_soc_dai ssm2602_dai = {
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE,},
+		.formats = SSM2602_FORMATS,},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE,},
+		.formats = SSM2602_FORMATS,},
 	.ops = {
 		.startup = ssm2602_startup,
 		.prepare = ssm2602_pcm_prepare,
 		.hw_params = ssm2602_hw_params,
 		.shutdown = ssm2602_shutdown,
-	},
-	.dai_ops = {
 		.digital_mute = ssm2602_mute,
 		.set_sysclk = ssm2602_set_dai_sysclk,
 		.set_fmt = ssm2602_set_dai_fmt,
@@ -601,7 +624,7 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 
 	ssm2602_add_controls(codec);
 	ssm2602_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		pr_err("ssm2602: failed to register card\n");
 		goto card_err;
@@ -770,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
 
+static int __init ssm2602_modinit(void)
+{
+	return snd_soc_register_dai(&ssm2602_dai);
+}
+module_init(ssm2602_modinit);
+
+static void __exit ssm2602_exit(void)
+{
+	snd_soc_unregister_dai(&ssm2602_dai);
+}
+module_exit(ssm2602_exit);
+
 MODULE_DESCRIPTION("ASoC ssm2602 driver");
 MODULE_AUTHOR("Cliff Cai");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 44308dac9e18..cfdea007c4cb 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -37,12 +37,6 @@
 
 #define AIC23_VERSION "0.1"
 
-struct tlv320aic23_srate_reg_info {
-	u32 sample_rate;
-	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
-	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
 /*
  * AIC23 register cache
  */
@@ -261,20 +255,156 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 };
 
-/* tlv320aic23 related */
-static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
-	{4000, 0x06, 1},	/*  4000 */
-	{8000, 0x06, 0},	/*  8000 */
-	{16000, 0x0C, 1},	/* 16000 */
-	{22050, 0x11, 1},	/* 22050 */
-	{24000, 0x00, 1},	/* 24000 */
-	{32000, 0x0C, 0},	/* 32000 */
-	{44100, 0x11, 0},	/* 44100 */
-	{48000, 0x00, 0},	/* 48000 */
-	{88200, 0x1F, 0},	/* 88200 */
-	{96000, 0x0E, 0},	/* 96000 */
+/* AIC23 driver data */
+struct aic23 {
+	struct snd_soc_codec codec;
+	int mclk;
+	int requested_adc;
+	int requested_dac;
 };
 
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k  /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k    /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k    /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k   /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+	128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11)        | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 0*/
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
+	LOWER_GROUP,			/* Usb, bosr - 0*/
+	UPPER_GROUP,			/* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (x) ? (SR_MULT/x) : 0
+static const unsigned char sr_adc_mult_table[] = {
+	A(2), A(2), A(12), A(12),  A(0), A(0), A(3), A(1),
+	A(2), A(2), A(11), A(11),  A(0), A(0), A(0), A(1)
+};
+static const unsigned char sr_dac_mult_table[] = {
+	A(2), A(12), A(2), A(12),  A(0), A(0), A(3), A(1),
+	A(2), A(11), A(2), A(11),  A(0), A(0), A(0), A(1)
+};
+
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+		int dac, int dac_l, int dac_h, int need_dac)
+{
+	if ((adc >= adc_l) && (adc <= adc_h) &&
+			(dac >= dac_l) && (dac <= dac_h)) {
+		int diff_adc = need_adc - adc;
+		int diff_dac = need_dac - dac;
+		return abs(diff_adc) + abs(diff_dac);
+	}
+	return UINT_MAX;
+}
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+	int i, j;
+	int best_i = -1;
+	int best_j = -1;
+	int best_div = 0;
+	unsigned best_score = UINT_MAX;
+	int adc_l, adc_h, dac_l, dac_h;
+
+	need_adc *= SR_MULT;
+	need_dac *= SR_MULT;
+	/*
+	 * rates given are +/- 1/32
+	 */
+	adc_l = need_adc - (need_adc >> 5);
+	adc_h = need_adc + (need_adc >> 5);
+	dac_l = need_dac - (need_dac >> 5);
+	dac_h = need_dac + (need_dac >> 5);
+	for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
+		int base = mclk / bosr_usb_divisor_table[i];
+		int mask = sr_valid_mask[i];
+		for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+				j++, mask >>= 1) {
+			int adc;
+			int dac;
+			int score;
+			if ((mask & 1) == 0)
+				continue;
+			adc = base * sr_adc_mult_table[j];
+			dac = base * sr_dac_mult_table[j];
+			score = get_score(adc, adc_l, adc_h, need_adc,
+					dac, dac_l, dac_h, need_dac);
+			if (best_score > score) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 0;
+			}
+			score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+					(dac >> 1), dac_l, dac_h, need_dac);
+			/* prefer to have a /2 */
+			if ((score != UINT_MAX) && (best_score >= score)) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 1;
+			}
+		}
+	}
+	return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+#ifdef DEBUG
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+		u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+	int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+	int sr = (src >> 2) & 0x0f;
+	int val = (mclk / bosr_usb_divisor_table[src & 3]);
+	int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+	int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+	if (src & TLV320AIC23_CLKIN_HALF) {
+		adc >>= 1;
+		dac >>= 1;
+	}
+	*sample_rate_adc = adc;
+	*sample_rate_dac = dac;
+}
+#endif
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+		u32 sample_rate_adc, u32 sample_rate_dac)
+{
+	/* Search for the right sample rate */
+	int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+	if (data < 0) {
+		printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+				__func__, sample_rate_adc, sample_rate_dac);
+		return -EINVAL;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+#ifdef DEBUG
+	{
+		u32 adc, dac;
+		get_current_sample_rates(codec, mclk, &adc, &dac);
+		printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+			adc, dac, data);
+	}
+#endif
+	return 0;
+}
+
 static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 {
 	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
@@ -288,32 +418,36 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	u16 iface_reg, data;
-	u8 count = 0;
+	u16 iface_reg;
+	int ret;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+	u32 sample_rate_adc = aic23->requested_adc;
+	u32 sample_rate_dac = aic23->requested_dac;
+	u32 sample_rate = params_rate(params);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		aic23->requested_dac = sample_rate_dac = sample_rate;
+		if (!sample_rate_adc)
+			sample_rate_adc = sample_rate;
+	} else {
+		aic23->requested_adc = sample_rate_adc = sample_rate;
+		if (!sample_rate_dac)
+			sample_rate_dac = sample_rate;
+	}
+	ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+			sample_rate_dac);
+	if (ret < 0)
+		return ret;
 
 	iface_reg =
 	    tlv320aic23_read_reg_cache(codec,
 				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-
-	/* Search for the right sample rate */
-	/* Verify what happens if the rate is not supported
-	 * now it goes to 96Khz */
-	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
-	       (count < ARRAY_SIZE(srate_reg_info))) {
-		count++;
-	}
-
-	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
-		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
-		TLV320AIC23_USB_CLK_ON;
-
-	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
-
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
@@ -332,7 +466,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -344,17 +479,23 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	/* deactivate */
 	if (!codec->active) {
 		udelay(50);
 		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		aic23->requested_dac = 0;
+	else
+		aic23->requested_adc = 0;
 }
 
 static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
@@ -400,7 +541,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case SND_SOC_DAIFMT_I2S:
 		iface_reg |= TLV320AIC23_FOR_I2S;
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
 		iface_reg |= TLV320AIC23_FOR_DSP;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
@@ -422,12 +563,9 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				      int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-
-	switch (freq) {
-	case 12000000:
-		return 0;
-	}
-	return -EINVAL;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+	aic23->mclk = freq;
+	return 0;
 }
 
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
@@ -478,12 +616,10 @@ struct snd_soc_dai tlv320aic23_dai = {
 		.prepare = tlv320aic23_pcm_prepare,
 		.hw_params = tlv320aic23_hw_params,
 		.shutdown = tlv320aic23_shutdown,
-		},
-	.dai_ops = {
-		    .digital_mute = tlv320aic23_mute,
-		    .set_fmt = tlv320aic23_set_dai_fmt,
-		    .set_sysclk = tlv320aic23_set_dai_sysclk,
-		    }
+		.digital_mute = tlv320aic23_mute,
+		.set_fmt = tlv320aic23_set_dai_fmt,
+		.set_sysclk = tlv320aic23_set_dai_sysclk,
+	}
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
@@ -584,7 +720,7 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
 
 	tlv320aic23_add_controls(codec);
 	tlv320aic23_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "tlv320aic23: failed to register card\n");
 		goto card_err;
@@ -659,14 +795,15 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
+	struct aic23 *aic23;
 	int ret = 0;
 
 	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
+	aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL);
+	if (aic23 == NULL)
 		return -ENOMEM;
-
+	codec = &aic23->codec;
 	socdev->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -687,6 +824,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	if (codec->control_data)
 		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -697,7 +835,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
 	i2c_del_driver(&tlv320aic23_i2c_driver);
 #endif
 	kfree(codec->reg_cache);
-	kfree(codec);
+	kfree(aic23);
 
 	return 0;
 }
@@ -709,6 +847,18 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
 
+static int __init tlv320aic23_modinit(void)
+{
+	return snd_soc_register_dai(&tlv320aic23_dai);
+}
+module_init(tlv320aic23_modinit);
+
+static void __exit tlv320aic23_exit(void)
+{
+	snd_soc_unregister_dai(&tlv320aic23_dai);
+}
+module_exit(tlv320aic23_exit);
+
 MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
 MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index bed8a9e63ddc..29f2f1a017fd 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
  * Digital Audio Interface Operations
  */
 static int aic26_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = {
 	},
 	.ops = {
 		.hw_params = aic26_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = aic26_mute,
 		.set_sysclk = aic26_set_sysclk,
 		.set_fmt = aic26_set_fmt,
@@ -360,7 +359,7 @@ static int aic26_probe(struct platform_device *pdev)
 
 	/* CODEC is setup, we can register the card now */
 	dev_dbg(&pdev->dev, "Registering card\n");
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "aic26: failed to register card\n");
 		goto card_err;
@@ -427,7 +426,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
 static int aic26_spi_probe(struct spi_device *spi)
 {
 	struct aic26 *aic26;
-	int rc, i, reg;
+	int ret, i, reg;
 
 	dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
 
@@ -457,6 +456,14 @@ static int aic26_spi_probe(struct spi_device *spi)
 	aic26->codec.reg_cache_size = AIC26_NUM_REGS;
 	aic26->codec.reg_cache = aic26->reg_cache;
 
+	aic26_dai.dev = &spi->dev;
+	ret = snd_soc_register_dai(&aic26_dai);
+	if (ret != 0) {
+		dev_err(&spi->dev, "Failed to register DAI: %d\n", ret);
+		kfree(aic26);
+		return ret;
+	}
+
 	/* Reset the codec to power on defaults */
 	aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
 
@@ -475,8 +482,8 @@ static int aic26_spi_probe(struct spi_device *spi)
 
 	/* Register the sysfs files for debugging */
 	/* Create SysFS files */
-	rc = device_create_file(&spi->dev, &dev_attr_keyclick);
-	if (rc)
+	ret = device_create_file(&spi->dev, &dev_attr_keyclick);
+	if (ret)
 		dev_info(&spi->dev, "error creating sysfs files\n");
 
 #if defined(CONFIG_SND_SOC_OF_SIMPLE)
@@ -493,6 +500,7 @@ static int aic26_spi_remove(struct spi_device *spi)
 {
 	struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
 
+	snd_soc_unregister_dai(&aic26_dai);
 	kfree(aic26);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cff276ee261e..b47a749c5ea2 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -253,11 +253,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 
 	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
 		     DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
-		     0x01, 0),
-	SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     PGAR_2_RLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
+	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
+	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
+		     DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
+		     0, 0x7f, 1),
+	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+		     LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
 		     LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
 
 	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
@@ -272,8 +278,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 		     DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
 		     PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
+		     0, 0x7f, 1),
 	SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
 		     LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
 
@@ -281,8 +291,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 		     DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
 	SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
-		     PGAR_2_HPRCOM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
+		     0, 0x7f, 1),
 	SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
 		     LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
 
@@ -333,7 +345,8 @@ SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
 
 /* Left DAC_L1 Mixer */
 static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0),
@@ -341,7 +354,8 @@ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
 
 /* Right DAC_R1 Mixer */
 static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
@@ -350,14 +364,18 @@ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
 /* Left PGA Mixer */
 static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = {
 	SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1),
 };
 
 /* Right PGA Mixer */
 static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
 	SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1),
 };
 
@@ -379,34 +397,42 @@ SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
 
 /* Left PGA Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0),
 };
 
 /* Right PGA Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
 };
 
 /* Left Line2 Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
 };
 
 /* Right Line2 Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
 };
 
 static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
@@ -439,22 +465,26 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	/* Mono Output */
 	SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
 
-	/* Left Inputs to Left ADC */
+	/* Inputs to Left ADC */
 	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
 	SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
 			   &aic3x_left_pga_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
 	SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_left_line1_mux_controls),
+	SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
+			 &aic3x_left_line1_mux_controls),
 	SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_left_line2_mux_controls),
 
-	/* Right Inputs to Right ADC */
+	/* Inputs to Right ADC */
 	SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
 			 LINE1R_2_RADC_CTRL, 2, 0),
 	SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
 			   &aic3x_right_pga_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
+	SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
+			 &aic3x_right_line1_mux_controls),
 	SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_right_line1_mux_controls),
 	SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
@@ -531,7 +561,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left DAC Mux", "DAC_L2", "Left DAC"},
 	{"Left DAC Mux", "DAC_L3", "Left DAC"},
 
-	{"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"},
+	{"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"},
+	{"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"},
 	{"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"},
 	{"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"},
 	{"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"},
@@ -557,7 +588,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right DAC Mux", "DAC_R2", "Right DAC"},
 	{"Right DAC Mux", "DAC_R3", "Right DAC"},
 
-	{"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"},
+	{"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"},
+	{"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"},
 	{"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"},
 	{"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"},
 	{"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"},
@@ -592,8 +624,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left Line2L Mux", "differential", "LINE2L"},
 
 	{"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
+	{"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
 	{"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
 	{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
+	{"Left PGA Mixer", "Mic3R Switch", "MIC3R"},
 
 	{"Left ADC", NULL, "Left PGA Mixer"},
 	{"Left ADC", NULL, "GPIO1 dmic modclk"},
@@ -605,18 +639,23 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right Line2R Mux", "single-ended", "LINE2R"},
 	{"Right Line2R Mux", "differential", "LINE2R"},
 
+	{"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"},
 	{"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
 	{"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
+	{"Right PGA Mixer", "Mic3L Switch", "MIC3L"},
 	{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
 
 	{"Right ADC", NULL, "Right PGA Mixer"},
 	{"Right ADC", NULL, "GPIO1 dmic modclk"},
 
 	/* Left PGA Bypass */
-	{"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"},
 	{"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"},
-	{"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"},
-	{"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"},
 
 	{"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"},
 	{"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"},
@@ -627,10 +666,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left HP Out", NULL, "Left PGA Bypass Mixer"},
 
 	/* Right PGA Bypass */
-	{"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"},
 	{"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"},
-	{"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"},
-	{"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"},
 
 	{"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"},
 	{"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"},
@@ -643,10 +685,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right HP Out", NULL, "Right PGA Bypass Mixer"},
 
 	/* Left Line2 Bypass */
-	{"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"},
+	{"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"},
+	{"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"},
 	{"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"},
 	{"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"},
-	{"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"},
+	{"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"},
 
 	{"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"},
 	{"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"},
@@ -657,10 +700,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left HP Out", NULL, "Left Line2 Bypass Mixer"},
 
 	/* Right Line2 Bypass */
-	{"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"},
+	{"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"},
+	{"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"},
 	{"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"},
 	{"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"},
-	{"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"},
+	{"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"},
 
 	{"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"},
 	{"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"},
@@ -694,7 +738,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int aic3x_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -846,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 	u8 iface_areg, iface_breg;
+	int delay = 0;
 
 	iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
 	iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
@@ -871,6 +917,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		       SND_SOC_DAIFMT_INV_MASK)) {
 	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
+	case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
+		delay = 1;
 	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
@@ -887,6 +935,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* set iface */
 	aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
 	aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
+	aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
 
 	return 0;
 }
@@ -981,14 +1030,41 @@ int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
 }
 EXPORT_SYMBOL_GPL(aic3x_get_gpio);
 
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+				 int headset_debounce, int button_debounce)
+{
+	u8 val;
+
+	val = ((detect & AIC3X_HEADSET_DETECT_MASK)
+		<< AIC3X_HEADSET_DETECT_SHIFT) |
+	      ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
+		<< AIC3X_HEADSET_DEBOUNCE_SHIFT) |
+	      ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
+		<< AIC3X_BUTTON_DEBOUNCE_SHIFT);
+
+	if (detect & AIC3X_HEADSET_DETECT_MASK)
+		val |= AIC3X_HEADSET_DETECT_ENABLED;
+
+	aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
+}
+EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
+
 int aic3x_headset_detected(struct snd_soc_codec *codec)
 {
 	u8 val;
-	aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
-	return (val >> 2) & 1;
+	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+	return (val >> 4) & 1;
 }
 EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 
+int aic3x_button_pressed(struct snd_soc_codec *codec)
+{
+	u8 val;
+	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+	return (val >> 5) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_button_pressed);
+
 #define AIC3X_RATES	SNDRV_PCM_RATE_8000_96000
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -1009,8 +1085,6 @@ struct snd_soc_dai aic3x_dai = {
 		.formats = AIC3X_FORMATS,},
 	.ops = {
 		.hw_params = aic3x_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = aic3x_mute,
 		.set_sysclk = aic3x_set_dai_sysclk,
 		.set_fmt = aic3x_set_dai_fmt,
@@ -1152,7 +1226,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
 
 	aic3x_add_controls(codec);
 	aic3x_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "aic3x: failed to register card\n");
 		goto card_err;
@@ -1341,6 +1415,18 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
 
+static int __init aic3x_modinit(void)
+{
+	return snd_soc_register_dai(&aic3x_dai);
+}
+module_init(aic3x_modinit);
+
+static void __exit aic3x_exit(void)
+{
+	snd_soc_unregister_dai(&aic3x_dai);
+}
+module_exit(aic3x_exit);
+
 MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 00a195aa02e4..ac827e578c4d 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -35,11 +35,15 @@
 #define AIC3X_ASD_INTF_CTRLA		8
 /* Audio serial data interface control register B */
 #define AIC3X_ASD_INTF_CTRLB		9
+/* Audio serial data interface control register C */
+#define AIC3X_ASD_INTF_CTRLC		10
 /* Audio overflow status and PLL R value programming register */
 #define AIC3X_OVRF_STATUS_AND_PLLR_REG	11
 /* Audio codec digital filter control register */
 #define AIC3X_CODEC_DFILT_CTRL		12
-
+/* Headset/button press detection register */
+#define AIC3X_HEADSET_DETECT_CTRL_A	13
+#define AIC3X_HEADSET_DETECT_CTRL_B	14
 /* ADC PGA Gain control registers */
 #define LADC_VOL			15
 #define RADC_VOL			16
@@ -48,7 +52,9 @@
 #define MIC3LR_2_RADC_CTRL		18
 /* Line1 Input control registers */
 #define LINE1L_2_LADC_CTRL		19
+#define LINE1R_2_LADC_CTRL		21
 #define LINE1R_2_RADC_CTRL		22
+#define LINE1L_2_RADC_CTRL		24
 /* Line2 Input control registers */
 #define LINE2L_2_LADC_CTRL		20
 #define LINE2R_2_RADC_CTRL		23
@@ -79,6 +85,8 @@
 #define LINE2L_2_HPLOUT_VOL		45
 #define LINE2R_2_HPROUT_VOL		62
 #define PGAL_2_HPLOUT_VOL		46
+#define PGAL_2_HPROUT_VOL		60
+#define PGAR_2_HPLOUT_VOL		49
 #define PGAR_2_HPROUT_VOL		63
 #define DACL1_2_HPLOUT_VOL		47
 #define DACR1_2_HPROUT_VOL		64
@@ -88,6 +96,8 @@
 #define LINE2L_2_HPLCOM_VOL		52
 #define LINE2R_2_HPRCOM_VOL		69
 #define PGAL_2_HPLCOM_VOL		53
+#define PGAR_2_HPLCOM_VOL		56
+#define PGAL_2_HPRCOM_VOL		67
 #define PGAR_2_HPRCOM_VOL		70
 #define DACL1_2_HPLCOM_VOL		54
 #define DACR1_2_HPRCOM_VOL		71
@@ -103,11 +113,17 @@
 #define MONOLOPM_CTRL			79
 /* Line Output Plus/Minus control registers */
 #define LINE2L_2_LLOPM_VOL		80
+#define LINE2L_2_RLOPM_VOL		87
+#define LINE2R_2_LLOPM_VOL		83
 #define LINE2R_2_RLOPM_VOL		90
 #define PGAL_2_LLOPM_VOL		81
+#define PGAL_2_RLOPM_VOL		88
+#define PGAR_2_LLOPM_VOL		84
 #define PGAR_2_RLOPM_VOL		91
 #define DACL1_2_LLOPM_VOL		82
+#define DACL1_2_RLOPM_VOL		89
 #define DACR1_2_RLOPM_VOL		92
+#define DACR1_2_LLOPM_VOL		85
 #define LLOPM_CTRL			86
 #define RLOPM_CTRL			93
 /* GPIO/IRQ registers */
@@ -221,7 +237,49 @@ enum {
 
 void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
 int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+
+/* headset detection / button API */
+
+/* The AIC3x supports detection of stereo headsets (GND + left + right signal)
+ * and cellular headsets (GND + speaker output + microphone input).
+ * It is recommended to enable MIC bias for this function to work properly.
+ * For more information, please refer to the datasheet. */
+enum {
+	AIC3X_HEADSET_DETECT_OFF	= 0,
+	AIC3X_HEADSET_DETECT_STEREO	= 1,
+	AIC3X_HEADSET_DETECT_CELLULAR   = 2,
+	AIC3X_HEADSET_DETECT_BOTH	= 3
+};
+
+enum {
+	AIC3X_HEADSET_DEBOUNCE_16MS	= 0,
+	AIC3X_HEADSET_DEBOUNCE_32MS	= 1,
+	AIC3X_HEADSET_DEBOUNCE_64MS	= 2,
+	AIC3X_HEADSET_DEBOUNCE_128MS	= 3,
+	AIC3X_HEADSET_DEBOUNCE_256MS	= 4,
+	AIC3X_HEADSET_DEBOUNCE_512MS	= 5
+};
+
+enum {
+	AIC3X_BUTTON_DEBOUNCE_0MS	= 0,
+	AIC3X_BUTTON_DEBOUNCE_8MS	= 1,
+	AIC3X_BUTTON_DEBOUNCE_16MS	= 2,
+	AIC3X_BUTTON_DEBOUNCE_32MS	= 3
+};
+
+#define AIC3X_HEADSET_DETECT_ENABLED	0x80
+#define AIC3X_HEADSET_DETECT_SHIFT	5
+#define AIC3X_HEADSET_DETECT_MASK	3
+#define AIC3X_HEADSET_DEBOUNCE_SHIFT	2
+#define AIC3X_HEADSET_DEBOUNCE_MASK	7
+#define AIC3X_BUTTON_DEBOUNCE_SHIFT 	0
+#define AIC3X_BUTTON_DEBOUNCE_MASK	3
+
+/* see the enums above for valid parameters to this function */
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+				 int headset_debounce, int button_debounce);
 int aic3x_headset_detected(struct snd_soc_codec *codec);
+int aic3x_button_pressed(struct snd_soc_codec *codec);
 
 struct aic3x_setup_data {
 	int i2c_bus;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
new file mode 100644
index 000000000000..51848880504a
--- /dev/null
+++ b/sound/soc/codecs/twl4030.c
@@ -0,0 +1,1317 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author:      Steve Sakoman, <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/i2c/twl4030.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "twl4030.h"
+
+/*
+ * twl4030 register cache & default register settings
+ */
+static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
+	0x00, /* this register not used		*/
+	0x93, /* REG_CODEC_MODE		(0x1)	*/
+	0xc3, /* REG_OPTION		(0x2)	*/
+	0x00, /* REG_UNKNOWN		(0x3)	*/
+	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
+	0x20, /* REG_ANAMICL		(0x5)	*/
+	0x00, /* REG_ANAMICR		(0x6)	*/
+	0x00, /* REG_AVADC_CTL		(0x7)	*/
+	0x00, /* REG_ADCMICSEL		(0x8)	*/
+	0x00, /* REG_DIGMIXING		(0x9)	*/
+	0x0c, /* REG_ATXL1PGA		(0xA)	*/
+	0x0c, /* REG_ATXR1PGA		(0xB)	*/
+	0x00, /* REG_AVTXL2PGA		(0xC)	*/
+	0x00, /* REG_AVTXR2PGA		(0xD)	*/
+	0x01, /* REG_AUDIO_IF		(0xE)	*/
+	0x00, /* REG_VOICE_IF		(0xF)	*/
+	0x00, /* REG_ARXR1PGA		(0x10)	*/
+	0x00, /* REG_ARXL1PGA		(0x11)	*/
+	0x6c, /* REG_ARXR2PGA		(0x12)	*/
+	0x6c, /* REG_ARXL2PGA		(0x13)	*/
+	0x00, /* REG_VRXPGA		(0x14)	*/
+	0x00, /* REG_VSTPGA		(0x15)	*/
+	0x00, /* REG_VRX2ARXPGA		(0x16)	*/
+	0x0c, /* REG_AVDAC_CTL		(0x17)	*/
+	0x00, /* REG_ARX2VTXPGA		(0x18)	*/
+	0x00, /* REG_ARXL1_APGA_CTL	(0x19)	*/
+	0x00, /* REG_ARXR1_APGA_CTL	(0x1A)	*/
+	0x4b, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
+	0x4b, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
+	0x00, /* REG_ATX2ARXPGA		(0x1D)	*/
+	0x00, /* REG_BT_IF		(0x1E)	*/
+	0x00, /* REG_BTPGA		(0x1F)	*/
+	0x00, /* REG_BTSTPGA		(0x20)	*/
+	0x00, /* REG_EAR_CTL		(0x21)	*/
+	0x24, /* REG_HS_SEL		(0x22)	*/
+	0x0a, /* REG_HS_GAIN_SET	(0x23)	*/
+	0x00, /* REG_HS_POPN_SET	(0x24)	*/
+	0x00, /* REG_PREDL_CTL		(0x25)	*/
+	0x00, /* REG_PREDR_CTL		(0x26)	*/
+	0x00, /* REG_PRECKL_CTL		(0x27)	*/
+	0x00, /* REG_PRECKR_CTL		(0x28)	*/
+	0x00, /* REG_HFL_CTL		(0x29)	*/
+	0x00, /* REG_HFR_CTL		(0x2A)	*/
+	0x00, /* REG_ALC_CTL		(0x2B)	*/
+	0x00, /* REG_ALC_SET1		(0x2C)	*/
+	0x00, /* REG_ALC_SET2		(0x2D)	*/
+	0x00, /* REG_BOOST_CTL		(0x2E)	*/
+	0x00, /* REG_SOFTVOL_CTL	(0x2F)	*/
+	0x00, /* REG_DTMF_FREQSEL	(0x30)	*/
+	0x00, /* REG_DTMF_TONEXT1H	(0x31)	*/
+	0x00, /* REG_DTMF_TONEXT1L	(0x32)	*/
+	0x00, /* REG_DTMF_TONEXT2H	(0x33)	*/
+	0x00, /* REG_DTMF_TONEXT2L	(0x34)	*/
+	0x00, /* REG_DTMF_TONOFF	(0x35)	*/
+	0x00, /* REG_DTMF_WANONOFF	(0x36)	*/
+	0x00, /* REG_I2S_RX_SCRAMBLE_H	(0x37)	*/
+	0x00, /* REG_I2S_RX_SCRAMBLE_M	(0x38)	*/
+	0x00, /* REG_I2S_RX_SCRAMBLE_L	(0x39)	*/
+	0x16, /* REG_APLL_CTL		(0x3A)	*/
+	0x00, /* REG_DTMF_CTL		(0x3B)	*/
+	0x00, /* REG_DTMF_PGA_CTL2	(0x3C)	*/
+	0x00, /* REG_DTMF_PGA_CTL1	(0x3D)	*/
+	0x00, /* REG_MISC_SET_1		(0x3E)	*/
+	0x00, /* REG_PCMBTMUX		(0x3F)	*/
+	0x00, /* not used		(0x40)	*/
+	0x00, /* not used		(0x41)	*/
+	0x00, /* not used		(0x42)	*/
+	0x00, /* REG_RX_PATH_SEL	(0x43)	*/
+	0x00, /* REG_VDL_APGA_CTL	(0x44)	*/
+	0x00, /* REG_VIBRA_CTL		(0x45)	*/
+	0x00, /* REG_VIBRA_SET		(0x46)	*/
+	0x00, /* REG_VIBRA_PWM_SET	(0x47)	*/
+	0x00, /* REG_ANAMIC_GAIN	(0x48)	*/
+	0x00, /* REG_MISC_SET_2		(0x49)	*/
+};
+
+/*
+ * read twl4030 register cache
+ */
+static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+
+	return cache[reg];
+}
+
+/*
+ * write twl4030 register cache
+ */
+static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
+						u8 reg, u8 value)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= TWL4030_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the twl4030 register space
+ */
+static int twl4030_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value)
+{
+	twl4030_write_reg_cache(codec, reg, value);
+	return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+}
+
+static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+{
+	u8 mode;
+
+	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+		mode & ~TWL4030_CODECPDZ);
+
+	/* REVISIT: this delay is present in TI sample drivers */
+	/* but there seems to be no TRM requirement for it     */
+	udelay(10);
+}
+
+static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
+{
+	u8 mode;
+
+	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+		mode | TWL4030_CODECPDZ);
+
+	/* REVISIT: this delay is present in TI sample drivers */
+	/* but there seems to be no TRM requirement for it     */
+	udelay(10);
+}
+
+static void twl4030_init_chip(struct snd_soc_codec *codec)
+{
+	int i;
+
+	/* clear CODECPDZ prior to setting register defaults */
+	twl4030_clear_codecpdz(codec);
+
+	/* set all audio section registers to reasonable defaults */
+	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
+		twl4030_write(codec, i,	twl4030_reg[i]);
+
+}
+
+/* Earpiece */
+static const char *twl4030_earpiece_texts[] =
+		{"Off", "DACL1", "DACL2", "Invalid", "DACR1"};
+
+static const struct soc_enum twl4030_earpiece_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1,
+			ARRAY_SIZE(twl4030_earpiece_texts),
+			twl4030_earpiece_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
+SOC_DAPM_ENUM("Route", twl4030_earpiece_enum);
+
+/* PreDrive Left */
+static const char *twl4030_predrivel_texts[] =
+		{"Off", "DACL1", "DACL2", "Invalid", "DACR2"};
+
+static const struct soc_enum twl4030_predrivel_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1,
+			ARRAY_SIZE(twl4030_predrivel_texts),
+			twl4030_predrivel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
+SOC_DAPM_ENUM("Route", twl4030_predrivel_enum);
+
+/* PreDrive Right */
+static const char *twl4030_predriver_texts[] =
+		{"Off", "DACR1", "DACR2", "Invalid", "DACL2"};
+
+static const struct soc_enum twl4030_predriver_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1,
+			ARRAY_SIZE(twl4030_predriver_texts),
+			twl4030_predriver_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
+SOC_DAPM_ENUM("Route", twl4030_predriver_enum);
+
+/* Headset Left */
+static const char *twl4030_hsol_texts[] =
+		{"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_hsol_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
+			ARRAY_SIZE(twl4030_hsol_texts),
+			twl4030_hsol_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
+SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+
+/* Headset Right */
+static const char *twl4030_hsor_texts[] =
+		{"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_hsor_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
+			ARRAY_SIZE(twl4030_hsor_texts),
+			twl4030_hsor_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
+SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+
+/* Carkit Left */
+static const char *twl4030_carkitl_texts[] =
+		{"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_carkitl_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
+			ARRAY_SIZE(twl4030_carkitl_texts),
+			twl4030_carkitl_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+
+/* Carkit Right */
+static const char *twl4030_carkitr_texts[] =
+		{"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_carkitr_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
+			ARRAY_SIZE(twl4030_carkitr_texts),
+			twl4030_carkitr_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+
+/* Handsfree Left */
+static const char *twl4030_handsfreel_texts[] =
+		{"Voice", "DACL1", "DACL2", "DACR2"};
+
+static const struct soc_enum twl4030_handsfreel_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
+			ARRAY_SIZE(twl4030_handsfreel_texts),
+			twl4030_handsfreel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+
+/* Handsfree Right */
+static const char *twl4030_handsfreer_texts[] =
+		{"Voice", "DACR1", "DACR2", "DACL2"};
+
+static const struct soc_enum twl4030_handsfreer_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
+			ARRAY_SIZE(twl4030_handsfreer_texts),
+			twl4030_handsfreer_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	int ret = 0;
+	int val;
+
+	switch (e->reg) {
+	case TWL4030_REG_PREDL_CTL:
+	case TWL4030_REG_PREDR_CTL:
+	case TWL4030_REG_EAR_CTL:
+		val = w->value >> e->shift_l;
+		if (val == 3) {
+			printk(KERN_WARNING
+			"Invalid MUX setting for register 0x%02x (%d)\n",
+			      e->reg, val);
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+static int handsfree_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
+	unsigned char hs_ctl;
+
+	hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+
+	if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+		hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
+		twl4030_write(w->codec, e->reg, hs_ctl);
+		hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
+		twl4030_write(w->codec, e->reg, hs_ctl);
+		hs_ctl |= TWL4030_HF_CTL_HB_EN;
+		twl4030_write(w->codec, e->reg, hs_ctl);
+	} else {
+		hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
+				| TWL4030_HF_CTL_HB_EN);
+		twl4030_write(w->codec, e->reg, hs_ctl);
+	}
+
+	return 0;
+}
+
+/*
+ * Some of the gain controls in TWL (mostly those which are associated with
+ * the outputs) are implemented in an interesting way:
+ * 0x0 : Power down (mute)
+ * 0x1 : 6dB
+ * 0x2 : 0 dB
+ * 0x3 : -6 dB
+ * Inverting not going to help with these.
+ * Custom volsw and volsw_2r get/put functions to handle these gain bits.
+ */
+#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\
+			       xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw_twl4030, \
+	.put = snd_soc_put_volsw_twl4030, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
+		 .max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\
+				 xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r, \
+	.get = snd_soc_get_volsw_r2_twl4030,\
+	.put = snd_soc_put_volsw_r2_twl4030, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		 .rshift = xshift, .max = xmax, .invert = xinvert} }
+#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
+	SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
+			       xinvert, tlv_array)
+
+static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int rshift = mc->rshift;
+	int max = mc->max;
+	int mask = (1 << fls(max)) - 1;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	if (ucontrol->value.integer.value[0])
+		ucontrol->value.integer.value[0] =
+			max + 1 - ucontrol->value.integer.value[0];
+
+	if (shift != rshift) {
+		ucontrol->value.integer.value[1] =
+			(snd_soc_read(codec, reg) >> rshift) & mask;
+		if (ucontrol->value.integer.value[1])
+			ucontrol->value.integer.value[1] =
+				max + 1 - ucontrol->value.integer.value[1];
+	}
+
+	return 0;
+}
+
+static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int rshift = mc->rshift;
+	int max = mc->max;
+	int mask = (1 << fls(max)) - 1;
+	unsigned short val, val2, val_mask;
+
+	val = (ucontrol->value.integer.value[0] & mask);
+
+	val_mask = mask << shift;
+	if (val)
+		val = max + 1 - val;
+	val = val << shift;
+	if (shift != rshift) {
+		val2 = (ucontrol->value.integer.value[1] & mask);
+		val_mask |= mask << rshift;
+		if (val2)
+			val2 = max + 1 - val2;
+		val |= val2 << rshift;
+	}
+	return snd_soc_update_bits(codec, reg, val_mask, val);
+}
+
+static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	int mask = (1<<fls(max))-1;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	ucontrol->value.integer.value[1] =
+		(snd_soc_read(codec, reg2) >> shift) & mask;
+
+	if (ucontrol->value.integer.value[0])
+		ucontrol->value.integer.value[0] =
+			max + 1 - ucontrol->value.integer.value[0];
+	if (ucontrol->value.integer.value[1])
+		ucontrol->value.integer.value[1] =
+			max + 1 - ucontrol->value.integer.value[1];
+
+	return 0;
+}
+
+static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	int mask = (1 << fls(max)) - 1;
+	int err;
+	unsigned short val, val2, val_mask;
+
+	val_mask = mask << shift;
+	val = (ucontrol->value.integer.value[0] & mask);
+	val2 = (ucontrol->value.integer.value[1] & mask);
+
+	if (val)
+		val = max + 1 - val;
+	if (val2)
+		val2 = max + 1 - val2;
+
+	val = val << shift;
+	val2 = val2 << shift;
+
+	err = snd_soc_update_bits(codec, reg, val_mask, val);
+	if (err < 0)
+		return err;
+
+	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+	return err;
+}
+
+static int twl4030_get_left_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	int result = 0;
+
+	/* one bit must be set a time */
+	reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
+			| TWL4030_MAINMIC_EN;
+	if (reg != 0) {
+		result++;
+		while ((reg & 1) == 0) {
+			result++;
+			reg >>= 1;
+		}
+	}
+
+	ucontrol->value.integer.value[0] = result;
+	return 0;
+}
+
+static int twl4030_put_left_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	int value = ucontrol->value.integer.value[0];
+	u8 anamicl, micbias, avadc_ctl;
+
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
+			| TWL4030_MAINMIC_EN);
+	micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
+	micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN);
+	avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
+
+	switch (value) {
+	case 1:
+		anamicl |= TWL4030_MAINMIC_EN;
+		micbias |= TWL4030_MICBIAS1_EN;
+		break;
+	case 2:
+		anamicl |= TWL4030_HSMIC_EN;
+		micbias |= TWL4030_HSMICBIAS_EN;
+		break;
+	case 3:
+		anamicl |= TWL4030_AUXL_EN;
+		break;
+	case 4:
+		anamicl |= TWL4030_CKMIC_EN;
+		break;
+	default:
+		break;
+	}
+
+	/* If some input is selected, enable amp and ADC */
+	if (value != 0) {
+		anamicl |= TWL4030_MICAMPL_EN;
+		avadc_ctl |= TWL4030_ADCL_EN;
+	} else {
+		anamicl &= ~TWL4030_MICAMPL_EN;
+		avadc_ctl &= ~TWL4030_ADCL_EN;
+	}
+
+	twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl);
+	twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
+	twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
+
+	return 1;
+}
+
+static int twl4030_get_right_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
+	int value = 0;
+
+	reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN;
+	switch (reg) {
+	case TWL4030_SUBMIC_EN:
+		value = 1;
+		break;
+	case TWL4030_AUXR_EN:
+		value = 2;
+		break;
+	default:
+		break;
+	}
+
+	ucontrol->value.integer.value[0] = value;
+	return 0;
+}
+
+static int twl4030_put_right_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	int value = ucontrol->value.integer.value[0];
+	u8 anamicr, micbias, avadc_ctl;
+
+	anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
+	anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN);
+	micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
+	micbias &= ~TWL4030_MICBIAS2_EN;
+	avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
+
+	switch (value) {
+	case 1:
+		anamicr |= TWL4030_SUBMIC_EN;
+		micbias |= TWL4030_MICBIAS2_EN;
+		break;
+	case 2:
+		anamicr |= TWL4030_AUXR_EN;
+		break;
+	default:
+		break;
+	}
+
+	if (value != 0) {
+		anamicr |= TWL4030_MICAMPR_EN;
+		avadc_ctl |= TWL4030_ADCR_EN;
+	} else {
+		anamicr &= ~TWL4030_MICAMPR_EN;
+		avadc_ctl &= ~TWL4030_ADCR_EN;
+	}
+
+	twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr);
+	twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
+	twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
+
+	return 1;
+}
+
+static const char *twl4030_left_in_sel[] = {
+	"None",
+	"Main Mic",
+	"Headset Mic",
+	"Line In",
+	"Carkit Mic",
+};
+
+static const char *twl4030_right_in_sel[] = {
+	"None",
+	"Sub Mic",
+	"Line In",
+};
+
+static const struct soc_enum twl4030_left_input_mux =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel),
+		twl4030_left_in_sel);
+
+static const struct soc_enum twl4030_right_input_mux =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel),
+		twl4030_right_in_sel);
+
+/*
+ * FGAIN volume control:
+ * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
+
+/*
+ * CGAIN volume control:
+ * 0 dB to 12 dB in 6 dB steps
+ * value 2 and 3 means 12 dB
+ */
+static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
+
+/*
+ * Analog playback gain
+ * -24 dB to 12 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
+
+/*
+ * Gain controls tied to outputs
+ * -6 dB to 6 dB in 6 dB steps (mute instead of -12)
+ */
+static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
+
+/*
+ * Capture gain after the ADCs
+ * from 0 dB to 31 dB in 1 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
+
+/*
+ * Gain control for input amplifiers
+ * 0 dB to 30 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+	/* Common playback gain controls */
+	SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
+		TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+		0, 0x3f, 0, digital_fine_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume",
+		TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+		0, 0x3f, 0, digital_fine_tlv),
+
+	SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume",
+		TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+		6, 0x2, 0, digital_coarse_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume",
+		TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+		6, 0x2, 0, digital_coarse_tlv),
+
+	SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume",
+		TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+		3, 0x12, 1, analog_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume",
+		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+		3, 0x12, 1, analog_tlv),
+	SOC_DOUBLE_R("DAC1 Analog Playback Switch",
+		TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+		1, 1, 0),
+	SOC_DOUBLE_R("DAC2 Analog Playback Switch",
+		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+		1, 1, 0),
+
+	/* Separate output gain controls */
+	SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
+		TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
+		4, 3, 0, output_tvl),
+
+	SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume",
+		TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl),
+
+	SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume",
+		TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL,
+		4, 3, 0, output_tvl),
+
+	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+
+	/* Common capture gain controls */
+	SOC_DOUBLE_R_TLV("Capture Volume",
+		TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
+		0, 0x1f, 0, digital_capture_tlv),
+
+	SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN,
+		0, 3, 5, 0, input_gain_tlv),
+
+	/* Input source controls */
+	SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux,
+		twl4030_get_left_input, twl4030_put_left_input),
+	SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux,
+		twl4030_get_right_input, twl4030_put_right_input),
+};
+
+/* add non dapm controls */
+static int twl4030_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&twl4030_snd_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("INL"),
+	SND_SOC_DAPM_INPUT("INR"),
+
+	SND_SOC_DAPM_OUTPUT("OUTL"),
+	SND_SOC_DAPM_OUTPUT("OUTR"),
+	SND_SOC_DAPM_OUTPUT("EARPIECE"),
+	SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
+	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
+	SND_SOC_DAPM_OUTPUT("HSOL"),
+	SND_SOC_DAPM_OUTPUT("HSOR"),
+	SND_SOC_DAPM_OUTPUT("CARKITL"),
+	SND_SOC_DAPM_OUTPUT("CARKITR"),
+	SND_SOC_DAPM_OUTPUT("HFL"),
+	SND_SOC_DAPM_OUTPUT("HFR"),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+			TWL4030_REG_AVDAC_CTL, 0, 0),
+	SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+			TWL4030_REG_AVDAC_CTL, 1, 0),
+	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+			TWL4030_REG_AVDAC_CTL, 2, 0),
+	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+			TWL4030_REG_AVDAC_CTL, 3, 0),
+
+	/* Analog PGAs */
+	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
+			0, 0, NULL, 0),
+
+	/* Output MUX controls */
+	/* Earpiece */
+	SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_earpiece_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	/* PreDrivL/R */
+	SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_predrivel_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_predriver_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	/* HeadsetL/R */
+	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsol_control),
+	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsor_control),
+	/* CarkitL/R */
+	SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_carkitl_control),
+	SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_carkitr_control),
+	/* HandsfreeL/R */
+	SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
+		&twl4030_dapm_handsfreel_control, handsfree_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
+		&twl4030_dapm_handsfreer_control, handsfree_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"ARXL1_APGA", NULL, "DAC Left1"},
+	{"ARXR1_APGA", NULL, "DAC Right1"},
+	{"ARXL2_APGA", NULL, "DAC Left2"},
+	{"ARXR2_APGA", NULL, "DAC Right2"},
+
+	/* Internal playback routings */
+	/* Earpiece */
+	{"Earpiece Mux", "DACL1", "ARXL1_APGA"},
+	{"Earpiece Mux", "DACL2", "ARXL2_APGA"},
+	{"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+	/* PreDrivL */
+	{"PredriveL Mux", "DACL1", "ARXL1_APGA"},
+	{"PredriveL Mux", "DACL2", "ARXL2_APGA"},
+	{"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+	/* PreDrivR */
+	{"PredriveR Mux", "DACR1", "ARXR1_APGA"},
+	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
+	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+	/* HeadsetL */
+	{"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
+	{"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+	/* HeadsetR */
+	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
+	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+	/* CarkitL */
+	{"CarkitL Mux", "DACL1", "ARXL1_APGA"},
+	{"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+	/* CarkitR */
+	{"CarkitR Mux", "DACR1", "ARXR1_APGA"},
+	{"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+	/* HandsfreeL */
+	{"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
+	{"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
+	{"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+	/* HandsfreeR */
+	{"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
+	{"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
+	{"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+
+	/* outputs */
+	{"OUTL", NULL, "ARXL2_APGA"},
+	{"OUTR", NULL, "ARXR2_APGA"},
+	{"EARPIECE", NULL, "Earpiece Mux"},
+	{"PREDRIVEL", NULL, "PredriveL Mux"},
+	{"PREDRIVER", NULL, "PredriveR Mux"},
+	{"HSOL", NULL, "HeadsetL Mux"},
+	{"HSOR", NULL, "HeadsetR Mux"},
+	{"CARKITL", NULL, "CarkitL Mux"},
+	{"CARKITR", NULL, "CarkitR Mux"},
+	{"HFL", NULL, "HandsfreeL Mux"},
+	{"HFR", NULL, "HandsfreeR Mux"},
+
+	/* inputs */
+	{"ADCL", NULL, "INL"},
+	{"ADCR", NULL, "INR"},
+};
+
+static int twl4030_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
+				 ARRAY_SIZE(twl4030_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+	u8 anamicl, regmisc1, byte, popn;
+	int i = 0;
+
+	/* set CODECPDZ to turn on codec */
+	twl4030_set_codecpdz(codec);
+
+	/* initiate offset cancellation */
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	twl4030_write(codec, TWL4030_REG_ANAMICL,
+		anamicl | TWL4030_CNCL_OFFSET_START);
+
+	/* wait for offset cancellation to complete */
+	do {
+		/* this takes a little while, so don't slam i2c */
+		udelay(2000);
+		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+				    TWL4030_REG_ANAMICL);
+	} while ((i++ < 100) &&
+		 ((byte & TWL4030_CNCL_OFFSET_START) ==
+		  TWL4030_CNCL_OFFSET_START));
+
+	/* anti-pop when changing analog gain */
+	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
+
+	/* toggle CODECPDZ as per TRM */
+	twl4030_clear_codecpdz(codec);
+	twl4030_set_codecpdz(codec);
+
+	/* program anti-pop with bias ramp delay */
+	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	popn &= TWL4030_RAMP_DELAY;
+	popn |=	TWL4030_RAMP_DELAY_645MS;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+	popn |=	TWL4030_VMID_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+	/* enable anti-pop ramp */
+	popn |= TWL4030_RAMP_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+}
+
+static void twl4030_power_down(struct snd_soc_codec *codec)
+{
+	u8 popn;
+
+	/* disable anti-pop ramp */
+	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	popn &= ~TWL4030_RAMP_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+	/* disable bias out */
+	popn &= ~TWL4030_VMID_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+	/* power down */
+	twl4030_clear_codecpdz(codec);
+}
+
+static int twl4030_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		twl4030_power_up(codec);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* TODO: develop a twl4030_prepare function */
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* TODO: develop a twl4030_standby function */
+		twl4030_power_down(codec);
+		break;
+	case SND_SOC_BIAS_OFF:
+		twl4030_power_down(codec);
+		break;
+	}
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static int twl4030_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u8 mode, old_mode, format, old_format;
+
+
+	/* bit rate */
+	old_mode = twl4030_read_reg_cache(codec,
+			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
+	mode = old_mode & ~TWL4030_APLL_RATE;
+
+	switch (params_rate(params)) {
+	case 8000:
+		mode |= TWL4030_APLL_RATE_8000;
+		break;
+	case 11025:
+		mode |= TWL4030_APLL_RATE_11025;
+		break;
+	case 12000:
+		mode |= TWL4030_APLL_RATE_12000;
+		break;
+	case 16000:
+		mode |= TWL4030_APLL_RATE_16000;
+		break;
+	case 22050:
+		mode |= TWL4030_APLL_RATE_22050;
+		break;
+	case 24000:
+		mode |= TWL4030_APLL_RATE_24000;
+		break;
+	case 32000:
+		mode |= TWL4030_APLL_RATE_32000;
+		break;
+	case 44100:
+		mode |= TWL4030_APLL_RATE_44100;
+		break;
+	case 48000:
+		mode |= TWL4030_APLL_RATE_48000;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	if (mode != old_mode) {
+		/* change rate and set CODECPDZ */
+		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+		twl4030_set_codecpdz(codec);
+	}
+
+	/* sample size */
+	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	format = old_format;
+	format &= ~TWL4030_DATA_WIDTH;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		format |= TWL4030_DATA_WIDTH_16S_16W;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		format |= TWL4030_DATA_WIDTH_32S_24W;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 hw params: unknown format %d\n",
+			params_format(params));
+		return -EINVAL;
+	}
+
+	if (format != old_format) {
+
+		/* clear CODECPDZ before changing format (codec requirement) */
+		twl4030_clear_codecpdz(codec);
+
+		/* change format */
+		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+		/* set CODECPDZ afterwards */
+		twl4030_set_codecpdz(codec);
+	}
+	return 0;
+}
+
+static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 infreq;
+
+	switch (freq) {
+	case 19200000:
+		infreq = TWL4030_APLL_INFREQ_19200KHZ;
+		break;
+	case 26000000:
+		infreq = TWL4030_APLL_INFREQ_26000KHZ;
+		break;
+	case 38400000:
+		infreq = TWL4030_APLL_INFREQ_38400KHZ;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
+			freq);
+		return -EINVAL;
+	}
+
+	infreq |= TWL4030_APLL_EN;
+	twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+	return 0;
+}
+
+static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			     unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 old_format, format;
+
+	/* get format */
+	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	format = old_format;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		format &= ~(TWL4030_AIF_SLAVE_EN);
+		format &= ~(TWL4030_CLK256FS_EN);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		format |= TWL4030_AIF_SLAVE_EN;
+		format |= TWL4030_CLK256FS_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	format &= ~TWL4030_AIF_FORMAT;
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		format |= TWL4030_AIF_FORMAT_CODEC;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (format != old_format) {
+
+		/* clear CODECPDZ before changing format (codec requirement) */
+		twl4030_clear_codecpdz(codec);
+
+		/* change format */
+		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+		/* set CODECPDZ afterwards */
+		twl4030_set_codecpdz(codec);
+	}
+
+	return 0;
+}
+
+#define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
+#define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_dai twl4030_dai = {
+	.name = "twl4030",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = TWL4030_RATES,
+		.formats = TWL4030_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = TWL4030_RATES,
+		.formats = TWL4030_FORMATS,},
+	.ops = {
+		.hw_params = twl4030_hw_params,
+		.set_sysclk = twl4030_set_dai_sysclk,
+		.set_fmt = twl4030_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL_GPL(twl4030_dai);
+
+static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int twl4030_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	twl4030_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+/*
+ * initialize the driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+
+static int twl4030_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	printk(KERN_INFO "TWL4030 Audio Codec init \n");
+
+	codec->name = "twl4030";
+	codec->owner = THIS_MODULE;
+	codec->read = twl4030_read_reg_cache;
+	codec->write = twl4030_write;
+	codec->set_bias_level = twl4030_set_bias_level;
+	codec->dai = &twl4030_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = sizeof(twl4030_reg);
+	codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
+					GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "twl4030: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	twl4030_init_chip(codec);
+
+	/* power on device */
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	twl4030_add_controls(codec);
+	twl4030_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "twl4030: failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+
+static struct snd_soc_device *twl4030_socdev;
+
+static int twl4030_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	twl4030_socdev = socdev;
+	twl4030_init(socdev);
+
+	return 0;
+}
+
+static int twl4030_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_twl4030 = {
+	.probe = twl4030_probe,
+	.remove = twl4030_remove,
+	.suspend = twl4030_suspend,
+	.resume = twl4030_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
+
+static int __init twl4030_modinit(void)
+{
+	return snd_soc_register_dai(&twl4030_dai);
+}
+module_init(twl4030_modinit);
+
+static void __exit twl4030_exit(void)
+{
+	snd_soc_unregister_dai(&twl4030_dai);
+}
+module_exit(twl4030_exit);
+
+MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
+MODULE_AUTHOR("Steve Sakoman");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
new file mode 100644
index 000000000000..54615c76802b
--- /dev/null
+++ b/sound/soc/codecs/twl4030.h
@@ -0,0 +1,219 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TWL4030_AUDIO_H__
+#define __TWL4030_AUDIO_H__
+
+#define TWL4030_REG_CODEC_MODE		0x1
+#define TWL4030_REG_OPTION		0x2
+#define TWL4030_REG_UNKNOWN		0x3
+#define TWL4030_REG_MICBIAS_CTL		0x4
+#define TWL4030_REG_ANAMICL		0x5
+#define TWL4030_REG_ANAMICR		0x6
+#define TWL4030_REG_AVADC_CTL		0x7
+#define TWL4030_REG_ADCMICSEL		0x8
+#define TWL4030_REG_DIGMIXING		0x9
+#define TWL4030_REG_ATXL1PGA		0xA
+#define TWL4030_REG_ATXR1PGA		0xB
+#define TWL4030_REG_AVTXL2PGA		0xC
+#define TWL4030_REG_AVTXR2PGA		0xD
+#define TWL4030_REG_AUDIO_IF		0xE
+#define TWL4030_REG_VOICE_IF		0xF
+#define TWL4030_REG_ARXR1PGA		0x10
+#define TWL4030_REG_ARXL1PGA		0x11
+#define TWL4030_REG_ARXR2PGA		0x12
+#define TWL4030_REG_ARXL2PGA		0x13
+#define TWL4030_REG_VRXPGA		0x14
+#define TWL4030_REG_VSTPGA		0x15
+#define TWL4030_REG_VRX2ARXPGA		0x16
+#define TWL4030_REG_AVDAC_CTL		0x17
+#define TWL4030_REG_ARX2VTXPGA		0x18
+#define TWL4030_REG_ARXL1_APGA_CTL	0x19
+#define TWL4030_REG_ARXR1_APGA_CTL	0x1A
+#define TWL4030_REG_ARXL2_APGA_CTL	0x1B
+#define TWL4030_REG_ARXR2_APGA_CTL	0x1C
+#define TWL4030_REG_ATX2ARXPGA		0x1D
+#define TWL4030_REG_BT_IF		0x1E
+#define TWL4030_REG_BTPGA		0x1F
+#define TWL4030_REG_BTSTPGA		0x20
+#define TWL4030_REG_EAR_CTL		0x21
+#define TWL4030_REG_HS_SEL		0x22
+#define TWL4030_REG_HS_GAIN_SET		0x23
+#define TWL4030_REG_HS_POPN_SET		0x24
+#define TWL4030_REG_PREDL_CTL		0x25
+#define TWL4030_REG_PREDR_CTL		0x26
+#define TWL4030_REG_PRECKL_CTL		0x27
+#define TWL4030_REG_PRECKR_CTL		0x28
+#define TWL4030_REG_HFL_CTL		0x29
+#define TWL4030_REG_HFR_CTL		0x2A
+#define TWL4030_REG_ALC_CTL		0x2B
+#define TWL4030_REG_ALC_SET1		0x2C
+#define TWL4030_REG_ALC_SET2		0x2D
+#define TWL4030_REG_BOOST_CTL		0x2E
+#define TWL4030_REG_SOFTVOL_CTL		0x2F
+#define TWL4030_REG_DTMF_FREQSEL	0x30
+#define TWL4030_REG_DTMF_TONEXT1H	0x31
+#define TWL4030_REG_DTMF_TONEXT1L	0x32
+#define TWL4030_REG_DTMF_TONEXT2H	0x33
+#define TWL4030_REG_DTMF_TONEXT2L	0x34
+#define TWL4030_REG_DTMF_TONOFF		0x35
+#define TWL4030_REG_DTMF_WANONOFF	0x36
+#define TWL4030_REG_I2S_RX_SCRAMBLE_H	0x37
+#define TWL4030_REG_I2S_RX_SCRAMBLE_M	0x38
+#define TWL4030_REG_I2S_RX_SCRAMBLE_L	0x39
+#define TWL4030_REG_APLL_CTL		0x3A
+#define TWL4030_REG_DTMF_CTL		0x3B
+#define TWL4030_REG_DTMF_PGA_CTL2	0x3C
+#define TWL4030_REG_DTMF_PGA_CTL1	0x3D
+#define TWL4030_REG_MISC_SET_1		0x3E
+#define TWL4030_REG_PCMBTMUX		0x3F
+#define TWL4030_REG_RX_PATH_SEL		0x43
+#define TWL4030_REG_VDL_APGA_CTL	0x44
+#define TWL4030_REG_VIBRA_CTL		0x45
+#define TWL4030_REG_VIBRA_SET		0x46
+#define TWL4030_REG_VIBRA_PWM_SET	0x47
+#define TWL4030_REG_ANAMIC_GAIN		0x48
+#define TWL4030_REG_MISC_SET_2		0x49
+
+#define TWL4030_CACHEREGNUM	(TWL4030_REG_MISC_SET_2 + 1)
+
+/* Bitfield Definitions */
+
+/* TWL4030_CODEC_MODE (0x01) Fields */
+
+#define TWL4030_APLL_RATE		0xF0
+#define TWL4030_APLL_RATE_8000		0x00
+#define TWL4030_APLL_RATE_11025		0x10
+#define TWL4030_APLL_RATE_12000		0x20
+#define TWL4030_APLL_RATE_16000		0x40
+#define TWL4030_APLL_RATE_22050		0x50
+#define TWL4030_APLL_RATE_24000		0x60
+#define TWL4030_APLL_RATE_32000		0x80
+#define TWL4030_APLL_RATE_44100		0x90
+#define TWL4030_APLL_RATE_48000		0xA0
+#define TWL4030_SEL_16K			0x04
+#define TWL4030_CODECPDZ		0x02
+#define TWL4030_OPT_MODE		0x01
+
+/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
+
+#define TWL4030_MICBIAS2_CTL		0x40
+#define TWL4030_MICBIAS1_CTL		0x20
+#define TWL4030_HSMICBIAS_EN		0x04
+#define TWL4030_MICBIAS2_EN		0x02
+#define TWL4030_MICBIAS1_EN		0x01
+
+/* ANAMICL (0x05) Fields */
+
+#define TWL4030_CNCL_OFFSET_START	0x80
+#define TWL4030_OFFSET_CNCL_SEL		0x60
+#define TWL4030_OFFSET_CNCL_SEL_ARX1	0x00
+#define TWL4030_OFFSET_CNCL_SEL_ARX2	0x20
+#define TWL4030_OFFSET_CNCL_SEL_VRX	0x40
+#define TWL4030_OFFSET_CNCL_SEL_ALL	0x60
+#define TWL4030_MICAMPL_EN		0x10
+#define TWL4030_CKMIC_EN		0x08
+#define TWL4030_AUXL_EN			0x04
+#define TWL4030_HSMIC_EN		0x02
+#define TWL4030_MAINMIC_EN		0x01
+
+/* ANAMICR (0x06) Fields */
+
+#define TWL4030_MICAMPR_EN		0x10
+#define TWL4030_AUXR_EN			0x04
+#define TWL4030_SUBMIC_EN		0x01
+
+/* AVADC_CTL (0x07) Fields */
+
+#define TWL4030_ADCL_EN			0x08
+#define TWL4030_AVADC_CLK_PRIORITY	0x04
+#define TWL4030_ADCR_EN			0x02
+
+/* AUDIO_IF (0x0E) Fields */
+
+#define TWL4030_AIF_SLAVE_EN		0x80
+#define TWL4030_DATA_WIDTH		0x60
+#define TWL4030_DATA_WIDTH_16S_16W	0x00
+#define TWL4030_DATA_WIDTH_32S_16W	0x40
+#define TWL4030_DATA_WIDTH_32S_24W	0x60
+#define TWL4030_AIF_FORMAT		0x18
+#define TWL4030_AIF_FORMAT_CODEC	0x00
+#define TWL4030_AIF_FORMAT_LEFT		0x08
+#define TWL4030_AIF_FORMAT_RIGHT	0x10
+#define TWL4030_AIF_FORMAT_TDM		0x18
+#define TWL4030_AIF_TRI_EN		0x04
+#define TWL4030_CLK256FS_EN		0x02
+#define TWL4030_AIF_EN			0x01
+
+/* HS_GAIN_SET (0x23) Fields */
+
+#define TWL4030_HSR_GAIN		0x0C
+#define TWL4030_HSR_GAIN_PWR_DOWN	0x00
+#define TWL4030_HSR_GAIN_PLUS_6DB	0x04
+#define TWL4030_HSR_GAIN_0DB		0x08
+#define TWL4030_HSR_GAIN_MINUS_6DB	0x0C
+#define TWL4030_HSL_GAIN		0x03
+#define TWL4030_HSL_GAIN_PWR_DOWN	0x00
+#define TWL4030_HSL_GAIN_PLUS_6DB	0x01
+#define TWL4030_HSL_GAIN_0DB		0x02
+#define TWL4030_HSL_GAIN_MINUS_6DB	0x03
+
+/* HS_POPN_SET (0x24) Fields */
+
+#define TWL4030_VMID_EN			0x40
+#define	TWL4030_EXTMUTE			0x20
+#define TWL4030_RAMP_DELAY		0x1C
+#define TWL4030_RAMP_DELAY_20MS		0x00
+#define TWL4030_RAMP_DELAY_40MS		0x04
+#define TWL4030_RAMP_DELAY_81MS		0x08
+#define TWL4030_RAMP_DELAY_161MS	0x0C
+#define TWL4030_RAMP_DELAY_323MS	0x10
+#define TWL4030_RAMP_DELAY_645MS	0x14
+#define TWL4030_RAMP_DELAY_1291MS	0x18
+#define TWL4030_RAMP_DELAY_2581MS	0x1C
+#define TWL4030_RAMP_EN			0x02
+
+/* HFL_CTL (0x29, 0x2A) Fields */
+#define TWL4030_HF_CTL_HB_EN		0x04
+#define TWL4030_HF_CTL_LOOP_EN		0x08
+#define TWL4030_HF_CTL_RAMP_EN		0x10
+#define TWL4030_HF_CTL_REF_EN		0x20
+
+/* APLL_CTL (0x3A) Fields */
+
+#define TWL4030_APLL_EN			0x10
+#define TWL4030_APLL_INFREQ		0x0F
+#define TWL4030_APLL_INFREQ_19200KHZ	0x05
+#define TWL4030_APLL_INFREQ_26000KHZ	0x06
+#define TWL4030_APLL_INFREQ_38400KHZ	0x0F
+
+/* REG_MISC_SET_1 (0x3E) Fields */
+
+#define TWL4030_CLK64_EN		0x80
+#define TWL4030_SCRAMBLE_EN		0x40
+#define TWL4030_FMLOOP_EN		0x20
+#define TWL4030_SMOOTH_ANAVOL_EN	0x02
+#define TWL4030_DIGMIC_LR_SWAP_EN	0x01
+
+extern struct snd_soc_dai twl4030_dai;
+extern struct snd_soc_codec_device soc_codec_dev_twl4030;
+
+#endif	/* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
new file mode 100644
index 000000000000..a2c5064a774b
--- /dev/null
+++ b/sound/soc/codecs/uda134x.c
@@ -0,0 +1,668 @@
+/*
+ * uda134x.c  --  UDA134X ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <sound/uda134x.h>
+#include <sound/l3.h>
+
+#include "uda134x.h"
+
+
+#define POWER_OFF_ON_STANDBY 1
+/*
+  ALSA SOC usually puts the device in standby mode when it's not used
+  for sometime. If you define POWER_OFF_ON_STANDBY the driver will
+  turn off the ADC/DAC when this callback is invoked and turn it back
+  on when needed. Unfortunately this will result in a very light bump
+  (it can be audible only with good earphones). If this bothers you
+  just comment this line, you will have slightly higher power
+  consumption . Please note that sending the L3 command for ADC is
+  enough to make the bump, so it doesn't make difference if you
+  completely take off power from the codec.
+ */
+
+#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda134x_priv {
+	int sysclk;
+	int dai_fmt;
+
+	struct snd_pcm_substream *master_substream;
+	struct snd_pcm_substream *slave_substream;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda134x_reg[UDA134X_REGS_NUM] = {
+	/* Extended address registers */
+	0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* Status, data regs */
+	0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+
+/*
+ * The codec has no support for reading its registers except for peak level...
+ */
+static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= UDA134X_REGS_NUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec,
+	u8 reg, unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= UDA134X_REGS_NUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * Write to the uda134x registers
+ *
+ */
+static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	int ret;
+	u8 addr;
+	u8 data = value;
+	struct uda134x_platform_data *pd = codec->control_data;
+
+	pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
+
+	if (reg >= UDA134X_REGS_NUM) {
+		printk(KERN_ERR "%s unkown register: reg: %d",
+		       __func__, reg);
+		return -EINVAL;
+	}
+
+	uda134x_write_reg_cache(codec, reg, value);
+
+	switch (reg) {
+	case UDA134X_STATUS0:
+	case UDA134X_STATUS1:
+		addr = UDA134X_STATUS_ADDR;
+		break;
+	case UDA134X_DATA000:
+	case UDA134X_DATA001:
+	case UDA134X_DATA010:
+		addr = UDA134X_DATA0_ADDR;
+		break;
+	case UDA134X_DATA1:
+		addr = UDA134X_DATA1_ADDR;
+		break;
+	default:
+		/* It's an extended address register */
+		addr =  (reg | UDA134X_EXTADDR_PREFIX);
+
+		ret = l3_write(&pd->l3,
+			       UDA134X_DATA0_ADDR, &addr, 1);
+		if (ret != 1)
+			return -EIO;
+
+		addr = UDA134X_DATA0_ADDR;
+		data = (value | UDA134X_EXTDATA_PREFIX);
+		break;
+	}
+
+	ret = l3_write(&pd->l3,
+		       addr, &data, 1);
+	if (ret != 1)
+		return -EIO;
+
+	return 0;
+}
+
+static inline void uda134x_reset(struct snd_soc_codec *codec)
+{
+	u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+	uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6));
+	msleep(1);
+	uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010);
+
+	pr_debug("%s mute: %d\n", __func__, mute);
+
+	if (mute)
+		mute_reg |= (1<<2);
+	else
+		mute_reg &= ~(1<<2);
+
+	uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+
+	return 0;
+}
+
+static int uda134x_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+	struct snd_pcm_runtime *master_runtime;
+
+	if (uda134x->master_substream) {
+		master_runtime = uda134x->master_substream->runtime;
+
+		pr_debug("%s constraining to %d bits at %d\n", __func__,
+			 master_runtime->sample_bits,
+			 master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_RATE,
+					     master_runtime->rate,
+					     master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+					     master_runtime->sample_bits,
+					     master_runtime->sample_bits);
+
+		uda134x->slave_substream = substream;
+	} else
+		uda134x->master_substream = substream;
+
+	return 0;
+}
+
+static void uda134x_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	if (uda134x->master_substream == substream)
+		uda134x->master_substream = uda134x->slave_substream;
+
+	uda134x->slave_substream = NULL;
+}
+
+static int uda134x_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+	u8 hw_params;
+
+	if (substream == uda134x->slave_substream) {
+		pr_debug("%s ignoring hw_params for slave substream\n",
+			 __func__);
+		return 0;
+	}
+
+	hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+	hw_params &= STATUS0_SYSCLK_MASK;
+	hw_params &= STATUS0_DAIFMT_MASK;
+
+	pr_debug("%s sysclk: %d, rate:%d\n", __func__,
+		 uda134x->sysclk, params_rate(params));
+
+	/* set SYSCLK / fs ratio */
+	switch (uda134x->sysclk / params_rate(params)) {
+	case 512:
+		break;
+	case 384:
+		hw_params |= (1<<4);
+		break;
+	case 256:
+		hw_params |= (1<<5);
+		break;
+	default:
+		printk(KERN_ERR "%s unsupported fs\n", __func__);
+		return -EINVAL;
+	}
+
+	pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
+		 uda134x->dai_fmt, params_format(params));
+
+	/* set DAI format and word length */
+	switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		switch (params_format(params)) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+			hw_params |= (1<<1);
+			break;
+		case SNDRV_PCM_FORMAT_S18_3LE:
+			hw_params |= (1<<2);
+			break;
+		case SNDRV_PCM_FORMAT_S20_3LE:
+			hw_params |= ((1<<2) | (1<<1));
+			break;
+		default:
+			printk(KERN_ERR "%s unsupported format (right)\n",
+			       __func__);
+			return -EINVAL;
+		}
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		hw_params |= (1<<3);
+		break;
+	default:
+		printk(KERN_ERR "%s unsupported format\n", __func__);
+		return -EINVAL;
+	}
+
+	uda134x_write(codec, UDA134X_STATUS0, hw_params);
+
+	return 0;
+}
+
+static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+		 clk_id, freq, dir);
+
+	/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+	   because the codec is slave. Of course limitations of the clock
+	   master (the IIS controller) apply.
+	   We'll error out on set_hw_params if it's not OK */
+	if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+		uda134x->sysclk = freq;
+		return 0;
+	}
+
+	printk(KERN_ERR "%s unsupported sysclk\n", __func__);
+	return -EINVAL;
+}
+
+static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	pr_debug("%s fmt: %08X\n", __func__, fmt);
+
+	/* codec supports only full slave mode */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		printk(KERN_ERR "%s unsupported slave mode\n", __func__);
+		return -EINVAL;
+	}
+
+	/* no support for clock inversion */
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
+		return -EINVAL;
+	}
+
+	/* We can't setup DAI format here as it depends on the word bit num */
+	/* so let's just store the value for later */
+	uda134x->dai_fmt = fmt;
+
+	return 0;
+}
+
+static int uda134x_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	u8 reg;
+	struct uda134x_platform_data *pd = codec->control_data;
+	int i;
+	u8 *cache = codec->reg_cache;
+
+	pr_debug("%s bias level %d\n", __func__, level);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* ADC, DAC on */
+		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+		uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* power on */
+		if (pd->power) {
+			pd->power(1);
+			/* Sync reg_cache with the hardware */
+			for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++)
+				codec->write(codec, i, *cache++);
+		}
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* ADC, DAC power off */
+		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+		uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* power off */
+		if (pd->power)
+			pd->power(0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
+					    "Minimum2", "Maximum"};
+static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda134x_mixmode[] = {"Differential", "Analog1",
+					"Analog2", "Both"};
+
+static const struct soc_enum uda134x_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
+SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
+SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
+SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
+
+SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
+SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
+
+SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
+SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new uda1340_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static int uda134x_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i, n;
+	const struct snd_kcontrol_new *ctrls;
+	struct uda134x_platform_data *pd = codec->control_data;
+
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1344:
+		n = ARRAY_SIZE(uda1340_snd_controls);
+		ctrls = uda1340_snd_controls;
+		break;
+	case UDA134X_UDA1341:
+		n = ARRAY_SIZE(uda1341_snd_controls);
+		ctrls = uda1341_snd_controls;
+		break;
+	default:
+		printk(KERN_ERR "%s unkown codec type: %d",
+		       __func__, pd->model);
+		return -EINVAL;
+	}
+
+	for (i = 0; i < n; i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&ctrls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+struct snd_soc_dai uda134x_dai = {
+	.name = "UDA134X",
+	/* playback capabilities */
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA134X_RATES,
+		.formats = UDA134X_FORMATS,
+	},
+	/* capture capabilities */
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA134X_RATES,
+		.formats = UDA134X_FORMATS,
+	},
+	/* pcm operations */
+	.ops = {
+		.startup = uda134x_startup,
+		.shutdown = uda134x_shutdown,
+		.hw_params = uda134x_hw_params,
+		.digital_mute = uda134x_mute,
+		.set_sysclk = uda134x_set_dai_sysclk,
+		.set_fmt = uda134x_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL(uda134x_dai);
+
+
+static int uda134x_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct uda134x_priv *uda134x;
+	void *codec_setup_data = socdev->codec_data;
+	int ret = -ENOMEM;
+	struct uda134x_platform_data *pd;
+
+	printk(KERN_INFO "UDA134X SoC Audio Codec\n");
+
+	if (!codec_setup_data) {
+		printk(KERN_ERR "UDA134X SoC codec: "
+		       "missing L3 bitbang function\n");
+		return -ENODEV;
+	}
+
+	pd = codec_setup_data;
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1341:
+	case UDA134X_UDA1344:
+		break;
+	default:
+		printk(KERN_ERR "UDA134X SoC codec: "
+		       "unsupported model %d\n",
+			pd->model);
+		return -EINVAL;
+	}
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return ret;
+
+	codec = socdev->codec;
+
+	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
+	if (uda134x == NULL)
+		goto priv_err;
+	codec->private_data = uda134x;
+
+	codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		goto reg_err;
+
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache_size = sizeof(uda134x_reg);
+	codec->reg_cache_step = 1;
+
+	codec->name = "UDA134X";
+	codec->owner = THIS_MODULE;
+	codec->dai = &uda134x_dai;
+	codec->num_dai = 1;
+	codec->read = uda134x_read_reg_cache;
+	codec->write = uda134x_write;
+#ifdef POWER_OFF_ON_STANDBY
+	codec->set_bias_level = uda134x_set_bias_level;
+#endif
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->control_data = codec_setup_data;
+
+	if (pd->power)
+		pd->power(1);
+
+	uda134x_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register pcms\n");
+		goto pcm_err;
+	}
+
+	ret = uda134x_add_controls(codec);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register controls\n");
+		goto pcm_err;
+	}
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+reg_err:
+	kfree(codec->private_data);
+priv_err:
+	kfree(codec);
+	return ret;
+}
+
+/* power down chip */
+static int uda134x_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	kfree(codec->private_data);
+	kfree(codec->reg_cache);
+	kfree(codec);
+
+	return 0;
+}
+
+#if defined(CONFIG_PM)
+static int uda134x_soc_suspend(struct platform_device *pdev,
+						pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int uda134x_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+	return 0;
+}
+#else
+#define uda134x_soc_suspend NULL
+#define uda134x_soc_resume NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda134x = {
+	.probe =        uda134x_soc_probe,
+	.remove =       uda134x_soc_remove,
+	.suspend =      uda134x_soc_suspend,
+	.resume =       uda134x_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
+
+static int __init uda134x_init(void)
+{
+	return snd_soc_register_dai(&uda134x_dai);
+}
+module_init(uda134x_init);
+
+static void __exit uda134x_exit(void)
+{
+	snd_soc_unregister_dai(&uda134x_dai);
+}
+module_exit(uda134x_exit);
+
+MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
new file mode 100644
index 000000000000..94f440490b31
--- /dev/null
+++ b/sound/soc/codecs/uda134x.h
@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR	5
+#define UDA134X_DATA0_ADDR	((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR	((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR	((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX	0xC0
+#define UDA134X_EXTDATA_PREFIX	0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000	0
+#define UDA134X_EA001	1
+#define UDA134X_EA010	2
+#define UDA134X_EA011	3
+#define UDA134X_EA100	4
+#define UDA134X_EA101	5
+#define UDA134X_EA110	6
+#define UDA134X_EA111	7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1	13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a69ee72a7af5..e6bf0844fbf3 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -407,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * when the DAI is being clocked by the CPU DAI. It's up to the
  * machine and cpu DAI driver to do this before we are called.
  */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -439,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
 }
 
 static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -477,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -560,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt,
 	},
@@ -579,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt,
 	},
@@ -598,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.set_fmt = uda1380_set_dai_fmt,
 	},
 },
@@ -680,7 +677,7 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	/* uda1380 init */
 	uda1380_add_controls(codec);
 	uda1380_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		pr_err("uda1380: failed to register card\n");
 		goto card_err;
@@ -844,6 +841,18 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
 
+static int __init uda1380_modinit(void)
+{
+	return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_init(uda1380_modinit);
+
+static void __exit uda1380_exit(void)
+{
+	snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_exit(uda1380_exit);
+
 MODULE_AUTHOR("Giorgio Padrin");
 MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
new file mode 100644
index 000000000000..e3989d406f54
--- /dev/null
+++ b/sound/soc/codecs/wm8350.c
@@ -0,0 +1,1583 @@
+/*
+ * wm8350.c -- WM8350 ALSA SoC audio driver
+ *
+ * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wm8350/audio.h>
+#include <linux/mfd/wm8350/core.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8350.h"
+
+#define WM8350_OUTn_0dB 0x39
+
+#define WM8350_RAMP_NONE	0
+#define WM8350_RAMP_UP		1
+#define WM8350_RAMP_DOWN	2
+
+/* We only include the analogue supplies here; the digital supplies
+ * need to be available well before this driver can be probed.
+ */
+static const char *supply_names[] = {
+	"AVDD",
+	"HPVDD",
+};
+
+struct wm8350_output {
+	u16 active;
+	u16 left_vol;
+	u16 right_vol;
+	u16 ramp;
+	u16 mute;
+};
+
+struct wm8350_data {
+	struct snd_soc_codec codec;
+	struct wm8350_output out1;
+	struct wm8350_output out2;
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
+					    unsigned int reg)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350->reg_cache[reg];
+}
+
+static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
+				      unsigned int reg)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350_reg_read(wm8350, reg);
+}
+
+static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+			      unsigned int value)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350_reg_write(wm8350, reg, value);
+}
+
+/*
+ * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
+{
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_data->out1;
+	struct wm8350 *wm8350 = codec->control_data;
+	int left_complete = 0, right_complete = 0;
+	u16 reg, val;
+
+	/* left channel */
+	reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME);
+	val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+
+	if (out1->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out1->left_vol) {
+			val++;
+			reg &= ~WM8350_OUT1L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else if (out1->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT1L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else
+		return 1;
+
+	/* right channel */
+	reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME);
+	val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	if (out1->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out1->right_vol) {
+			val++;
+			reg &= ~WM8350_OUT1R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	} else if (out1->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT1R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	}
+
+	/* only hit the update bit if either volume has changed this step */
+	if (!left_complete || !right_complete)
+		wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU);
+
+	return left_complete & right_complete;
+}
+
+/*
+ * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
+{
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out2 = &wm8350_data->out2;
+	struct wm8350 *wm8350 = codec->control_data;
+	int left_complete = 0, right_complete = 0;
+	u16 reg, val;
+
+	/* left channel */
+	reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME);
+	val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	if (out2->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out2->left_vol) {
+			val++;
+			reg &= ~WM8350_OUT2L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else if (out2->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT2L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else
+		return 1;
+
+	/* right channel */
+	reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME);
+	val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	if (out2->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out2->right_vol) {
+			val++;
+			reg &= ~WM8350_OUT2R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	} else if (out2->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT2R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	}
+
+	/* only hit the update bit if either volume has changed this step */
+	if (!left_complete || !right_complete)
+		wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU);
+
+	return left_complete & right_complete;
+}
+
+/*
+ * This work ramps both output PGAs at stream start/stop time to
+ * minimise pop associated with DAPM power switching.
+ * It's best to enable Zero Cross when ramping occurs to minimise any
+ * zipper noises.
+ */
+static void wm8350_pga_work(struct work_struct *work)
+{
+	struct snd_soc_codec *codec =
+	    container_of(work, struct snd_soc_codec, delayed_work.work);
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_data->out1,
+	    *out2 = &wm8350_data->out2;
+	int i, out1_complete, out2_complete;
+
+	/* do we need to ramp at all ? */
+	if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE)
+		return;
+
+	/* PGA volumes have 6 bits of resolution to ramp */
+	for (i = 0; i <= 63; i++) {
+		out1_complete = 1, out2_complete = 1;
+		if (out1->ramp != WM8350_RAMP_NONE)
+			out1_complete = wm8350_out1_ramp_step(codec);
+		if (out2->ramp != WM8350_RAMP_NONE)
+			out2_complete = wm8350_out2_ramp_step(codec);
+
+		/* ramp finished ? */
+		if (out1_complete && out2_complete)
+			break;
+
+		/* we need to delay longer on the up ramp */
+		if (out1->ramp == WM8350_RAMP_UP ||
+		    out2->ramp == WM8350_RAMP_UP) {
+			/* delay is longer over 0dB as increases are larger */
+			if (i >= WM8350_OUTn_0dB)
+				schedule_timeout_interruptible(msecs_to_jiffies
+							       (2));
+			else
+				schedule_timeout_interruptible(msecs_to_jiffies
+							       (1));
+		} else
+			udelay(50);	/* doesn't matter if we delay longer */
+	}
+
+	out1->ramp = WM8350_RAMP_NONE;
+	out2->ramp = WM8350_RAMP_NONE;
+}
+
+/*
+ * WM8350 Controls
+ */
+
+static int pga_event(struct snd_soc_dapm_widget *w,
+		     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out;
+
+	switch (w->shift) {
+	case 0:
+	case 1:
+		out = &wm8350_data->out1;
+		break;
+	case 2:
+	case 3:
+		out = &wm8350_data->out2;
+		break;
+
+	default:
+		BUG();
+		return -1;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		out->ramp = WM8350_RAMP_UP;
+		out->active = 1;
+
+		if (!delayed_work_pending(&codec->delayed_work))
+			schedule_delayed_work(&codec->delayed_work,
+					      msecs_to_jiffies(1));
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		out->ramp = WM8350_RAMP_DOWN;
+		out->active = 0;
+
+		if (!delayed_work_pending(&codec->delayed_work))
+			schedule_delayed_work(&codec->delayed_work,
+					      msecs_to_jiffies(1));
+		break;
+	}
+
+	return 0;
+}
+
+static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_output *out = NULL;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int ret;
+	unsigned int reg = mc->reg;
+	u16 val;
+
+	/* For OUT1 and OUT2 we shadow the values and only actually write
+	 * them out when active in order to ensure the amplifier comes on
+	 * as quietly as possible. */
+	switch (reg) {
+	case WM8350_LOUT1_VOLUME:
+		out = &wm8350_priv->out1;
+		break;
+	case WM8350_LOUT2_VOLUME:
+		out = &wm8350_priv->out2;
+		break;
+	default:
+		break;
+	}
+
+	if (out) {
+		out->left_vol = ucontrol->value.integer.value[0];
+		out->right_vol = ucontrol->value.integer.value[1];
+		if (!out->active)
+			return 1;
+	}
+
+	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+	if (ret < 0)
+		return ret;
+
+	/* now hit the volume update bits (always bit 8) */
+	val = wm8350_codec_read(codec, reg);
+	wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+	return 1;
+}
+
+static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_priv->out1;
+	struct wm8350_output *out2 = &wm8350_priv->out2;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	unsigned int reg = mc->reg;
+
+	/* If these are cached registers use the cache */
+	switch (reg) {
+	case WM8350_LOUT1_VOLUME:
+		ucontrol->value.integer.value[0] = out1->left_vol;
+		ucontrol->value.integer.value[1] = out1->right_vol;
+		return 0;
+
+	case WM8350_LOUT2_VOLUME:
+		ucontrol->value.integer.value[0] = out2->left_vol;
+		ucontrol->value.integer.value[1] = out2->right_vol;
+		return 0;
+
+	default:
+		break;
+	}
+
+	return snd_soc_get_volsw_2r(kcontrol, ucontrol);
+}
+
+/* double control with volume update */
+#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+				xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+		SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r, \
+	.get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		 .rshift = xshift, .max = xmax, .invert = xinvert}, }
+
+static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
+static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
+static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
+static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
+static const char *wm8350_adcfilter[] = { "None", "High Pass" };
+static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
+static const char *wm8350_lr[] = { "Left", "Right" };
+
+static const struct soc_enum wm8350_enum[] = {
+	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp),
+	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
+	SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
+};
+
+static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525);
+static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600);
+static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
+static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
+static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
+
+static const unsigned int capture_sd_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1),
+	13, 15, TLV_DB_SCALE_ITEM(0, 0, 0),
+};
+
+static const struct snd_kcontrol_new wm8350_snd_controls[] = {
+	SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
+	SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
+	SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume",
+				WM8350_DAC_DIGITAL_VOLUME_L,
+				WM8350_DAC_DIGITAL_VOLUME_R,
+				0, 255, 0, dac_pcm_tlv),
+	SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
+	SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
+	SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
+	SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
+	SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
+	SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+	SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
+				WM8350_ADC_DIGITAL_VOLUME_L,
+				WM8350_ADC_DIGITAL_VOLUME_R,
+				0, 255, 0, adc_pcm_tlv),
+	SOC_DOUBLE_TLV("Capture Sidetone Volume",
+		       WM8350_ADC_DIVIDER,
+		       8, 4, 15, 1, capture_sd_tlv),
+	SOC_WM8350_DOUBLE_R_TLV("Capture Volume",
+				WM8350_LEFT_INPUT_VOLUME,
+				WM8350_RIGHT_INPUT_VOLUME,
+				2, 63, 0, pre_amp_tlv),
+	SOC_DOUBLE_R("Capture ZC Switch",
+		     WM8350_LEFT_INPUT_VOLUME,
+		     WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
+	SOC_SINGLE_TLV("Left Input Left Sidetone Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Left Input Right Sidetone Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Left Input Bypass Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+		       9, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Left Sidetone Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       1, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Right Sidetone Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Bypass Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       13, 7, 0, out_mix_tlv),
+	SOC_SINGLE("Left Input Mixer +20dB Switch",
+		   WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0),
+	SOC_SINGLE("Right Input Mixer +20dB Switch",
+		   WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0),
+	SOC_SINGLE_TLV("Out4 Capture Volume",
+		       WM8350_INPUT_MIXER_VOLUME,
+		       1, 7, 0, out_mix_tlv),
+	SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume",
+				WM8350_LOUT1_VOLUME,
+				WM8350_ROUT1_VOLUME,
+				2, 63, 0, out_pga_tlv),
+	SOC_DOUBLE_R("Out1 Playback ZC Switch",
+		     WM8350_LOUT1_VOLUME,
+		     WM8350_ROUT1_VOLUME, 13, 1, 0),
+	SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume",
+				WM8350_LOUT2_VOLUME,
+				WM8350_ROUT2_VOLUME,
+				2, 63, 0, out_pga_tlv),
+	SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
+		     WM8350_ROUT2_VOLUME, 13, 1, 0),
+	SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),
+	SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+
+	SOC_DOUBLE_R("Out1 Playback Switch",
+		     WM8350_LOUT1_VOLUME,
+		     WM8350_ROUT1_VOLUME,
+		     14, 1, 1),
+	SOC_DOUBLE_R("Out2 Playback Switch",
+		     WM8350_LOUT2_VOLUME,
+		     WM8350_ROUT2_VOLUME,
+		     14, 1, 1),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Playback Mixer */
+static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch",
+			WM8350_LEFT_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch",
+			WM8350_LEFT_MIXER_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch",
+			WM8350_LEFT_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Left Sidetone Switch",
+			WM8350_LEFT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Right Sidetone Switch",
+			WM8350_LEFT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Right Playback Mixer */
+static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Sidetone Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Right Sidetone Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Out4 Mixer */
+static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Right Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Right Capture Switch",
+			WM8350_OUT4_MIXER_CONTROL, 9, 1, 0),
+	SOC_DAPM_SINGLE("Out3 Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Right Mixer Switch",
+			WM8350_OUT4_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("Left Mixer Switch",
+			WM8350_OUT4_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Out3 Mixer */
+static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_OUT3_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Capture Switch",
+			WM8350_OUT3_MIXER_CONTROL, 8, 1, 0),
+	SOC_DAPM_SINGLE("Out4 Playback Switch",
+			WM8350_OUT3_MIXER_CONTROL, 3, 1, 0),
+	SOC_DAPM_SINGLE("Left Mixer Switch",
+			WM8350_OUT3_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Left Input Mixer */
+static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE("PGA Capture Switch",
+			WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Right Input Mixer */
+static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE("PGA Capture Switch",
+			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Left Mic Mixer */
+static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = {
+	SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0),
+};
+
+/* Right Mic Mixer */
+static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = {
+	SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0),
+	SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0),
+	SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0),
+};
+
+/* Beep Switch */
+static const struct snd_kcontrol_new wm8350_beep_switch_controls =
+SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
+
+/* Out4 Capture Mux */
+static const struct snd_kcontrol_new wm8350_out4_capture_controls =
+SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+
+static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
+
+	SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL,
+			   0, pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0,
+			   pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL,
+			   0, pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0,
+			   pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2,
+			   7, 0, &wm8350_right_capt_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_capt_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2,
+			   6, 0, &wm8350_left_capt_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_capt_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0,
+			   &wm8350_out4_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_out4_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0,
+			   &wm8350_out3_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_out3_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0,
+			   &wm8350_right_play_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_play_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0,
+			   &wm8350_left_play_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_play_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0,
+			   &wm8350_left_mic_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_mic_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0,
+			   &wm8350_right_mic_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_mic_mixer_controls)),
+
+	/* virtual mixer for Beep and Out2R */
+	SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0,
+			    &wm8350_beep_switch_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
+			 WM8350_POWER_MGMT_4, 3, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture",
+			 WM8350_POWER_MGMT_4, 2, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback",
+			 WM8350_POWER_MGMT_4, 5, 0),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback",
+			 WM8350_POWER_MGMT_4, 4, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0),
+
+	SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0,
+			 &wm8350_out4_capture_controls),
+
+	SND_SOC_DAPM_OUTPUT("OUT1R"),
+	SND_SOC_DAPM_OUTPUT("OUT1L"),
+	SND_SOC_DAPM_OUTPUT("OUT2R"),
+	SND_SOC_DAPM_OUTPUT("OUT2L"),
+	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_OUTPUT("OUT4"),
+
+	SND_SOC_DAPM_INPUT("IN1RN"),
+	SND_SOC_DAPM_INPUT("IN1RP"),
+	SND_SOC_DAPM_INPUT("IN2R"),
+	SND_SOC_DAPM_INPUT("IN1LP"),
+	SND_SOC_DAPM_INPUT("IN1LN"),
+	SND_SOC_DAPM_INPUT("IN2L"),
+	SND_SOC_DAPM_INPUT("IN3R"),
+	SND_SOC_DAPM_INPUT("IN3L"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* left playback mixer */
+	{"Left Playback Mixer", "Playback Switch", "Left DAC"},
+	{"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"},
+	{"Left Playback Mixer", "Right Playback Switch", "Right DAC"},
+	{"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+	{"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+	/* right playback mixer */
+	{"Right Playback Mixer", "Playback Switch", "Right DAC"},
+	{"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"},
+	{"Right Playback Mixer", "Left Playback Switch", "Left DAC"},
+	{"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+	{"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+	/* out4 playback mixer */
+	{"Out4 Mixer", "Right Playback Switch", "Right DAC"},
+	{"Out4 Mixer", "Left Playback Switch", "Left DAC"},
+	{"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"},
+	{"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"},
+	{"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"},
+	{"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+	{"OUT4", NULL, "Out4 Mixer"},
+
+	/* out3 playback mixer */
+	{"Out3 Mixer", "Left Playback Switch", "Left DAC"},
+	{"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"},
+	{"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+	{"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"},
+	{"OUT3", NULL, "Out3 Mixer"},
+
+	/* out2 */
+	{"Right Out2 PGA", NULL, "Right Playback Mixer"},
+	{"Left Out2 PGA", NULL, "Left Playback Mixer"},
+	{"OUT2L", NULL, "Left Out2 PGA"},
+	{"OUT2R", NULL, "Right Out2 PGA"},
+
+	/* out1 */
+	{"Right Out1 PGA", NULL, "Right Playback Mixer"},
+	{"Left Out1 PGA", NULL, "Left Playback Mixer"},
+	{"OUT1L", NULL, "Left Out1 PGA"},
+	{"OUT1R", NULL, "Right Out1 PGA"},
+
+	/* ADCs */
+	{"Left ADC", NULL, "Left Capture Mixer"},
+	{"Right ADC", NULL, "Right Capture Mixer"},
+
+	/* Left capture mixer */
+	{"Left Capture Mixer", "L2 Capture Volume", "IN2L"},
+	{"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"},
+	{"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"},
+	{"Left Capture Mixer", NULL, "Out4 Capture Channel"},
+
+	/* Right capture mixer */
+	{"Right Capture Mixer", "L2 Capture Volume", "IN2R"},
+	{"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"},
+	{"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"},
+	{"Right Capture Mixer", NULL, "Out4 Capture Channel"},
+
+	/* L3 Inputs */
+	{"IN3L PGA", NULL, "IN3L"},
+	{"IN3R PGA", NULL, "IN3R"},
+
+	/* Left Mic mixer */
+	{"Left Mic Mixer", "INN Capture Switch", "IN1LN"},
+	{"Left Mic Mixer", "INP Capture Switch", "IN1LP"},
+	{"Left Mic Mixer", "IN2 Capture Switch", "IN2L"},
+
+	/* Right Mic mixer */
+	{"Right Mic Mixer", "INN Capture Switch", "IN1RN"},
+	{"Right Mic Mixer", "INP Capture Switch", "IN1RP"},
+	{"Right Mic Mixer", "IN2 Capture Switch", "IN2R"},
+
+	/* out 4 capture */
+	{"Out4 Capture Channel", NULL, "Out4 Mixer"},
+
+	/* Beep */
+	{"Beep", NULL, "IN3R PGA"},
+};
+
+static int wm8350_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&wm8350_snd_controls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static int wm8350_add_widgets(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec,
+					wm8350_dapm_widgets,
+					ARRAY_SIZE(wm8350_dapm_widgets));
+	if (ret != 0) {
+		dev_err(codec->dev, "dapm control register failed\n");
+		return ret;
+	}
+
+	/* set up audio paths */
+	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	if (ret != 0) {
+		dev_err(codec->dev, "DAPM route register failed\n");
+		return ret;
+	}
+
+	return snd_soc_dapm_new_widgets(codec);
+}
+
+static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	u16 fll_4;
+
+	switch (clk_id) {
+	case WM8350_MCLK_SEL_MCLK:
+		wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+				  WM8350_MCLK_SEL);
+		break;
+	case WM8350_MCLK_SEL_PLL_MCLK:
+	case WM8350_MCLK_SEL_PLL_DAC:
+	case WM8350_MCLK_SEL_PLL_ADC:
+	case WM8350_MCLK_SEL_PLL_32K:
+		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+				WM8350_MCLK_SEL);
+		fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+		    ~WM8350_FLL_CLK_SRC_MASK;
+		wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+		break;
+	}
+
+	/* MCLK direction */
+	if (dir == WM8350_MCLK_DIR_OUT)
+		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+				WM8350_MCLK_DIR);
+	else
+		wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+				  WM8350_MCLK_DIR);
+
+	return 0;
+}
+
+static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 val;
+
+	switch (div_id) {
+	case WM8350_ADC_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+		    ~WM8350_ADC_CLKDIV_MASK;
+		wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+		break;
+	case WM8350_DAC_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+		    ~WM8350_DAC_CLKDIV_MASK;
+		wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+		break;
+	case WM8350_BCLK_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_BCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_OPCLK_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_OPCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_SYS_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_MCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_DACLR_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+		    ~WM8350_DACLRC_RATE_MASK;
+		wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+		break;
+	case WM8350_ADCLR_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+		    ~WM8350_ADCLRC_RATE_MASK;
+		wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	    ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
+	u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+	    ~WM8350_BCLK_MSTR;
+	u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+	    ~WM8350_DACLRC_ENA;
+	u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+	    ~WM8350_ADCLRC_ENA;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		master |= WM8350_BCLK_MSTR;
+		dac_lrc |= WM8350_DACLRC_ENA;
+		adc_lrc |= WM8350_ADCLRC_ENA;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x2 << 8;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x1 << 8;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x3 << 8;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x3 << 8;	/* lg not sure which mode */
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= WM8350_AIF_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= WM8350_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
+	wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+	wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
+	return 0;
+}
+
+static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
+			      int cmd, struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
+	    WM8350_BCLK_MSTR;
+	int enabled = 0;
+
+	/* Check that the DACs or ADCs are enabled since they are
+	 * required for LRC in master mode. The DACs or ADCs need a
+	 * valid audio path i.e. pin -> ADC or DAC -> pin before
+	 * the LRC will be enabled in master mode. */
+	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+		return 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+		    (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
+	} else {
+		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+		    (WM8350_DACR_ENA | WM8350_DACL_ENA);
+	}
+
+	if (!enabled) {
+		dev_err(codec->dev,
+		       "%s: invalid audio path - no clocks available\n",
+		       __func__);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	    ~WM8350_AIF_WL_MASK;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x1 << 10;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x2 << 10;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x3 << 10;
+		break;
+	}
+
+	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	return 0;
+}
+
+static int wm8350_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+
+	if (mute)
+		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+	else
+		wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+	return 0;
+}
+
+/* FLL divisors */
+struct _fll_div {
+	int div;		/* FLL_OUTDIV */
+	int n;
+	int k;
+	int ratio;		/* FLL_FRATIO */
+};
+
+/* The size in bits of the fll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
+			      unsigned int output)
+{
+	u64 Kpart;
+	unsigned int t1, t2, K, Nmod;
+
+	if (output >= 2815250 && output <= 3125000)
+		fll_div->div = 0x4;
+	else if (output >= 5625000 && output <= 6250000)
+		fll_div->div = 0x3;
+	else if (output >= 11250000 && output <= 12500000)
+		fll_div->div = 0x2;
+	else if (output >= 22500000 && output <= 25000000)
+		fll_div->div = 0x1;
+	else {
+		printk(KERN_ERR "wm8350: fll freq %d out of range\n", output);
+		return -EINVAL;
+	}
+
+	if (input > 48000)
+		fll_div->ratio = 1;
+	else
+		fll_div->ratio = 8;
+
+	t1 = output * (1 << (fll_div->div + 1));
+	t2 = input * fll_div->ratio;
+
+	fll_div->n = t1 / t2;
+	Nmod = t1 % t2;
+
+	if (Nmod) {
+		Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+		do_div(Kpart, t2);
+		K = Kpart & 0xFFFFFFFF;
+
+		/* Check if we need to round */
+		if ((K % 10) >= 5)
+			K += 5;
+
+		/* Move down to proper range now rounding is done */
+		K /= 10;
+		fll_div->k = K;
+	} else
+		fll_div->k = 0;
+
+	return 0;
+}
+
+static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
+			  int pll_id, unsigned int freq_in,
+			  unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	struct _fll_div fll_div;
+	int ret = 0;
+	u16 fll_1, fll_4;
+
+	/* power down FLL - we need to do this for reconfiguration */
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+			  WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
+
+	if (freq_out == 0 || freq_in == 0)
+		return ret;
+
+	ret = fll_factors(&fll_div, freq_in, freq_out);
+	if (ret < 0)
+		return ret;
+	dev_dbg(wm8350->dev,
+		"FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+		freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
+		fll_div.ratio);
+
+	/* set up N.K & dividers */
+	fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+	    ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+			   fll_1 | (fll_div.div << 8) | 0x50);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+			   (fll_div.ratio << 11) | (fll_div.
+						    n & WM8350_FLL_N_MASK));
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+	fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+	    ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+			   fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
+			   (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
+
+	/* power FLL on */
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+
+	return 0;
+}
+
+static int wm8350_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_audio_platform_data *platform =
+		wm8350->codec.platform_data;
+	u16 pm1;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_VMID_50K |
+				 platform->codec_current_on << 14);
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1);
+		pm1 &= ~WM8350_VMID_MASK;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_VMID_50K);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
+						    priv->supplies);
+			if (ret != 0)
+				return ret;
+
+			/* Enable the system clock */
+			wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4,
+					WM8350_SYSCLK_ENA);
+
+			/* mute DAC & outputs */
+			wm8350_set_bits(wm8350, WM8350_DAC_MUTE,
+					WM8350_DAC_MUTE_ENA);
+
+			/* discharge cap memory */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+					 platform->dis_out1 |
+					 (platform->dis_out2 << 2) |
+					 (platform->dis_out3 << 4) |
+					 (platform->dis_out4 << 6));
+
+			/* wait for discharge */
+			schedule_timeout_interruptible(msecs_to_jiffies
+						       (platform->
+							cap_discharge_msecs));
+
+			/* enable antipop */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+					 (platform->vmid_s_curve << 8));
+
+			/* ramp up vmid */
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 (platform->
+					  codec_current_charge << 14) |
+					 WM8350_VMID_5K | WM8350_VMIDEN |
+					 WM8350_VBUFEN);
+
+			/* wait for vmid */
+			schedule_timeout_interruptible(msecs_to_jiffies
+						       (platform->
+							vmid_charge_msecs));
+
+			/* turn on vmid 300k  */
+			pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+			    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+			pm1 |= WM8350_VMID_300K |
+				(platform->codec_current_standby << 14);
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 pm1);
+
+
+			/* enable analogue bias */
+			pm1 |= WM8350_BIASEN;
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+			/* disable antipop */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+		} else {
+			/* turn on vmid 300k and reduce current */
+			pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+			    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 pm1 | WM8350_VMID_300K |
+					 (platform->
+					  codec_current_standby << 14));
+
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+
+		/* mute DAC & enable outputs */
+		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+
+		wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3,
+				WM8350_OUT1L_ENA | WM8350_OUT1R_ENA |
+				WM8350_OUT2L_ENA | WM8350_OUT2R_ENA);
+
+		/* enable anti pop S curve */
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+				 (platform->vmid_s_curve << 8));
+
+		/* turn off vmid  */
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~WM8350_VMIDEN;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+		/* wait */
+		schedule_timeout_interruptible(msecs_to_jiffies
+					       (platform->
+						vmid_discharge_msecs));
+
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+				 (platform->vmid_s_curve << 8) |
+				 platform->dis_out1 |
+				 (platform->dis_out2 << 2) |
+				 (platform->dis_out3 << 4) |
+				 (platform->dis_out4 << 6));
+
+		/* turn off VBuf and drain */
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~(WM8350_VBUFEN | WM8350_VMID_MASK);
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_OUTPUT_DRAIN_EN);
+
+		/* wait */
+		schedule_timeout_interruptible(msecs_to_jiffies
+					       (platform->drain_msecs));
+
+		pm1 &= ~WM8350_BIASEN;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+		/* disable anti-pop */
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+		wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME,
+				  WM8350_OUT1L_ENA);
+		wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME,
+				  WM8350_OUT1R_ENA);
+		wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME,
+				  WM8350_OUT2L_ENA);
+		wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME,
+				  WM8350_OUT2R_ENA);
+
+		/* disable clock gen */
+		wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+				  WM8350_SYSCLK_ENA);
+
+		regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
+				       priv->supplies);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8350_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm8350_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8350_codec;
+
+static int wm8350_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct wm8350 *wm8350;
+	struct wm8350_data *priv;
+	int ret;
+	struct wm8350_output *out1;
+	struct wm8350_output *out2;
+
+	BUG_ON(!wm8350_codec);
+
+	socdev->codec = wm8350_codec;
+	codec = socdev->codec;
+	wm8350 = codec->control_data;
+	priv = codec->private_data;
+
+	/* Enable the codec */
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	/* Enable robust clocking mode in ADC */
+	wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
+	wm8350_codec_write(codec, 0xde, 0x13);
+	wm8350_codec_write(codec, WM8350_SECURITY, 0);
+
+	/* read OUT1 & OUT2 volumes */
+	out1 = &priv->out1;
+	out2 = &priv->out2;
+	out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) &
+			  WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) &
+			   WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) &
+			  WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) &
+			   WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0);
+
+	/* Latch VU bits & mute */
+	wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME,
+			WM8350_OUT1_VU | WM8350_OUT1L_MUTE);
+	wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME,
+			WM8350_OUT2_VU | WM8350_OUT2L_MUTE);
+	wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME,
+			WM8350_OUT1_VU | WM8350_OUT1R_MUTE);
+	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
+			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		return ret;
+	}
+
+	wm8350_add_controls(codec);
+	wm8350_add_widgets(codec);
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+	return ret;
+}
+
+static int wm8350_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(&codec->delayed_work);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for its completion */
+	if (ret) {
+		schedule_delayed_work(&codec->delayed_work, 0);
+		flush_scheduled_work();
+	}
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	return 0;
+}
+
+#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8350_dai = {
+	.name = "WM8350",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8350_RATES,
+		.formats = WM8350_FORMATS,
+	},
+	.capture = {
+		 .stream_name = "Capture",
+		 .channels_min = 1,
+		 .channels_max = 2,
+		 .rates = WM8350_RATES,
+		 .formats = WM8350_FORMATS,
+	 },
+	.ops = {
+		 .hw_params = wm8350_pcm_hw_params,
+		 .digital_mute = wm8350_mute,
+		 .trigger = wm8350_pcm_trigger,
+		 .set_fmt = wm8350_set_dai_fmt,
+		 .set_sysclk = wm8350_set_dai_sysclk,
+		 .set_pll = wm8350_set_fll,
+		 .set_clkdiv = wm8350_set_clkdiv,
+	 },
+};
+EXPORT_SYMBOL_GPL(wm8350_dai);
+
+struct snd_soc_codec_device soc_codec_dev_wm8350 = {
+	.probe = 	wm8350_probe,
+	.remove = 	wm8350_remove,
+	.suspend = 	wm8350_suspend,
+	.resume =	wm8350_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
+
+static int wm8350_codec_probe(struct platform_device *pdev)
+{
+	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+	struct wm8350_data *priv;
+	struct snd_soc_codec *codec;
+	int ret, i;
+
+	if (wm8350->codec.platform_data == NULL) {
+		dev_err(&pdev->dev, "No audio platform data supplied\n");
+		return -EINVAL;
+	}
+
+	priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		priv->supplies[i].supply = supply_names[i];
+
+	ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies),
+				 priv->supplies);
+	if (ret != 0)
+		goto err_priv;
+
+	codec = &priv->codec;
+	wm8350->codec.codec = codec;
+
+	wm8350_dai.dev = &pdev->dev;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec->dev = &pdev->dev;
+	codec->name = "WM8350";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8350_codec_read;
+	codec->write = wm8350_codec_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8350_set_bias_level;
+	codec->dai = &wm8350_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8350_MAX_REGISTER;
+	codec->private_data = priv;
+	codec->control_data = wm8350;
+
+	/* Put the codec into reset if it wasn't already */
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0)
+		goto err_supply;
+
+	wm8350_codec = codec;
+
+	ret = snd_soc_register_dai(&wm8350_dai);
+	if (ret != 0)
+		goto err_codec;
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err_supply:
+	regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+err_priv:
+	kfree(priv);
+	wm8350_codec = NULL;
+	return ret;
+}
+
+static int __devexit wm8350_codec_remove(struct platform_device *pdev)
+{
+	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = wm8350->codec.codec;
+	struct wm8350_data *priv = codec->private_data;
+
+	snd_soc_unregister_dai(&wm8350_dai);
+	snd_soc_unregister_codec(codec);
+	regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+	kfree(priv);
+	wm8350_codec = NULL;
+	return 0;
+}
+
+static struct platform_driver wm8350_codec_driver = {
+	.driver = {
+		   .name = "wm8350-codec",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = wm8350_codec_probe,
+	.remove = __devexit_p(wm8350_codec_remove),
+};
+
+static __init int wm8350_init(void)
+{
+	return platform_driver_register(&wm8350_codec_driver);
+}
+module_init(wm8350_init);
+
+static __exit void wm8350_exit(void)
+{
+	platform_driver_unregister(&wm8350_codec_driver);
+}
+module_exit(wm8350_exit);
+
+MODULE_DESCRIPTION("ASoC WM8350 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8350-codec");
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
new file mode 100644
index 000000000000..cc2887aa6c38
--- /dev/null
+++ b/sound/soc/codecs/wm8350.h
@@ -0,0 +1,20 @@
+/*
+ * wm8350.h - WM8903 audio codec interface
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#ifndef _WM8350_H
+#define _WM8350_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm8350_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index d8ca2da8d634..40f8238df717 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -463,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -585,8 +586,6 @@ struct snd_soc_dai wm8510_dai = {
 		.formats = WM8510_FORMATS,},
 	.ops = {
 		.hw_params = wm8510_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8510_mute,
 		.set_fmt = wm8510_set_dai_fmt,
 		.set_clkdiv = wm8510_set_dai_clkdiv,
@@ -659,7 +658,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm8510_add_controls(codec);
 	wm8510_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8510: failed to register card\n");
 		goto card_err;
@@ -890,6 +889,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
 
+static int __init wm8510_modinit(void)
+{
+	return snd_soc_register_dai(&wm8510_dai);
+}
+module_init(wm8510_modinit);
+
+static void __exit wm8510_exit(void)
+{
+	snd_soc_unregister_dai(&wm8510_dai);
+}
+module_exit(wm8510_exit);
+
 MODULE_DESCRIPTION("ASoC WM8510 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 627ebfb4209b..d004e5845298 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -548,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai_link *dai = rtd->dai;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id);
+	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
 
 	paifb &= ~WM8580_AIF_LENGTH_MASK;
 	/* bit size */
@@ -574,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb);
+	wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb);
 	return 0;
 }
 
@@ -798,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = {
 		},
 		.ops = {
 			 .hw_params = wm8580_paif_hw_params,
-		 },
-		.dai_ops = {
 			 .set_fmt = wm8580_set_paif_dai_fmt,
 			 .set_clkdiv = wm8580_set_dai_clkdiv,
 			 .set_pll = wm8580_set_dai_pll,
@@ -818,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = {
 		},
 		.ops = {
 			 .hw_params = wm8580_paif_hw_params,
-		 },
-		.dai_ops = {
 			 .set_fmt = wm8580_set_paif_dai_fmt,
 			 .set_clkdiv = wm8580_set_dai_clkdiv,
 			 .set_pll = wm8580_set_dai_pll,
@@ -873,7 +869,7 @@ static int wm8580_init(struct snd_soc_device *socdev)
 	wm8580_add_controls(codec);
 	wm8580_add_widgets(codec);
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8580: failed to register card\n");
 		goto card_err;
@@ -900,85 +896,85 @@ static struct snd_soc_device *wm8580_socdev;
  *    low  = 0x1a
  *    high = 0x1b
  */
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
 
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-
-static struct i2c_driver wm8580_i2c_driver;
-static struct i2c_client client_template;
-
-static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+static int wm8580_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8580_socdev;
-	struct wm8580_setup_data *setup = socdev->codec_data;
 	struct snd_soc_codec *codec = socdev->codec;
-	struct i2c_client *i2c;
 	int ret;
 
-	if (addr != setup->i2c_address)
-		return -ENODEV;
-
-	client_template.adapter = adap;
-	client_template.addr = addr;
-
-	i2c =  kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
-	ret = i2c_attach_client(i2c);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr);
-		goto err;
-	}
-
 	ret = wm8580_init(socdev);
-	if (ret < 0) {
+	if (ret < 0)
 		dev_err(&i2c->dev, "failed to initialise WM8580\n");
-		goto err;
-	}
-
-	return ret;
-
-err:
-	kfree(codec);
-	kfree(i2c);
 	return ret;
 }
 
-static int wm8580_i2c_detach(struct i2c_client *client)
+static int wm8580_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	i2c_detach_client(client);
 	kfree(codec->reg_cache);
-	kfree(client);
 	return 0;
 }
 
-static int wm8580_i2c_attach(struct i2c_adapter *adap)
-{
-	return i2c_probe(adap, &addr_data, wm8580_codec_probe);
-}
+static const struct i2c_device_id wm8580_i2c_id[] = {
+	{ "wm8580", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
-/* corgi i2c codec control layer */
 static struct i2c_driver wm8580_i2c_driver = {
 	.driver = {
 		.name = "WM8580 I2C Codec",
 		.owner = THIS_MODULE,
 	},
-	.attach_adapter = wm8580_i2c_attach,
-	.detach_client =  wm8580_i2c_detach,
-	.command =        NULL,
+	.probe =    wm8580_i2c_probe,
+	.remove =   wm8580_i2c_remove,
+	.id_table = wm8580_i2c_id,
 };
 
-static struct i2c_client client_template = {
-	.name =   "WM8580",
-	.driver = &wm8580_i2c_driver,
-};
+static int wm8580_add_i2c_device(struct platform_device *pdev,
+				 const struct wm8580_setup_data *setup)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+	int ret;
+
+	ret = i2c_add_driver(&wm8580_i2c_driver);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "can't add i2c driver\n");
+		return ret;
+	}
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = setup->i2c_address;
+	strlcpy(info.type, "wm8580", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(setup->i2c_bus);
+	if (!adapter) {
+		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+			setup->i2c_bus);
+		goto err_driver;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		goto err_driver;
+	}
+
+	return 0;
+
+err_driver:
+	i2c_del_driver(&wm8580_i2c_driver);
+	return -ENODEV;
+}
 #endif
 
 static int wm8580_probe(struct platform_device *pdev)
@@ -1011,11 +1007,8 @@ static int wm8580_probe(struct platform_device *pdev)
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	if (setup->i2c_address) {
-		normal_i2c[0] = setup->i2c_address;
 		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = i2c_add_driver(&wm8580_i2c_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add i2c driver");
+		ret = wm8580_add_i2c_device(pdev, setup);
 	}
 #else
 		/* Add other interfaces here */
@@ -1034,6 +1027,7 @@ static int wm8580_remove(struct platform_device *pdev)
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8580_i2c_driver);
 #endif
 	kfree(codec->private_data);
@@ -1048,6 +1042,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
 
+static int __init wm8580_modinit(void)
+{
+	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_init(wm8580_modinit);
+
+static void __exit wm8580_exit(void)
+{
+	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_exit(wm8580_exit);
+
 MODULE_DESCRIPTION("ASoC WM8580 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 589ddaba21d7..09e4422f6f2f 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -29,6 +29,7 @@
 #define WM8580_CLKSRC_NONE 5
 
 struct wm8580_setup_data {
+	int i2c_bus;
 	unsigned short i2c_address;
 };
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
new file mode 100644
index 000000000000..80b11983e137
--- /dev/null
+++ b/sound/soc/codecs/wm8728.c
@@ -0,0 +1,585 @@
+/*
+ * wm8728.c  --  WM8728 ALSA SoC Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8728.h"
+
+struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+/*
+ * We can't read the WM8728 register space so we cache them instead.
+ * Note that the defaults here aren't the physical defaults, we latch
+ * the volume update bits, mute the output and enable infinite zero
+ * detect.
+ */
+static const u16 wm8728_reg_defaults[] = {
+	0x1ff,
+	0x1ff,
+	0x001,
+	0x100,
+};
+
+static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	return cache[reg];
+}
+
+static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	cache[reg] = value;
+}
+
+/*
+ * write to the WM8728 register space
+ */
+static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8728 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8728_write_reg_cache(codec, reg, value);
+
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new wm8728_snd_controls[] = {
+
+SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
+		 0, 255, 0, wm8728_tlv),
+
+SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
+};
+
+static int wm8728_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&wm8728_snd_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/*
+ * DAPM controls.
+ */
+static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"VOUTL", NULL, "DAC"},
+	{"VOUTR", NULL, "DAC"},
+};
+
+static int wm8728_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets,
+				  ARRAY_SIZE(wm8728_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+static int wm8728_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+	if (mute)
+		wm8728_write(codec, WM8728_DACCTL, mute_reg | 1);
+	else
+		wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1);
+
+	return 0;
+}
+
+static int wm8728_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+	dac &= ~0x18;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		dac |= 0x10;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		dac |= 0x08;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8728_write(codec, WM8728_DACCTL, dac);
+
+	return 0;
+}
+
+static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL);
+
+	/* Currently only I2S is supported by the driver, though the
+	 * hardware is more flexible.
+	 */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* The hardware only support full slave mode */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		iface &= ~0x22;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |=  0x20;
+		iface &= ~0x02;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x02;
+		iface &= ~0x20;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x22;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8728_write(codec, WM8728_IFCTL, iface);
+	return 0;
+}
+
+static int wm8728_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 reg;
+	int i;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Power everything up... */
+			reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+			wm8728_write(codec, WM8728_DACCTL, reg & ~0x4);
+
+			/* ..then sync in the register cache. */
+			for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++)
+				wm8728_write(codec, i,
+					     wm8728_read_reg_cache(codec, i));
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+		wm8728_write(codec, WM8728_DACCTL, reg | 0x4);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000)
+
+#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8728_dai = {
+	.name = "WM8728",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = WM8728_RATES,
+		.formats = WM8728_FORMATS,
+	},
+	.ops = {
+		 .hw_params = wm8728_hw_params,
+		 .digital_mute = wm8728_mute,
+		 .set_fmt = wm8728_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL_GPL(wm8728_dai);
+
+static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm8728_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8728_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+/*
+ * initialise the WM8728 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8728_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	codec->name = "WM8728";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8728_read_reg_cache;
+	codec->write = wm8728_write;
+	codec->set_bias_level = wm8728_set_bias_level;
+	codec->dai = &wm8728_dai;
+	codec->num_dai = 1;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults);
+	codec->reg_cache = kmemdup(wm8728_reg_defaults,
+				   sizeof(wm8728_reg_defaults),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8728: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	wm8728_add_controls(codec);
+	wm8728_add_widgets(codec);
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8728: failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+
+static struct snd_soc_device *wm8728_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8728 2 wire address is determined by GPIO5
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+
+static int wm8728_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct snd_soc_device *socdev = wm8728_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = wm8728_init(socdev);
+	if (ret < 0)
+		pr_err("failed to initialise WM8728\n");
+
+	return ret;
+}
+
+static int wm8728_i2c_remove(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	kfree(codec->reg_cache);
+	return 0;
+}
+
+static const struct i2c_device_id wm8728_i2c_id[] = {
+	{ "wm8728", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id);
+
+static struct i2c_driver wm8728_i2c_driver = {
+	.driver = {
+		.name = "WM8728 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8728_i2c_probe,
+	.remove =   wm8728_i2c_remove,
+	.id_table = wm8728_i2c_id,
+};
+
+static int wm8728_add_i2c_device(struct platform_device *pdev,
+				 const struct wm8728_setup_data *setup)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+	int ret;
+
+	ret = i2c_add_driver(&wm8728_i2c_driver);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "can't add i2c driver\n");
+		return ret;
+	}
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = setup->i2c_address;
+	strlcpy(info.type, "wm8728", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(setup->i2c_bus);
+	if (!adapter) {
+		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+			setup->i2c_bus);
+		goto err_driver;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		goto err_driver;
+	}
+
+	return 0;
+
+err_driver:
+	i2c_del_driver(&wm8728_i2c_driver);
+	return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8728_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8728_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8728_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8728\n");
+
+	return ret;
+}
+
+static int __devexit wm8728_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8728_spi_driver = {
+	.driver = {
+		.name	= "wm8728",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8728_spi_probe,
+	.remove		= __devexit_p(wm8728_spi_remove),
+};
+
+static int wm8728_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+static int wm8728_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct wm8728_setup_data *setup;
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	setup = socdev->codec_data;
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	wm8728_socdev = socdev;
+	ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	if (setup->i2c_address) {
+		codec->hw_write = (hw_write_t)i2c_master_send;
+		ret = wm8728_add_i2c_device(pdev, setup);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	if (setup->spi) {
+		codec->hw_write = (hw_write_t)wm8728_spi_write;
+		ret = spi_register_driver(&wm8728_spi_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add spi driver");
+	}
+#endif
+
+	if (ret != 0)
+		kfree(codec);
+
+	return ret;
+}
+
+/* power down chip */
+static int wm8728_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec->control_data)
+		wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_unregister_device(codec->control_data);
+	i2c_del_driver(&wm8728_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8728_spi_driver);
+#endif
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8728 = {
+	.probe = 	wm8728_probe,
+	.remove = 	wm8728_remove,
+	.suspend = 	wm8728_suspend,
+	.resume =	wm8728_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728);
+
+static int __init wm8728_modinit(void)
+{
+	return snd_soc_register_dai(&wm8728_dai);
+}
+module_init(wm8728_modinit);
+
+static void __exit wm8728_exit(void)
+{
+	snd_soc_unregister_dai(&wm8728_dai);
+}
+module_exit(wm8728_exit);
+
+MODULE_DESCRIPTION("ASoC WM8728 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h
new file mode 100644
index 000000000000..d269c132474b
--- /dev/null
+++ b/sound/soc/codecs/wm8728.h
@@ -0,0 +1,30 @@
+/*
+ * wm8728.h  --  WM8728 ASoC codec driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8728_H
+#define _WM8728_H
+
+#define WM8728_DACLVOL   0x00
+#define WM8728_DACRVOL   0x01
+#define WM8728_DACCTL    0x02
+#define WM8728_IFCTL     0x03
+
+struct wm8728_setup_data {
+	int            spi;
+	int            i2c_bus;
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8728_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7f8a7e36b33e..c444b9f2701e 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -264,7 +264,8 @@ static inline int get_coeff(int mclk, int rate)
 }
 
 static int wm8731_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -293,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -305,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void wm8731_shutdown(struct snd_pcm_substream *substream)
+static void wm8731_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -461,8 +464,6 @@ struct snd_soc_dai wm8731_dai = {
 		.prepare = wm8731_pcm_prepare,
 		.hw_params = wm8731_hw_params,
 		.shutdown = wm8731_shutdown,
-	},
-	.dai_ops = {
 		.digital_mute = wm8731_mute,
 		.set_sysclk = wm8731_set_dai_sysclk,
 		.set_fmt = wm8731_set_dai_fmt,
@@ -544,7 +545,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
 
 	wm8731_add_controls(codec);
 	wm8731_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8731: failed to register card\n");
 		goto card_err;
@@ -792,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
+static int __init wm8731_modinit(void)
+{
+	return snd_soc_register_dai(&wm8731_dai);
+}
+module_init(wm8731_modinit);
+
+static void __exit wm8731_exit(void)
+{
+	snd_soc_unregister_dai(&wm8731_dai);
+}
+module_exit(wm8731_exit);
+
 MODULE_DESCRIPTION("ASoC WM8731 driver");
 MODULE_AUTHOR("Richard Purdie");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9b7296ee5b08..5997fa68e0d5 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -614,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -709,8 +710,6 @@ struct snd_soc_dai wm8750_dai = {
 		.formats = WM8750_FORMATS,},
 	.ops = {
 		.hw_params = wm8750_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8750_mute,
 		.set_fmt = wm8750_set_dai_fmt,
 		.set_sysclk = wm8750_set_dai_sysclk,
@@ -819,7 +818,7 @@ static int wm8750_init(struct snd_soc_device *socdev)
 
 	wm8750_add_controls(codec);
 	wm8750_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8750: failed to register card\n");
 		goto card_err;
@@ -1086,6 +1085,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
 
+static int __init wm8750_modinit(void)
+{
+	return snd_soc_register_dai(&wm8750_dai);
+}
+module_init(wm8750_modinit);
+
+static void __exit wm8750_exit(void)
+{
+	snd_soc_unregister_dai(&wm8750_dai);
+}
+module_exit(wm8750_exit);
+
 MODULE_DESCRIPTION("ASoC WM8750 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d426eaa22185..6c21b50c9375 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -922,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1155,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1323,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS},
 	.capture = { /* dummy for fast DAI switching */
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode1h_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1356,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_pcm_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_pcm_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode1v_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1385,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_pcm_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_pcm_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode2_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1410,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode3_4_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1439,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode3_4_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1608,7 +1605,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
 
 	wm8753_add_controls(codec);
 	wm8753_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8753: failed to register card\n");
 		goto card_err;
@@ -1877,6 +1874,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
+static int __init wm8753_modinit(void)
+{
+	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_init(wm8753_modinit);
+
+static void __exit wm8753_exit(void)
+{
+	snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_exit(wm8753_exit);
+
 MODULE_DESCRIPTION("ASoC WM8753 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3b326c9b5586..6767de10ded0 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -138,6 +138,10 @@
 struct snd_soc_codec_device soc_codec_dev_wm8900;
 
 struct wm8900_priv {
+	struct snd_soc_codec codec;
+
+	u16 reg_cache[WM8900_MAXREG];
+
 	u32 fll_in; /* FLL input frequency */
 	u32 fll_out; /* FLL output frequency */
 };
@@ -727,7 +731,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int wm8900_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1117,8 +1122,6 @@ struct snd_soc_dai wm8900_dai = {
 	 },
 	.ops = {
 		.hw_params = wm8900_hw_params,
-	 },
-	.dai_ops = {
 		 .set_clkdiv = wm8900_set_dai_clkdiv,
 		 .set_pll = wm8900_set_dai_pll,
 		 .set_fmt = wm8900_set_dai_fmt,
@@ -1283,16 +1286,28 @@ static int wm8900_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8900 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8900_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8900_codec;
+
+static int wm8900_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret = 0;
+	struct wm8900_priv *wm8900;
+	struct snd_soc_codec *codec;
 	unsigned int reg;
-	struct i2c_client *i2c_client = socdev->codec->control_data;
+	int ret;
+
+	wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
+	if (wm8900 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8900->codec;
+	codec->private_data = wm8900;
+	codec->reg_cache = &wm8900->reg_cache[0];
+	codec->reg_cache_size = WM8900_MAXREG;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
 	codec->name = "WM8900";
 	codec->owner = THIS_MODULE;
@@ -1300,33 +1315,28 @@ static int wm8900_init(struct snd_soc_device *socdev)
 	codec->write = wm8900_write;
 	codec->dai = &wm8900_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = WM8900_MAXREG;
-	codec->reg_cache = kmemdup(wm8900_reg_defaults,
-				   sizeof(wm8900_reg_defaults), GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	codec->control_data = i2c;
+	codec->set_bias_level = wm8900_set_bias_level;
+	codec->dev = &i2c->dev;
 
 	reg = wm8900_read(codec, WM8900_REG_ID);
 	if (reg != 0x8900) {
-		dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n",
-			reg);
-		return -ENODEV;
-	}
-
-	codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
-	if (codec->private_data == NULL) {
-		ret = -ENOMEM;
-		goto priv_err;
+		dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg);
+		ret = -ENODEV;
+		goto err;
 	}
 
 	/* Read back from the chip */
 	reg = wm8900_chip_read(codec, WM8900_REG_POWER1);
 	reg = (reg >> 12) & 0xf;
-	dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg);
+	dev_info(&i2c->dev, "WM8900 revision %d\n", reg);
 
 	wm8900_reset(codec);
 
+	/* Turn the chip on */
+	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
 	/* Latch the volume update bits */
 	wm8900_write(codec, WM8900_REG_LINVOL,
 		     wm8900_read(codec, WM8900_REG_LINVOL) | 0x100);
@@ -1352,160 +1362,98 @@ static int wm8900_init(struct snd_soc_device *socdev)
 	/* Set the DAC and mixer output bias */
 	wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
 
+	wm8900_dai.dev = &i2c->dev;
+
+	wm8900_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8900_dai);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return ret;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8900);
+	wm8900_codec = NULL;
+	return ret;
+}
+
+static int wm8900_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_dai(&wm8900_dai);
+	snd_soc_unregister_codec(wm8900_codec);
+
+	wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF);
+
+	wm8900_dai.dev = NULL;
+	kfree(wm8900_codec->private_data);
+	wm8900_codec = NULL;
+
+	return 0;
+}
+
+static const struct i2c_device_id wm8900_i2c_id[] = {
+	{ "wm8900", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id);
+
+static struct i2c_driver wm8900_i2c_driver = {
+	.driver = {
+		.name = "WM8900",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8900_i2c_probe,
+	.remove = wm8900_i2c_remove,
+	.id_table = wm8900_i2c_id,
+};
+
+static int wm8900_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (!wm8900_codec) {
+		dev_err(&pdev->dev, "I2C client not yet instantiated\n");
+		return -ENODEV;
+	}
+
+	codec = wm8900_codec;
+	socdev->codec = codec;
+
 	/* Register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		dev_err(&i2c_client->dev, "Failed to register new PCMs\n");
+		dev_err(&pdev->dev, "Failed to register new PCMs\n");
 		goto pcm_err;
 	}
 
-	/* Turn the chip on */
-	codec->bias_level = SND_SOC_BIAS_OFF;
-	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	wm8900_add_controls(codec);
 	wm8900_add_widgets(codec);
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		dev_err(&i2c_client->dev, "Failed to register card\n");
+		dev_err(&pdev->dev, "Failed to register card\n");
 		goto card_err;
 	}
+
 	return ret;
 
 card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
-priv_err:
-	kfree(codec->private_data);
-	return ret;
-}
-
-static struct snd_soc_device *wm8900_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-
-static struct i2c_driver wm8900_i2c_driver;
-static struct i2c_client client_template;
-
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-{
-	struct snd_soc_device *socdev = wm8900_socdev;
-	struct wm8900_setup_data *setup = socdev->codec_data;
-	struct snd_soc_codec *codec = socdev->codec;
-	struct i2c_client *i2c;
-	int ret;
-
-	if (addr != setup->i2c_address)
-		return -ENODEV;
-
-	dev_err(&adap->dev, "Probe on %x\n", addr);
-
-	client_template.adapter = adap;
-	client_template.addr = addr;
-
-	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = i2c_attach_client(i2c);
-	if (ret < 0) {
-		dev_err(&adap->dev,
-			"failed to attach codec at addr %x\n", addr);
-		goto err;
-	}
-
-	ret = wm8900_init(socdev);
-	if (ret < 0) {
-		dev_err(&adap->dev, "failed to initialise WM8900\n");
-		goto err;
-	}
-	return ret;
-
-err:
-	kfree(codec);
-	kfree(i2c);
-	return ret;
-}
-
-static int wm8900_i2c_detach(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	i2c_detach_client(client);
-	kfree(codec->reg_cache);
-	kfree(client);
-	return 0;
-}
-
-static int wm8900_i2c_attach(struct i2c_adapter *adap)
-{
-	return i2c_probe(adap, &addr_data, wm8900_codec_probe);
-}
-
-/* corgi i2c codec control layer */
-static struct i2c_driver wm8900_i2c_driver = {
-	.driver = {
-		.name = "WM8900 I2C codec",
-		.owner = THIS_MODULE,
-	},
-	.attach_adapter = wm8900_i2c_attach,
-	.detach_client =  wm8900_i2c_detach,
-	.command =        NULL,
-};
-
-static struct i2c_client client_template = {
-	.name =   "WM8900",
-	.driver = &wm8900_i2c_driver,
-};
-#endif
-
-static int wm8900_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8900_setup_data *setup;
-	struct snd_soc_codec *codec;
-	int ret = 0;
-
-	dev_info(&pdev->dev, "WM8900 Audio Codec\n");
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	socdev->codec = codec;
-
-	codec->set_bias_level = wm8900_set_bias_level;
-
-	wm8900_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		normal_i2c[0] = setup->i2c_address;
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = i2c_add_driver(&wm8900_i2c_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add i2c driver");
-	}
-#else
-#error Non-I2C interfaces not yet supported
-#endif
 	return ret;
 }
 
@@ -1513,17 +1461,9 @@ static int wm8900_probe(struct platform_device *pdev)
 static int wm8900_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-
-	if (codec->control_data)
-		wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_del_driver(&wm8900_i2c_driver);
-#endif
-	kfree(codec);
 
 	return 0;
 }
@@ -1536,6 +1476,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
 
+static int __init wm8900_modinit(void)
+{
+	return i2c_add_driver(&wm8900_i2c_driver);
+}
+module_init(wm8900_modinit);
+
+static void __exit wm8900_exit(void)
+{
+	i2c_del_driver(&wm8900_i2c_driver);
+}
+module_exit(wm8900_exit);
+
 MODULE_DESCRIPTION("ASoC WM8900 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h
index ba450d99e902..fd15007d10c7 100644
--- a/sound/soc/codecs/wm8900.h
+++ b/sound/soc/codecs/wm8900.h
@@ -52,12 +52,6 @@
 #define WM8900_DAC_CLKDIV_5_5 0x14
 #define WM8900_DAC_CLKDIV_6   0x18
 
-#define WM8900_
-
-struct wm8900_setup_data {
-	unsigned short i2c_address;
-};
-
 extern struct snd_soc_dai wm8900_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8900;
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce40d7877605..bde74546db4a 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -33,19 +33,6 @@
 
 #include "wm8903.h"
 
-struct wm8903_priv {
-	int sysclk;
-
-	/* Reference counts */
-	int charge_pump_users;
-	int class_w_users;
-	int playback_active;
-	int capture_active;
-
-	struct snd_pcm_substream *master_substream;
-	struct snd_pcm_substream *slave_substream;
-};
-
 /* Register defaults at reset */
 static u16 wm8903_reg_defaults[] = {
 	0x8903,     /* R0   - SW Reset and ID */
@@ -223,6 +210,23 @@ static u16 wm8903_reg_defaults[] = {
 	0x0000,     /* R172 - Analogue Output Bias 0 */
 };
 
+struct wm8903_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[ARRAY_SIZE(wm8903_reg_defaults)];
+
+	int sysclk;
+
+	/* Reference counts */
+	int charge_pump_users;
+	int class_w_users;
+	int playback_active;
+	int capture_active;
+
+	struct snd_pcm_substream *master_substream;
+	struct snd_pcm_substream *slave_substream;
+};
+
+
 static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec,
 						 unsigned int reg)
 {
@@ -360,6 +364,8 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
 static void wm8903_reset(struct snd_soc_codec *codec)
 {
 	wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0);
+	memcpy(codec->reg_cache, wm8903_reg_defaults,
+	       sizeof(wm8903_reg_defaults));
 }
 
 #define WM8903_OUTPUT_SHORT 0x8
@@ -392,6 +398,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		break;
 	default:
 		BUG();
+		return -EINVAL;  /* Spurious warning from some compilers */
 	}
 
 	switch (w->shift) {
@@ -403,6 +410,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		break;
 	default:
 		BUG();
+		return -EINVAL;  /* Spurious warning from some compilers */
 	}
 
 	if (event & SND_SOC_DAPM_PRE_PMU) {
@@ -773,14 +781,14 @@ static const struct snd_kcontrol_new left_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
 SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new right_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0),
 SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new left_speaker_mixer[] = {
@@ -788,7 +796,7 @@ SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0),
 SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0),
 SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0,
-		1, 1, 0),
+		0, 1, 0),
 };
 
 static const struct snd_kcontrol_new right_speaker_mixer[] = {
@@ -797,7 +805,7 @@ SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 2, 1, 0),
 SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
 		1, 1, 0),
 SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
-		1, 1, 0),
+		0, 1, 0),
 };
 
 static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
@@ -989,6 +997,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			wm8903_write(codec, WM8903_CLOCK_RATES_2,
+				     WM8903_CLK_SYS_ENA);
+
 			wm8903_run_sequence(codec, 0);
 			wm8903_sync_reg_cache(codec, codec->reg_cache);
 
@@ -1019,6 +1030,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_OFF:
 		wm8903_run_sequence(codec, 32);
+		reg = wm8903_read(codec, WM8903_CLOCK_RATES_2);
+		reg &= ~WM8903_CLK_SYS_ENA;
+		wm8903_write(codec, WM8903_CLOCK_RATES_2, reg);
 		break;
 	}
 
@@ -1257,7 +1271,8 @@ static struct {
 	{ 0,      0 },
 };
 
-static int wm8903_startup(struct snd_pcm_substream *substream)
+static int wm8903_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1298,7 +1313,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void wm8903_shutdown(struct snd_pcm_substream *substream)
+static void wm8903_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1317,7 +1333,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream)
 }
 
 static int wm8903_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1515,8 +1532,6 @@ struct snd_soc_dai wm8903_dai = {
 		 .startup = wm8903_startup,
 		 .shutdown = wm8903_shutdown,
 		 .hw_params = wm8903_hw_params,
-	},
-	.dai_ops = {
 		 .digital_mute = wm8903_digital_mute,
 		 .set_fmt = wm8903_set_dai_fmt,
 		 .set_sysclk = wm8903_set_dai_sysclk
@@ -1560,17 +1575,43 @@ static int wm8903_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8903 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8903_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8903_codec;
+
+static int wm8903_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	struct i2c_client *i2c = codec->control_data;
-	int ret = 0;
+	struct wm8903_priv *wm8903;
+	struct snd_soc_codec *codec;
+	int ret;
 	u16 val;
 
+	wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
+	if (wm8903 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8903->codec;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->dev = &i2c->dev;
+	codec->name = "WM8903";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8903_read;
+	codec->write = wm8903_write;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8903_set_bias_level;
+	codec->dai = &wm8903_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache);
+	codec->reg_cache = &wm8903->reg_cache[0];
+	codec->private_data = wm8903;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
 	val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
 	if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
 		dev_err(&i2c->dev,
@@ -1578,39 +1619,12 @@ static int wm8903_init(struct snd_soc_device *socdev)
 		return -ENODEV;
 	}
 
-	codec->name = "WM8903";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8903_read;
-	codec->write = wm8903_write;
-	codec->bias_level = SND_SOC_BIAS_OFF;
-	codec->set_bias_level = wm8903_set_bias_level;
-	codec->dai = &wm8903_dai;
-	codec->num_dai = 1;
-	codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults);
-	codec->reg_cache = kmemdup(wm8903_reg_defaults,
-				   sizeof(wm8903_reg_defaults),
-				   GFP_KERNEL);
-	if (codec->reg_cache == NULL) {
-		dev_err(&i2c->dev, "Failed to allocate register cache\n");
-		return -ENOMEM;
-	}
-
 	val = wm8903_read(codec, WM8903_REVISION_NUMBER);
 	dev_info(&i2c->dev, "WM8903 revision %d\n",
 		 val & WM8903_CHIP_REV_MASK);
 
 	wm8903_reset(codec);
 
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "failed to create pcms\n");
-		goto pcm_err;
-	}
-
-	/* SYSCLK is required for pretty much anything */
-	wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA);
-
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1645,47 +1659,45 @@ static int wm8903_init(struct snd_soc_device *socdev)
 	val |= WM8903_DAC_MUTEMODE;
 	wm8903_write(codec, WM8903_DAC_DIGITAL_1, val);
 
-	wm8903_add_controls(codec);
-	wm8903_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "wm8903: failed to register card\n");
-		goto card_err;
+	wm8903_dai.dev = &i2c->dev;
+	wm8903_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8903_dai);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
 	}
 
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
-	return ret;
-}
-
-static struct snd_soc_device *wm8903_socdev;
-
-static int wm8903_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
-{
-	struct snd_soc_device *socdev = wm8903_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = wm8903_init(socdev);
-	if (ret < 0)
-		dev_err(&i2c->dev, "Device initialisation failed\n");
-
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	wm8903_codec = NULL;
+	kfree(wm8903);
 	return ret;
 }
 
 static int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+
+	snd_soc_unregister_dai(&wm8903_dai);
+	snd_soc_unregister_codec(codec);
+
+	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	kfree(codec->private_data);
+
+	wm8903_codec = NULL;
+	wm8903_dai.dev = NULL;
+
 	return 0;
 }
 
@@ -1709,75 +1721,37 @@ static struct i2c_driver wm8903_i2c_driver = {
 static int wm8903_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8903_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8903_priv *wm8903;
-	struct i2c_board_info board_info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *i2c_client;
 	int ret = 0;
 
-	setup = socdev->codec_data;
-
-	if (!setup->i2c_address) {
-		dev_err(&pdev->dev, "No codec address provided\n");
-		return -ENODEV;
+	if (!wm8903_codec) {
+		dev_err(&pdev->dev, "I2C device not yet probed\n");
+		goto err;
 	}
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
+	socdev->codec = wm8903_codec;
 
-	wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
-	if (wm8903 == NULL) {
-		ret = -ENOMEM;
-		goto err_codec;
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		goto err;
 	}
 
-	codec->private_data = wm8903;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
+	wm8903_add_controls(socdev->codec);
+	wm8903_add_widgets(socdev->codec);
 
-	wm8903_socdev = socdev;
-
-	codec->hw_write = (hw_write_t)i2c_master_send;
-	ret = i2c_add_driver(&wm8903_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		goto err_priv;
-	} else {
-		memset(&board_info, 0, sizeof(board_info));
-		strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE);
-		board_info.addr = setup->i2c_address;
-
-		adapter = i2c_get_adapter(setup->i2c_bus);
-		if (!adapter) {
-			dev_err(&pdev->dev, "Can't get I2C bus %d\n",
-				setup->i2c_bus);
-			ret = -ENODEV;
-			goto err_adapter;
-		}
-
-		i2c_client = i2c_new_device(adapter, &board_info);
-		i2c_put_adapter(adapter);
-		if (i2c_client == NULL) {
-			dev_err(&pdev->dev,
-				"I2C driver registration failed\n");
-			ret = -ENODEV;
-			goto err_adapter;
-		}
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "wm8903: failed to register card\n");
+		goto card_err;
 	}
 
 	return ret;
 
-err_adapter:
-	i2c_del_driver(&wm8903_i2c_driver);
-err_priv:
-	kfree(codec->private_data);
-err_codec:
-	kfree(codec);
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+err:
 	return ret;
 }
 
@@ -1792,10 +1766,6 @@ static int wm8903_remove(struct platform_device *pdev)
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-	i2c_unregister_device(socdev->codec->control_data);
-	i2c_del_driver(&wm8903_i2c_driver);
-	kfree(codec->private_data);
-	kfree(codec);
 
 	return 0;
 }
@@ -1808,6 +1778,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
 
+static int __init wm8903_modinit(void)
+{
+	return i2c_add_driver(&wm8903_i2c_driver);
+}
+module_init(wm8903_modinit);
+
+static void __exit wm8903_exit(void)
+{
+	i2c_del_driver(&wm8903_i2c_driver);
+}
+module_exit(wm8903_exit);
+
 MODULE_DESCRIPTION("ASoC WM8903 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index cec622f2f660..0ea27e2b9963 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -18,11 +18,6 @@
 extern struct snd_soc_dai wm8903_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8903;
 
-struct wm8903_setup_data {
-	int i2c_bus;
-	int i2c_address;
-};
-
 #define WM8903_MCLK_DIV_2 1
 #define WM8903_CLK_SYS    2
 #define WM8903_BCLK       3
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index f41a578ddd4f..88ead7f8dd98 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -541,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -634,8 +635,6 @@ struct snd_soc_dai wm8971_dai = {
 		.formats = WM8971_FORMATS,},
 	.ops = {
 		.hw_params = wm8971_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8971_mute,
 		.set_fmt = wm8971_set_dai_fmt,
 		.set_sysclk = wm8971_set_dai_sysclk,
@@ -748,7 +747,7 @@ static int wm8971_init(struct snd_soc_device *socdev)
 
 	wm8971_add_controls(codec);
 	wm8971_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8971: failed to register card\n");
 		goto card_err;
@@ -936,6 +935,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = {
 
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
 
+static int __init wm8971_modinit(void)
+{
+	return snd_soc_register_dai(&wm8971_dai);
+}
+module_init(wm8971_modinit);
+
+static void __exit wm8971_exit(void)
+{
+	snd_soc_unregister_dai(&wm8971_dai);
+}
+module_exit(wm8971_exit);
+
 MODULE_DESCRIPTION("ASoC WM8971 driver");
 MODULE_AUTHOR("Lab126");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 572d22b0880b..5b5afc144478 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -106,6 +106,7 @@ static const u16 wm8990_reg[] = {
 	0x0008,     /* R60 - PLL1 */
 	0x0031,     /* R61 - PLL2 */
 	0x0026,     /* R62 - PLL3 */
+	0x0000,	    /* R63 - Driver internal */
 };
 
 /*
@@ -126,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
 
-	/* Reset register is uncached */
-	if (reg == 0)
+	/* Reset register and reserved registers are uncached */
+	if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
 		return;
 
 	cache[reg] = value;
@@ -1172,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8990_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1222,8 +1223,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		break;
+
 	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*50k */
+		val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+			~WM8990_VMID_MODE_MASK;
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
 		break;
+
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable all output discharge bits */
@@ -1272,10 +1279,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
 			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 			wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
-		} else {
-			/* ON -> standby */
 
+			/* Enable workaround for ADC clocking issue. */
+			wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
+			wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
+			wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
 		}
+
+		/* VMID=2*250k */
+		val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+			~WM8990_VMID_MODE_MASK;
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -1349,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = {
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
 	.ops = {
-		.hw_params = wm8990_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8990_hw_params,
 		.digital_mute = wm8990_mute,
 		.set_fmt = wm8990_set_dai_fmt,
 		.set_clkdiv = wm8990_set_dai_clkdiv,
@@ -1449,7 +1462,7 @@ static int wm8990_init(struct snd_soc_device *socdev)
 
 	wm8990_add_controls(codec);
 	wm8990_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8990: failed to register card\n");
 		goto card_err;
@@ -1630,6 +1643,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
 
+static int __init wm8990_modinit(void)
+{
+	return snd_soc_register_dai(&wm8990_dai);
+}
+module_init(wm8990_modinit);
+
+static void __exit wm8990_exit(void)
+{
+	snd_soc_unregister_dai(&wm8990_dai);
+}
+module_exit(wm8990_exit);
+
 MODULE_DESCRIPTION("ASoC WM8990 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 0e192f3b0788..7114ddc88b4b 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -80,8 +80,8 @@
 #define WM8990_PLL3                             0x3E
 #define WM8990_INTDRIVBITS			0x3F
 
-#define WM8990_REGISTER_COUNT                   60
-#define WM8990_MAX_REGISTER                     0x3F
+#define WM8990_EXT_ACCESS_ENA			0x75
+#define WM8990_EXT_CTL1				0x7a
 
 /*
  * Field Definitions.
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ffb471e420e2..af83d629078a 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -487,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -507,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -533,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97_BUS,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "HiFi Playback",
 		.channels_min = 1,
@@ -688,7 +690,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 
 	ret = wm9712_reset(codec, 0);
 	if (ret < 0) {
-		printk(KERN_ERR "AC97 link error\n");
+		printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
 		goto reset_err;
 	}
 
@@ -698,7 +700,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm9712_add_controls(codec);
 	wm9712_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm9712: failed to register card\n");
 		goto reset_err;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 945b32ed9884..f3ca8aaf0139 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -928,11 +928,10 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
 
 	switch (params_format(params)) {
@@ -954,11 +953,10 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	u16 status;
 
 	/* Gracefully shut down the voice interface. */
@@ -969,12 +967,11 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
 	ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
-static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
+	struct snd_soc_codec *codec = dai->codec;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
 	int reg;
 	u16 vra;
 
@@ -989,12 +986,11 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
+	struct snd_soc_codec *codec = dai->codec;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
 	u16 vra, xsle;
 
 	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
@@ -1028,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97_BUS,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "HiFi Playback",
 		.channels_min = 1,
@@ -1042,8 +1038,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
-		.prepare = ac97_hifi_prepare,},
-	.dai_ops = {
+		.prepare = ac97_hifi_prepare,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,},
 	},
@@ -1056,8 +1051,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
-		.prepare = ac97_aux_prepare,},
-	.dai_ops = {
+		.prepare = ac97_aux_prepare,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,},
 	},
@@ -1077,8 +1071,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.formats = WM9713_PCM_FORMATS,},
 	.ops = {
 		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,},
-	.dai_ops = {
+		.shutdown = wm9713_voiceshutdown,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,
 		.set_fmt = wm9713_set_dai_fmt,
@@ -1097,6 +1090,8 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
 	}
 
 	soc_ac97_ops.reset(codec->ac97);
+	if (soc_ac97_ops.warm_reset)
+		soc_ac97_ops.warm_reset(codec->ac97);
 	if (ac97_read(codec, 0) != wm9713_reg[0])
 		return -EIO;
 	return 0;
@@ -1240,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	wm9713_reset(codec, 0);
 	ret = wm9713_reset(codec, 1);
 	if (ret < 0) {
-		printk(KERN_ERR "AC97 link error\n");
+		printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n");
 		goto reset_err;
 	}
 
@@ -1252,7 +1247,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 
 	wm9713_add_controls(codec);
 	wm9713_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
 		goto reset_err;
 	return 0;
@@ -1288,7 +1283,6 @@ static int wm9713_soc_remove(struct platform_device *pdev)
 	snd_soc_free_ac97_codec(codec);
 	kfree(codec->private_data);
 	kfree(codec->reg_cache);
-	kfree(codec->dai);
 	kfree(codec);
 	return 0;
 }
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 8f7e33834902..b502741692d6 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -17,3 +17,13 @@ config SND_DAVINCI_SOC_EVM
 	help
 	  Say Y if you want to add support for SoC audio on TI
 	  DaVinci EVM platform.
+
+config SND_DAVINCI_SOC_SFFSDR
+	tristate "SoC Audio support for SFFSDR"
+	depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+	select SND_DAVINCI_SOC_I2S
+	select SND_SOC_PCM3008
+	select SFFSDR_FPGA
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Lyrtech SFFSDR board.
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index ca772e5b4637..ca8bae1fc3f6 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 
 # DAVINCI Machine Support
 snd-soc-evm-objs := davinci-evm.o
+snd-soc-sffsdr-objs := davinci-sffsdr.o
 
 obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9e6062cd6b59..01b948bb55a1 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -28,6 +28,8 @@
 
 #define EVM_CODEC_CLOCK 22579200
 
+#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
+		SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
 static int evm_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
@@ -37,14 +39,12 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 
 	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-					 SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
 	if (ret < 0)
 		return ret;
 
 	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
-				       SND_SOC_DAIFMT_IB_NF);
+	ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
 	if (ret < 0)
 		return ret;
 
@@ -128,8 +128,9 @@ static struct snd_soc_dai_link evm_dai = {
 };
 
 /* davinci-evm audio machine driver */
-static struct snd_soc_machine snd_soc_machine_evm = {
+static struct snd_soc_card snd_soc_card_evm = {
 	.name = "DaVinci EVM",
+	.platform = &davinci_soc_platform,
 	.dai_link = &evm_dai,
 	.num_links = 1,
 };
@@ -142,8 +143,7 @@ static struct aic3x_setup_data evm_aic3x_setup = {
 
 /* evm audio subsystem */
 static struct snd_soc_device evm_snd_devdata = {
-	.machine = &snd_soc_machine_evm,
-	.platform = &davinci_soc_platform,
+	.card = &snd_soc_card_evm,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &evm_aic3x_setup,
 };
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index abb5fedb0b1e..0fee779e3c76 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -59,6 +59,7 @@
 #define DAVINCI_MCBSP_PCR_CLKXP		(1 << 1)
 #define DAVINCI_MCBSP_PCR_FSRP		(1 << 2)
 #define DAVINCI_MCBSP_PCR_FSXP		(1 << 3)
+#define DAVINCI_MCBSP_PCR_SCLKME	(1 << 7)
 #define DAVINCI_MCBSP_PCR_CLKRM		(1 << 8)
 #define DAVINCI_MCBSP_PCR_CLKXM		(1 << 9)
 #define DAVINCI_MCBSP_PCR_FSRM		(1 << 10)
@@ -110,17 +111,60 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_platform *platform = socdev->card->platform;
 	u32 w;
+	int ret;
 
 	/* Start the sample generator and enable transmitter/receiver */
 	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
-	else
-		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
 
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* Stop the DMA to avoid data loss */
+		/* while the transmitter is out of reset to handle XSYNCERR */
+		if (platform->pcm_ops->trigger) {
+			ret = platform->pcm_ops->trigger(substream,
+				SNDRV_PCM_TRIGGER_STOP);
+			if (ret < 0)
+				printk(KERN_DEBUG "Playback DMA stop failed\n");
+		}
+
+		/* Enable the transmitter */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+		/* wait for any unexpected frame sync error to occur */
+		udelay(100);
+
+		/* Disable the transmitter to clear any outstanding XSYNCERR */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+		/* Restart the DMA */
+		if (platform->pcm_ops->trigger) {
+			ret = platform->pcm_ops->trigger(substream,
+				SNDRV_PCM_TRIGGER_START);
+			if (ret < 0)
+				printk(KERN_DEBUG "Playback DMA start failed\n");
+		}
+		/* Enable the transmitter */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+	} else {
+
+		/* Enable the reciever */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+	}
+
+
 	/* Start frame sync */
 	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1);
@@ -144,7 +188,8 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream)
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -155,61 +200,138 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
+#define DEFAULT_BITPERSAMPLE	16
+
 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 				   unsigned int fmt)
 {
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
-	u32 w;
+	unsigned int pcr;
+	unsigned int srgr;
+	unsigned int rcr;
+	unsigned int xcr;
+	srgr = DAVINCI_MCBSP_SRGR_FSGM |
+		DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
+		DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG,
-					DAVINCI_MCBSP_PCR_FSXM |
-					DAVINCI_MCBSP_PCR_FSRM |
-					DAVINCI_MCBSP_PCR_CLKXM |
-					DAVINCI_MCBSP_PCR_CLKRM);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG,
-					DAVINCI_MCBSP_SRGR_FSGM);
+		/* cpu is master */
+		pcr = DAVINCI_MCBSP_PCR_FSXM |
+			DAVINCI_MCBSP_PCR_FSRM |
+			DAVINCI_MCBSP_PCR_CLKXM |
+			DAVINCI_MCBSP_PCR_CLKRM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		/* McBSP CLKR pin is the input for the Sample Rate Generator.
+		 * McBSP FSR and FSX are driven by the Sample Rate Generator. */
+		pcr = DAVINCI_MCBSP_PCR_SCLKME |
+			DAVINCI_MCBSP_PCR_FSXM |
+			DAVINCI_MCBSP_PCR_FSRM;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0);
+		/* codec is master */
+		pcr = 0;
 		break;
 	default:
+		printk(KERN_ERR "%s:bad master\n", __func__);
+		return -EINVAL;
+	}
+
+	rcr = DAVINCI_MCBSP_RCR_RFRLEN1(1);
+	xcr = DAVINCI_MCBSP_XCR_XFIG | DAVINCI_MCBSP_XCR_XFRLEN1(1);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_B:
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		/* Davinci doesn't support TRUE I2S, but some codecs will have
+		 * the left and right channels contiguous. This allows
+		 * dsp_a mode to be used with an inverted normal frame clk.
+		 * If your codec is master and does not have contiguous
+		 * channels, then you will have sound on only one channel.
+		 * Try using a different mode, or codec as slave.
+		 *
+		 * The TLV320AIC33 is an example of a codec where this works.
+		 * It has a variable bit clock frequency allowing it to have
+		 * valid data on every bit clock.
+		 *
+		 * The TLV320AIC23 is an example of a codec where this does not
+		 * work. It has a fixed bit clock frequency with progressively
+		 * more empty bit clock slots between channels as the sample
+		 * rate is lowered.
+		 */
+		fmt ^= SND_SOC_DAIFMT_NB_IF;
+	case SND_SOC_DAIFMT_DSP_A:
+		rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
+		xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
+		break;
+	default:
+		printk(KERN_ERR "%s:bad format\n", __func__);
 		return -EINVAL;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_IB_NF:
-		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
-		MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
-			       DAVINCI_MCBSP_PCR_CLKRP, 1);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
-		break;
-	case SND_SOC_DAIFMT_NB_IF:
-		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
-		MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP |
-			       DAVINCI_MCBSP_PCR_FSRP, 1);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+	case SND_SOC_DAIFMT_NB_NF:
+		/* CLKRP Receive clock polarity,
+		 *	1 - sampled on rising edge of CLKR
+		 *	valid on rising edge
+		 * CLKXP Transmit clock polarity,
+		 *	1 - clocked on falling edge of CLKX
+		 *	valid on rising edge
+		 * FSRP  Receive frame sync pol, 0 - active high
+		 * FSXP  Transmit frame sync pol, 0 - active high
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP);
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
-		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
-		MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
-			       DAVINCI_MCBSP_PCR_CLKRP |
-			       DAVINCI_MCBSP_PCR_FSXP |
-			       DAVINCI_MCBSP_PCR_FSRP, 1);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+		/* CLKRP Receive clock polarity,
+		 *	0 - sampled on falling edge of CLKR
+		 *	valid on falling edge
+		 * CLKXP Transmit clock polarity,
+		 *	0 - clocked on rising edge of CLKX
+		 *	valid on falling edge
+		 * FSRP  Receive frame sync pol, 1 - active low
+		 * FSXP  Transmit frame sync pol, 1 - active low
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
 		break;
-	case SND_SOC_DAIFMT_NB_NF:
+	case SND_SOC_DAIFMT_NB_IF:
+		/* CLKRP Receive clock polarity,
+		 *	1 - sampled on rising edge of CLKR
+		 *	valid on rising edge
+		 * CLKXP Transmit clock polarity,
+		 *	1 - clocked on falling edge of CLKX
+		 *	valid on rising edge
+		 * FSRP  Receive frame sync pol, 1 - active low
+		 * FSXP  Transmit frame sync pol, 1 - active low
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP |
+			DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* CLKRP Receive clock polarity,
+		 *	0 - sampled on falling edge of CLKR
+		 *	valid on falling edge
+		 * CLKXP Transmit clock polarity,
+		 *	0 - clocked on rising edge of CLKX
+		 *	valid on falling edge
+		 * FSRP  Receive frame sync pol, 0 - active high
+		 * FSXP  Transmit frame sync pol, 0 - active high
+		 */
 		break;
 	default:
 		return -EINVAL;
 	}
-
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
 	return 0;
 }
 
 static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
@@ -219,25 +341,20 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	u32 w;
 
 	/* general line settings */
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
-				DAVINCI_MCBSP_SPCR_RINTM(3) |
-				DAVINCI_MCBSP_SPCR_XINTM(3) |
-				DAVINCI_MCBSP_SPCR_FREE);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG,
-				DAVINCI_MCBSP_RCR_RFRLEN1(1) |
-				DAVINCI_MCBSP_RCR_RDATDLY(1));
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG,
-				DAVINCI_MCBSP_XCR_XFRLEN1(1) |
-				DAVINCI_MCBSP_XCR_XDATDLY(1) |
-				DAVINCI_MCBSP_XCR_XFIG);
+	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+	} else {
+		w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+	}
 
 	i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+	w = DAVINCI_MCBSP_SRGR_FSGM;
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
 
 	i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1);
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
 
@@ -260,20 +377,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
-	MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
-		       DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+			       DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
 
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
-	MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
-		       DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+	} else {
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+			       DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
 
+	}
 	return 0;
 }
 
-static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -299,8 +420,8 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea;
 	struct evm_snd_platform_data *pdata;
@@ -361,8 +482,8 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
 	struct resource *mem;
 
@@ -381,7 +502,6 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = davinci_i2s_probe,
 	.remove = davinci_i2s_remove,
 	.playback = {
@@ -397,13 +517,24 @@ struct snd_soc_dai davinci_i2s_dai = {
 	.ops = {
 		.startup = davinci_i2s_startup,
 		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = davinci_i2s_hw_params,
 		.set_fmt = davinci_i2s_set_dai_fmt,
 	},
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
+static int __init davinci_i2s_init(void)
+{
+	return snd_soc_register_dai(&davinci_i2s_dai);
+}
+module_init(davinci_i2s_init);
+
+static void __exit davinci_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&davinci_i2s_dai);
+}
+module_exit(davinci_i2s_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 76feaa657375..74abc9b4f1cc 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -14,6 +14,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
+#include <linux/kernel.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -24,13 +25,6 @@
 
 #include "davinci-pcm.h"
 
-#define DAVINCI_PCM_DEBUG 0
-#if DAVINCI_PCM_DEBUG
-#define DPRINTK(x...) printk(KERN_DEBUG x)
-#else
-#define DPRINTK(x...)
-#endif
-
 static struct snd_pcm_hardware davinci_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
@@ -78,8 +72,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
 
-	DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x "
-		"period_size=%x\n", lch, dma_pos, period_size);
+	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
+		"dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
 
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
@@ -112,7 +106,7 @@ static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
 	struct snd_pcm_substream *substream = data;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 
-	DPRINTK("lch=%d, status=0x%x\n", lch, ch_status);
+	pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status);
 
 	if (unlikely(ch_status != DMA_COMPLETE))
 		return;
@@ -316,8 +310,8 @@ static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
 					   &buf->addr, GFP_KERNEL);
 
-	DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
-		(void *) buf->area, (void *) buf->addr, size);
+	pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, "
+		"size=%d\n", (void *) buf->area, (void *) buf->addr, size);
 
 	if (!buf->area)
 		return -ENOMEM;
@@ -384,6 +378,18 @@ struct snd_soc_platform davinci_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(davinci_soc_platform);
 
+static int __init davinci_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&davinci_soc_platform);
+}
+module_init(davinci_soc_platform_init);
+
+static void __exit davinci_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&davinci_soc_platform);
+}
+module_exit(davinci_soc_platform_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
new file mode 100644
index 000000000000..f67579d52765
--- /dev/null
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -0,0 +1,157 @@
+/*
+ * ASoC driver for Lyrtech SFFSDR board.
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow:
+ * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/dma.h>
+#include <asm/plat-sffsdr/sffsdr-fpga.h>
+
+#include <mach/mcbsp.h>
+#include <mach/edma.h>
+
+#include "../codecs/pcm3008.h"
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+
+static int sffsdr_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int fs;
+	int ret = 0;
+
+	/* Set cpu DAI configuration:
+	 * CLKX and CLKR are the inputs for the Sample Rate Generator.
+	 * FSX and FSR are outputs, driven by the sample Rate Generator. */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_RIGHT_J |
+				  SND_SOC_DAIFMT_CBM_CFS |
+				  SND_SOC_DAIFMT_IB_NF);
+	if (ret < 0)
+		return ret;
+
+	/* Fsref can be 32000, 44100 or 48000. */
+	fs = params_rate(params);
+
+	pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
+
+	return sffsdr_fpga_set_codec_fs(fs);
+}
+
+static struct snd_soc_ops sffsdr_ops = {
+	.hw_params = sffsdr_hw_params,
+};
+
+/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sffsdr_dai = {
+	.name = "PCM3008", /* Codec name */
+	.stream_name = "PCM3008 HiFi",
+	.cpu_dai = &davinci_i2s_dai,
+	.codec_dai = &pcm3008_dai,
+	.ops = &sffsdr_ops,
+};
+
+/* davinci-sffsdr audio machine driver */
+static struct snd_soc_card snd_soc_sffsdr = {
+	.name = "DaVinci SFFSDR",
+	.platform = &davinci_soc_platform,
+	.dai_link = &sffsdr_dai,
+	.num_links = 1,
+};
+
+/* sffsdr audio private data */
+static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
+	.dem0_pin = GPIO(45),
+	.dem1_pin = GPIO(46),
+	.pdad_pin = GPIO(47),
+	.pdda_pin = GPIO(38),
+};
+
+/* sffsdr audio subsystem */
+static struct snd_soc_device sffsdr_snd_devdata = {
+	.card = &snd_soc_sffsdr,
+	.codec_dev = &soc_codec_dev_pcm3008,
+	.codec_data = &sffsdr_pcm3008_setup,
+};
+
+static struct resource sffsdr_snd_resources[] = {
+	{
+		.start = DAVINCI_MCBSP_BASE,
+		.end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+		.flags = IORESOURCE_MEM,
+	},
+};
+
+static struct evm_snd_platform_data sffsdr_snd_data = {
+	.tx_dma_ch	= DAVINCI_DMA_MCBSP_TX,
+	.rx_dma_ch	= DAVINCI_DMA_MCBSP_RX,
+};
+
+static struct platform_device *sffsdr_snd_device;
+
+static int __init sffsdr_init(void)
+{
+	int ret;
+
+	sffsdr_snd_device = platform_device_alloc("soc-audio", 0);
+	if (!sffsdr_snd_device) {
+		printk(KERN_ERR "platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata);
+	sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev;
+	sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data;
+
+	ret = platform_device_add_resources(sffsdr_snd_device,
+					    sffsdr_snd_resources,
+					    ARRAY_SIZE(sffsdr_snd_resources));
+	if (ret) {
+		printk(KERN_ERR "platform device add ressources failed\n");
+		goto error;
+	}
+
+	ret = platform_device_add(sffsdr_snd_device);
+	if (ret)
+		goto error;
+
+	return ret;
+
+error:
+	platform_device_put(sffsdr_snd_device);
+	return ret;
+}
+
+static void __exit sffsdr_exit(void)
+{
+	platform_device_unregister(sffsdr_snd_device);
+}
+
+module_init(sffsdr_init);
+module_exit(sffsdr_exit);
+
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 8d73edc56102..95c12b26fe37 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -20,7 +20,7 @@ config SND_SOC_MPC8610_HPCD
 
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
-	depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM
+	depends on PPC_MPC52xx && PPC_BESTCOMM
 	select SND_SOC_OF_SIMPLE
 	select PPC_BESTCOMM_GEN_BD
 	help
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index d2d3da9729f2..64993eda5679 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -284,7 +284,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
  * fsl_dma_new: initialize this PCM driver.
  *
  * This function is called when the codec driver calls snd_soc_new_pcms(),
- * once for each .dai_link in the machine driver's snd_soc_machine
+ * once for each .dai_link in the machine driver's snd_soc_card
  * structure.
  */
 static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
@@ -853,6 +853,18 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info)
 }
 EXPORT_SYMBOL_GPL(fsl_dma_configure);
 
+static int __init fsl_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&fsl_soc_platform);
+}
+module_init(fsl_soc_platform_init);
+
+static void __exit fsl_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&fsl_soc_platform);
+}
+module_exit(fsl_soc_platform_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 157a7895ffa1..c6d6eb71dc1d 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
  * If this is the first stream open, then grab the IRQ and program most of
  * the SSI registers.
  */
-static int fsl_ssi_startup(struct snd_pcm_substream *substream)
+static int fsl_ssi_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
  * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
  * clock master.
  */
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
+static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
  * The DMA channel is in external master start and pause mode, which
  * means the SSI completely controls the flow of data.
  */
-static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the SSI if there are no other substreams open.
  */
-static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.prepare = fsl_ssi_prepare,
 		.shutdown = fsl_ssi_shutdown,
 		.trigger = fsl_ssi_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = fsl_ssi_set_sysclk,
 		.set_fmt = fsl_ssi_set_fmt,
 	},
@@ -671,6 +673,14 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	fsl_ssi_dai->private_data = ssi_private;
 	fsl_ssi_dai->name = ssi_private->name;
 	fsl_ssi_dai->id = ssi_info->id;
+	fsl_ssi_dai->dev = ssi_info->dev;
+
+	ret = snd_soc_register_dai(fsl_ssi_dai);
+	if (ret != 0) {
+		dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
+		kfree(fsl_ssi_dai);
+		return NULL;
+	}
 
 	return fsl_ssi_dai;
 }
@@ -688,6 +698,8 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 
 	device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
 
+	snd_soc_unregister_dai(&ssi_private->cpu_dai);
+
 	kfree(ssi_private);
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 94a02eaa4825..9eb1ce185bd0 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
  * If this is the first stream open, then grab the IRQ and program most of
  * the PSC registers.
  */
-static int psc_i2s_startup(struct snd_pcm_substream *substream)
+static int psc_i2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
@@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
  * This function is called by ALSA to start, stop, pause, and resume the DMA
  * transfer of data.
  */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the PSC if there are no other substreams open.
  */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -464,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai psc_i2s_dai_template = {
-	.type = SND_SOC_DAI_I2S,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -483,8 +487,6 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.hw_free = psc_i2s_hw_free,
 		.shutdown = psc_i2s_shutdown,
 		.trigger = psc_i2s_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = psc_i2s_set_sysclk,
 		.set_fmt = psc_i2s_set_fmt,
 	},
@@ -826,6 +828,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	if (rc)
 		dev_info(psc_i2s->dev, "error creating sysfs files\n");
 
+	snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
+
 	/* Tell the ASoC OF helpers about it */
 	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
 				     &psc_i2s->dai);
@@ -839,6 +843,8 @@ static int __devexit psc_i2s_of_remove(struct of_device *op)
 
 	dev_dbg(&op->dev, "psc_i2s_remove()\n");
 
+	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+
 	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
 	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 94f89debde1f..bcec3f60bad9 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -29,7 +29,7 @@
 struct mpc8610_hpcd_data {
 	struct snd_soc_device sound_devdata;
 	struct snd_soc_dai_link dai;
-	struct snd_soc_machine machine;
+	struct snd_soc_card machine;
 	unsigned int dai_format;
 	unsigned int codec_clk_direction;
 	unsigned int cpu_clk_direction;
@@ -185,7 +185,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 /**
  * mpc8610_hpcd_machine: ASoC machine data
  */
-static struct snd_soc_machine mpc8610_hpcd_machine = {
+static struct snd_soc_card mpc8610_hpcd_machine = {
 	.probe = mpc8610_hpcd_machine_probe,
 	.remove = mpc8610_hpcd_machine_remove,
 	.name = "MPC8610 HPCD",
@@ -465,9 +465,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 		goto error;
 	}
 
-	machine_data->sound_devdata.machine = &mpc8610_hpcd_machine;
+	machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
 	machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
-	machine_data->sound_devdata.platform = &fsl_soc_platform;
+	machine_data->machine.platform = &fsl_soc_platform;
 
 	sound_device->dev.platform_data = machine_data;
 
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
index 0382fdac51cd..8bc5cd9e972f 100644
--- a/sound/soc/fsl/soc-of-simple.c
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -31,7 +31,7 @@ struct of_snd_soc_device {
 	int id;
 	struct list_head list;
 	struct snd_soc_device device;
-	struct snd_soc_machine machine;
+	struct snd_soc_card card;
 	struct snd_soc_dai_link dai_link;
 	struct platform_device *pdev;
 	struct device_node *platform_node;
@@ -58,9 +58,9 @@ of_snd_soc_get_device(struct device_node *codec_node)
 	/* Initialize the structure and add it to the global list */
 	of_soc->codec_node = codec_node;
 	of_soc->id = of_snd_soc_next_index++;
-	of_soc->machine.dai_link = &of_soc->dai_link;
-	of_soc->machine.num_links = 1;
-	of_soc->device.machine = &of_soc->machine;
+	of_soc->card.dai_link = &of_soc->dai_link;
+	of_soc->card.num_links = 1;
+	of_soc->device.card = &of_soc->card;
 	of_soc->dai_link.ops = &of_snd_soc_ops;
 	list_add(&of_soc->list, &of_snd_soc_device_list);
 
@@ -158,8 +158,8 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform,
 
 	of_soc->platform_node = node;
 	of_soc->dai_link.cpu_dai = cpu_dai;
-	of_soc->device.platform = platform;
-	of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+	of_soc->card.platform = platform;
+	of_soc->card.name = of_soc->dai_link.cpu_dai->name;
 
 	/* Now try to register the SoC device */
 	of_snd_soc_register_device(of_soc);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 8b7766b998d7..a7b1d77b2105 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,6 +1,6 @@
 config SND_OMAP_SOC
 	tristate "SoC Audio for the Texas Instruments OMAP chips"
-	depends on ARCH_OMAP && SND_SOC
+	depends on ARCH_OMAP
 
 config SND_OMAP_SOC_MCBSP
 	tristate
@@ -21,3 +21,36 @@ config SND_OMAP_SOC_OSK5912
 	select SND_SOC_TLV320AIC23
 	help
 	  Say Y if you want to add support for SoC audio on osk5912.
+
+config SND_OMAP_SOC_OVERO
+	tristate "SoC Audio support for Gumstix Overo"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the Gumstix Overo.
+
+config SND_OMAP_SOC_OMAP2EVM
+	tristate "SoC Audio support for OMAP2EVM board"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the omap2evm board.
+
+config SND_OMAP_SOC_SDP3430
+	tristate "SoC Audio support for Texas Instruments SDP3430"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on Texas Instruments
+	  SDP3430.
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+	tristate "SoC Audio support for OMAP3 Pandora"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index e09d1f297f64..76fedd96e365 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -8,6 +8,14 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
 snd-soc-osk5912-objs := osk5912.o
+snd-soc-overo-objs := overo.o
+snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-omap3pandora-objs := omap3pandora.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
+obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index fae3ad36e0bf..25593fee9121 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -70,9 +70,13 @@ static void n810_ext_control(struct snd_soc_codec *codec)
 
 static int n810_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->socdev->codec;
 
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+
 	n810_ext_control(codec);
 	return clk_enable(sys_clkout2);
 }
@@ -282,8 +286,9 @@ static struct snd_soc_dai_link n810_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_n810 = {
+static struct snd_soc_card snd_soc_n810 = {
 	.name = "N810",
+	.platform = &omap_soc_platform,
 	.dai_link = &n810_dai,
 	.num_links = 1,
 };
@@ -298,8 +303,7 @@ static struct aic3x_setup_data n810_aic33_setup = {
 
 /* Audio subsystem */
 static struct snd_soc_device n810_snd_devdata = {
-	.machine = &snd_soc_machine_n810,
-	.platform = &omap_soc_platform,
+	.card = &snd_soc_n810,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &n810_aic33_setup,
 };
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8485a8a9d0ff..ec5e18a78758 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -36,9 +36,7 @@
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 
-#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_44100 | \
-				 SNDRV_PCM_RATE_48000 | \
-				 SNDRV_PCM_RATE_KNOT)
+#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_8000_96000)
 
 struct omap_mcbsp_data {
 	unsigned int			bus_id;
@@ -140,7 +138,8 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
 static const unsigned long omap34xx_mcbsp_port[][2] = {};
 #endif
 
-static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -153,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 	return err;
 }
 
-static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -165,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
 	}
 }
 
-static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -194,14 +195,15 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
-				    struct snd_pcm_hw_params *params)
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
-	int wlen;
+	int wlen, channels;
 	unsigned long port;
 
 	if (cpu_class_is_omap1()) {
@@ -230,12 +232,17 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		return 0;
 	}
 
-	switch (params_channels(params)) {
+	channels = params_channels(params);
+	switch (channels) {
 	case 2:
-		/* Set 1 word per (McBPSP) frame and use dual-phase frames */
-		regs->rcr2	|= RFRLEN2(1 - 1) | RPHASE;
+		/* Use dual-phase frames */
+		regs->rcr2	|= RPHASE;
+		regs->xcr2	|= XPHASE;
+	case 1:
+		/* Set 1 word per (McBSP) frame */
+		regs->rcr2	|= RFRLEN2(1 - 1);
 		regs->rcr1	|= RFRLEN1(1 - 1);
-		regs->xcr2	|= XFRLEN2(1 - 1) | XPHASE;
+		regs->xcr2	|= XFRLEN2(1 - 1);
 		regs->xcr1	|= XFRLEN1(1 - 1);
 		break;
 	default:
@@ -263,9 +270,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		regs->srgr2	|= FPER(wlen * 2 - 1);
 		regs->srgr1	|= FWID(wlen - 1);
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		regs->srgr2	|= FPER(wlen * 2 - 1);
-		regs->srgr1	|= FWID(wlen * 2 - 2);
+	case SND_SOC_DAIFMT_DSP_B:
+		regs->srgr2	|= FPER(wlen * channels - 1);
+		regs->srgr1	|= FWID(wlen * channels - 2);
 		break;
 	}
 
@@ -302,7 +309,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
 		regs->xcr2      |= XDATDLY(0);
@@ -452,17 +459,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 #define OMAP_MCBSP_DAI_BUILDER(link_id)				\
 {								\
-	.name = "omap-mcbsp-dai-(link_id)",			\
+	.name = "omap-mcbsp-dai-"#link_id,			\
 	.id = (link_id),					\
-	.type = SND_SOC_DAI_I2S,				\
 	.playback = {						\
-		.channels_min = 2,				\
+		.channels_min = 1,				\
 		.channels_max = 2,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
 	.capture = {						\
-		.channels_min = 2,				\
+		.channels_min = 1,				\
 		.channels_max = 2,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
@@ -472,8 +478,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.shutdown = omap_mcbsp_dai_shutdown,		\
 		.trigger = omap_mcbsp_dai_trigger,		\
 		.hw_params = omap_mcbsp_dai_hw_params,		\
-	},							\
-	.dai_ops = {						\
 		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
 		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
 		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
@@ -495,6 +499,19 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
 
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
+static int __init snd_omap_mcbsp_init(void)
+{
+	return snd_soc_register_dais(omap_mcbsp_dai,
+				     ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_init(snd_omap_mcbsp_init);
+
+static void __exit snd_omap_mcbsp_exit(void)
+{
+	snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_exit(snd_omap_mcbsp_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index acd68efb2b75..b0362dfd5b71 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	prtd->dma_data = dma_data;
 	err = omap_request_dma(dma_data->dma_req, dma_data->name,
 			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
-	if (!err & !cpu_is_omap1510()) {
+	if (!err && !cpu_is_omap1510()) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -354,6 +354,18 @@ struct snd_soc_platform omap_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(omap_soc_platform);
 
+static int __init omap_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&omap_soc_platform);
+}
+module_init(omap_soc_platform_init);
+
+static void __exit omap_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&omap_soc_platform);
+}
+module_exit(omap_soc_platform_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
new file mode 100644
index 000000000000..0c2322dcf02a
--- /dev/null
+++ b/sound/soc/omap/omap2evm.c
@@ -0,0 +1,151 @@
+/*
+ * omap2evm.c  --  SoC audio machine driver for omap2evm board
+ *
+ * Author: Arun KS <arunks@mistralsolutions.com>
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap2evm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap2evm_ops = {
+	.hw_params = omap2evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap2evm_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &omap2evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap2evm = {
+	.name = "omap2evm",
+	.platform = &omap_soc_platform,
+	.dai_link = &omap2evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap2evm_snd_devdata = {
+	.card = &snd_soc_omap2evm,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap2evm_snd_device;
+
+static int __init omap2evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap2evm()) {
+		pr_debug("Not omap2evm!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "omap2evm SoC init\n");
+
+	omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap2evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata);
+	omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev;
+	*(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(omap2evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap2evm_snd_device);
+
+	return ret;
+}
+module_init(omap2evm_soc_init);
+
+static void __exit omap2evm_soc_exit(void)
+{
+	platform_device_unregister(omap2evm_snd_device);
+}
+module_exit(omap2evm_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC omap2evm");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
new file mode 100644
index 000000000000..fd24a4acd2f5
--- /dev/null
+++ b/sound/soc/omap/omap3beagle.c
@@ -0,0 +1,149 @@
+/*
+ * omap3beagle.c  --  SoC audio for OMAP3 Beagle
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3beagle_ops = {
+	.hw_params = omap3beagle_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3beagle_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &omap3beagle_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3beagle = {
+	.name = "omap3beagle",
+	.platform = &omap_soc_platform,
+	.dai_link = &omap3beagle_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3beagle_snd_devdata = {
+	.card = &snd_soc_omap3beagle,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3beagle_snd_device;
+
+static int __init omap3beagle_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3_beagle()) {
+		pr_debug("Not OMAP3 Beagle!\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3 Beagle SoC init\n");
+
+	omap3beagle_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap3beagle_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap3beagle_snd_device, &omap3beagle_snd_devdata);
+	omap3beagle_snd_devdata.dev = &omap3beagle_snd_device->dev;
+	*(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(omap3beagle_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap3beagle_snd_device);
+
+	return ret;
+}
+
+static void __exit omap3beagle_soc_exit(void)
+{
+	platform_device_unregister(omap3beagle_snd_device);
+}
+
+module_init(omap3beagle_soc_init);
+module_exit(omap3beagle_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
new file mode 100644
index 000000000000..bd91594496b1
--- /dev/null
+++ b/sound/soc/omap/omap3pandora.c
@@ -0,0 +1,311 @@
+/*
+ * omap3pandora.c  --  SoC audio for Pandora Handheld Console
+ *
+ * Author: Gražvydas Ignotas <notasas@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define OMAP3_PANDORA_DAC_POWER_GPIO	118
+#define OMAP3_PANDORA_AMP_POWER_GPIO	14
+
+#define PREFIX "ASoC omap3pandora: "
+
+static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
+	struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set codec system clock\n");
+		return ret;
+	}
+
+	/* Set McBSP clock to external */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set cpu system clock\n");
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set SRG clock divider\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+					  SND_SOC_DAIFMT_I2S |
+					  SND_SOC_DAIFMT_IB_NF |
+					  SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+					  SND_SOC_DAIFMT_I2S |
+					  SND_SOC_DAIFMT_NB_NF |
+					  SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
+	} else {
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+		mdelay(1);
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+	}
+
+	return 0;
+}
+
+/*
+ * Audio paths on Pandora board:
+ *
+ *  |O| ---> PCM DAC +-> AMP -> Headphone Jack
+ *  |M|         A    +--------> Line Out
+ *  |A| <~~clk~~+
+ *  |P| <--- TWL4030 <--------- Line In and MICs
+ */
+static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
+			   0, 0, NULL, 0, omap3pandora_hp_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+};
+
+static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_MIC("Mic (external)", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"Headphone Amplifier", NULL, "PCM DAC"},
+	{"Line Out", NULL, "PCM DAC"},
+	{"Headphone Jack", NULL, "Headphone Amplifier"},
+};
+
+static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
+	{"INL", NULL, "Line In"},
+	{"INR", NULL, "Line In"},
+	{"INL", NULL, "Mic (Internal)"},
+	{"INR", NULL, "Mic (external)"},
+};
+
+static int omap3pandora_out_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
+				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_add_routes(codec, omap3pandora_out_map,
+		ARRAY_SIZE(omap3pandora_out_map));
+
+	return snd_soc_dapm_sync(codec);
+}
+
+static int omap3pandora_in_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
+				ARRAY_SIZE(omap3pandora_in_dapm_widgets));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_add_routes(codec, omap3pandora_in_map,
+		ARRAY_SIZE(omap3pandora_in_map));
+
+	return snd_soc_dapm_sync(codec);
+}
+
+static struct snd_soc_ops omap3pandora_out_ops = {
+	.hw_params = omap3pandora_out_hw_params,
+};
+
+static struct snd_soc_ops omap3pandora_in_ops = {
+	.hw_params = omap3pandora_in_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+	{
+		.name = "PCM1773",
+		.stream_name = "HiFi Out",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &twl4030_dai,
+		.ops = &omap3pandora_out_ops,
+		.init = omap3pandora_out_init,
+	}, {
+		.name = "TWL4030",
+		.stream_name = "Line/Mic In",
+		.cpu_dai = &omap_mcbsp_dai[1],
+		.codec_dai = &twl4030_dai,
+		.ops = &omap3pandora_in_ops,
+		.init = omap3pandora_in_init,
+	}
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_omap3pandora = {
+	.name = "omap3pandora",
+	.platform = &omap_soc_platform,
+	.dai_link = omap3pandora_dai,
+	.num_links = ARRAY_SIZE(omap3pandora_dai),
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3pandora_snd_data = {
+	.card = &snd_soc_card_omap3pandora,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3pandora_snd_device;
+
+static int __init omap3pandora_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3_pandora()) {
+		pr_debug(PREFIX "Not OMAP3 Pandora\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3 Pandora SoC init\n");
+
+	ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get DAC power GPIO\n");
+		return ret;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
+		goto fail0;
+	}
+
+	ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get amp power GPIO\n");
+		goto fail0;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set amp power GPIO direction\n");
+		goto fail1;
+	}
+
+	omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
+	if (omap3pandora_snd_device == NULL) {
+		pr_err(PREFIX "Platform device allocation failed\n");
+		ret = -ENOMEM;
+		goto fail1;
+	}
+
+	platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data);
+	omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev;
+	*(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */
+	*(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */
+
+	ret = platform_device_add(omap3pandora_snd_device);
+	if (ret) {
+		pr_err(PREFIX "Unable to add platform device\n");
+		goto fail2;
+	}
+
+	return 0;
+
+fail2:
+	platform_device_put(omap3pandora_snd_device);
+fail1:
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+fail0:
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+	return ret;
+}
+module_init(omap3pandora_soc_init);
+
+static void __exit omap3pandora_soc_exit(void)
+{
+	platform_device_unregister(omap3pandora_snd_device);
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+}
+module_exit(omap3pandora_soc_exit);
+
+MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 0fe733796898..cd41a948df7b 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -61,7 +61,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 
 	/* Set codec DAI configuration */
 	err = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_DSP_B |
 				  SND_SOC_DAIFMT_NB_IF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
@@ -71,7 +71,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 
 	/* Set cpu DAI configuration */
 	err = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_DSP_B |
 				  SND_SOC_DAIFMT_NB_IF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
@@ -143,16 +143,16 @@ static struct snd_soc_dai_link osk_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_osk = {
+static struct snd_soc_card snd_soc_card_osk = {
 	.name = "OSK5912",
+	.platform = &omap_soc_platform,
 	.dai_link = &osk_dai,
 	.num_links = 1,
 };
 
 /* Audio subsystem */
 static struct snd_soc_device osk_snd_devdata = {
-	.machine = &snd_soc_machine_osk,
-	.platform = &omap_soc_platform,
+	.card = &snd_soc_card_osk,
 	.codec_dev = &soc_codec_dev_tlv320aic23,
 };
 
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
new file mode 100644
index 000000000000..a72dc4e159e5
--- /dev/null
+++ b/sound/soc/omap/overo.c
@@ -0,0 +1,148 @@
+/*
+ * overo.c  --  SoC audio for Gumstix Overo
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int overo_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops overo_ops = {
+	.hw_params = overo_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link overo_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &overo_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_overo = {
+	.name = "overo",
+	.platform = &omap_soc_platform,
+	.dai_link = &overo_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device overo_snd_devdata = {
+	.card = &snd_soc_card_overo,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *overo_snd_device;
+
+static int __init overo_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_overo()) {
+		pr_debug("Not Overo!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "overo SoC init\n");
+
+	overo_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!overo_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(overo_snd_device, &overo_snd_devdata);
+	overo_snd_devdata.dev = &overo_snd_device->dev;
+	*(unsigned int *)overo_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(overo_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(overo_snd_device);
+
+	return ret;
+}
+module_init(overo_soc_init);
+
+static void __exit overo_soc_exit(void)
+{
+	platform_device_unregister(overo_snd_device);
+}
+module_exit(overo_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC overo");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
new file mode 100644
index 000000000000..ad97836818b1
--- /dev/null
+++ b/sound/soc/omap/sdp3430.c
@@ -0,0 +1,152 @@
+/*
+ * sdp3430.c  --  SoC audio for TI OMAP3430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * Based on:
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int sdp3430_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops sdp3430_ops = {
+	.hw_params = sdp3430_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp3430_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &sdp3430_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+	.name = "SDP3430",
+	.platform = &omap_soc_platform,
+	.dai_link = &sdp3430_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device sdp3430_snd_devdata = {
+	.machine = &snd_soc_machine_sdp3430,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *sdp3430_snd_device;
+
+static int __init sdp3430_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_3430sdp()) {
+		pr_debug("Not SDP3430!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "SDP3430 SoC init\n");
+
+	sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!sdp3430_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
+	sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
+	*(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(sdp3430_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(sdp3430_snd_device);
+
+	return ret;
+}
+module_init(sdp3430_soc_init);
+
+static void __exit sdp3430_soc_exit(void)
+{
+	platform_device_unregister(sdp3430_snd_device);
+}
+module_exit(sdp3430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP3430");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f8c1cdd940ac..f82e10699471 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -21,6 +21,9 @@ config SND_PXA2XX_SOC_AC97
 config SND_PXA2XX_SOC_I2S
 	tristate
 
+config SND_PXA_SOC_SSP
+	tristate
+
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
 	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -75,3 +78,22 @@ config SND_PXA2XX_SOC_EM_X270
 	help
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
+
+config SND_PXA2XX_SOC_PALM27X
+	bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Palm T|X, T5 or LifeDrive handheld computer.
+
+config SND_SOC_ZYLONITE
+	tristate "SoC Audio support for Marvell Zylonite"
+	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+	select SND_PXA2XX_SOC_AC97
+	select SND_PXA_SOC_SSP
+	select SND_SOC_WM9713
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Zylonite reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 5bc8edf9dca9..08a9f2797729 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -2,10 +2,12 @@
 snd-soc-pxa2xx-objs := pxa2xx-pcm.o
 snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
 snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+snd-soc-pxa-ssp-objs := pxa-ssp.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
 obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
@@ -14,6 +16,8 @@ snd-soc-tosa-objs := tosa.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
+snd-soc-zylonite-objs := zylonite.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -21,3 +25,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 2718eaf7895f..1ba25a559524 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -108,15 +108,11 @@ static int corgi_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on corgi */
-static int corgi_shutdown(struct snd_pcm_substream *substream)
+static void corgi_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
-
 	/* set = unmute headphone */
 	gpio_set_value(CORGI_GPIO_MUTE_L, 1);
 	gpio_set_value(CORGI_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int corgi_hw_params(struct snd_pcm_substream *substream,
@@ -314,8 +310,9 @@ static struct snd_soc_dai_link corgi_dai = {
 };
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &corgi_dai,
 	.num_links = 1,
 };
@@ -328,8 +325,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_corgi,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
 };
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 6781c5be242f..2e3386dfa0f0 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -29,7 +29,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine e800;
+static struct snd_soc_card e800;
 
 static struct snd_soc_dai_link e800_dai[] = {
 {
@@ -40,15 +40,15 @@ static struct snd_soc_dai_link e800_dai[] = {
 },
 };
 
-static struct snd_soc_machine e800 = {
+static struct snd_soc_card e800 = {
 	.name = "Toshiba e800",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = e800_dai,
 	.num_links = ARRAY_SIZE(e800_dai),
 };
 
 static struct snd_soc_device e800_snd_devdata = {
-	.machine = &e800,
-	.platform = &pxa2xx_soc_platform,
+	.card = &e800,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index e6ff6929ab4b..fe4a729ea648 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -23,7 +23,6 @@
 #include <linux/moduleparam.h>
 #include <linux/device.h>
 
-#include <sound/driver.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -53,15 +52,15 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 	},
 };
 
-static struct snd_soc_machine em_x270 = {
+static struct snd_soc_card em_x270 = {
 	.name = "EM-X270",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = em_x270_dai,
 	.num_links = ARRAY_SIZE(em_x270_dai),
 };
 
 static struct snd_soc_device em_x270_snd_devdata = {
-	.machine = &em_x270,
-	.platform = &pxa2xx_soc_platform,
+	.card = &em_x270,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
new file mode 100644
index 000000000000..4a9cf3083af0
--- /dev/null
+++ b/sound/soc/pxa/palm27x.c
@@ -0,0 +1,269 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/palmasoc.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int palm27x_jack_func = 1;
+static int palm27x_spk_func = 1;
+static int palm27x_ep_gpio = -1;
+
+static void palm27x_ext_control(struct snd_soc_codec *codec)
+{
+	if (!palm27x_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(codec, "Speaker");
+
+	if (!palm27x_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int palm27x_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	palm27x_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops palm27x_ops = {
+	.startup = palm27x_startup,
+};
+
+static irqreturn_t palm27x_interrupt(int irq, void *v)
+{
+	palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
+	palm27x_jack_func = !palm27x_spk_func;
+	return IRQ_HANDLED;
+}
+
+static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_jack_func;
+	return 0;
+}
+
+static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_jack_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_spk_func;
+	return 0;
+}
+
+static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_spk_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+/* PalmTX machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to ROUT2, LOUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+};
+
+static const char *jack_function[] = {"Headphone", "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum palm27x_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new palm27x_controls[] = {
+	SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
+		palm27x_set_jack),
+	SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
+		palm27x_set_spk),
+};
+
+static int palm27x_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+	/* add palm27x specific controls */
+	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&palm27x_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add palm27x specific widgets */
+	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+				ARRAY_SIZE(palm27x_dapm_widgets));
+
+	/* set up palm27x specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = palm27x_ac97_init,
+	.ops = &palm27x_ops,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.ops = &palm27x_ops,
+},
+};
+
+static struct snd_soc_card palm27x_asoc = {
+	.name = "Palm/PXA27x",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = palm27x_dai,
+	.num_links = ARRAY_SIZE(palm27x_dai),
+};
+
+static struct snd_soc_device palm27x_snd_devdata = {
+	.card = &palm27x_asoc,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *palm27x_snd_device;
+
+static int __init palm27x_asoc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_palmtx() || machine_is_palmt5() ||
+		machine_is_palmld()))
+		return -ENODEV;
+
+	ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_input(palm27x_ep_gpio);
+	if (ret)
+		goto err_alloc;
+
+	if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
+			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+			"Headphone jack", NULL))
+		goto err_alloc;
+
+	palm27x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!palm27x_snd_device) {
+		ret = -ENOMEM;
+		goto err_dev;
+	}
+
+	platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
+	palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+	ret = platform_device_add(palm27x_snd_device);
+
+	if (ret != 0)
+		goto put_device;
+
+	return 0;
+
+put_device:
+	platform_device_put(palm27x_snd_device);
+err_dev:
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+err_alloc:
+	gpio_free(palm27x_ep_gpio);
+
+	return ret;
+}
+
+static void __exit palm27x_asoc_exit(void)
+{
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+	gpio_free(palm27x_ep_gpio);
+	platform_device_unregister(palm27x_snd_device);
+}
+
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+{
+	palm27x_ep_gpio = data->jack_gpio;
+}
+
+module_init(palm27x_asoc_init);
+module_exit(palm27x_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 4d9930c52789..6e9827189fff 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,8 +276,9 @@ static struct snd_soc_dai_link poodle_dai = {
 };
 
 /* poodle audio machine driver */
-static struct snd_soc_machine snd_soc_machine_poodle = {
+static struct snd_soc_card snd_soc_poodle = {
 	.name = "Poodle",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &poodle_dai,
 	.num_links = 1,
 };
@@ -290,8 +291,7 @@ static struct wm8731_setup_data poodle_wm8731_setup = {
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
-	.machine = &snd_soc_machine_poodle,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_poodle,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &poodle_wm8731_setup,
 };
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 000000000000..73cb6b4c2f2d
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,931 @@
+#define DEBUG
+/*
+ * pxa-ssp.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright 2005,2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ * TODO:
+ *  o Test network mode for > 16bit sample size
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/regs-ssp.h>
+#include <mach/audio.h>
+#include <mach/ssp.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+	struct ssp_dev dev;
+	unsigned int sysclk;
+	int dai_fmt;
+#ifdef CONFIG_PM
+	struct ssp_state state;
+#endif
+};
+
+#define PXA2xx_SSP1_BASE	0x41000000
+#define PXA27x_SSP2_BASE	0x41700000
+#define PXA27x_SSP3_BASE	0x41900000
+#define PXA3xx_SSP4_BASE	0x41a00000
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
+	.name			= "SSP1 PCM Mono out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
+	.name			= "SSP1 PCM Mono in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
+	.name			= "SSP1 PCM Stereo out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
+	.name			= "SSP1 PCM Stereo in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
+	.name			= "SSP2 PCM Mono out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
+	.name			= "SSP2 PCM Mono in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
+	.name			= "SSP2 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
+	.name			= "SSP2 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
+	.name			= "SSP3 PCM Mono out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
+	.name			= "SSP3 PCM Mono in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
+	.name			= "SSP3 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
+	.name			= "SSP3 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
+	.name			= "SSP4 PCM Mono out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
+	.name			= "SSP4 PCM Mono in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
+	.name			= "SSP4 PCM Stereo out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
+	.name			= "SSP4 PCM Stereo in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+		 ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1),
+		 ssp_read_reg(ssp, SSTO));
+
+	dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+		 ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR),
+		 ssp_read_reg(ssp, SSACD));
+}
+
+static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
+	{
+		&pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
+		&pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
+		&pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
+		&pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
+		&pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
+	},
+};
+
+static int pxa_ssp_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	int ret = 0;
+
+	if (!cpu_dai->active) {
+		ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
+		if (ret < 0)
+			return ret;
+		ssp_disable(&priv->dev);
+	}
+	return ret;
+}
+
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active) {
+		ssp_disable(&priv->dev);
+		ssp_exit(&priv->dev);
+	}
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssp_save_state(&priv->dev, &priv->state);
+	clk_disable(priv->dev.ssp->clk);
+	return 0;
+}
+
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	clk_enable(priv->dev.ssp->clk);
+	ssp_restore_state(&priv->dev, &priv->state);
+	ssp_enable(&priv->dev);
+
+	return 0;
+}
+
+#else
+#define pxa_ssp_suspend	NULL
+#define pxa_ssp_resume	NULL
+#endif
+
+/**
+ * ssp_set_clkdiv - set SSP clock divider
+ * @div: serial clock rate divider
+ */
+static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+{
+	struct ssp_device *ssp = dev->ssp;
+	u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+
+	ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+	int clk_id, unsigned int freq, int dir)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+
+	dev_dbg(&ssp->pdev->dev,
+		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+		cpu_dai->id, clk_id, freq);
+
+	switch (clk_id) {
+	case PXA_SSP_CLK_NET_PLL:
+		sscr0 |= SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_PLL:
+		/* Internal PLL is fixed */
+		if (cpu_is_pxa25x())
+			priv->sysclk = 1843200;
+		else
+			priv->sysclk = 13000000;
+		break;
+	case PXA_SSP_CLK_EXT:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_ECS;
+		break;
+	case PXA_SSP_CLK_NET:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_NCS | SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_AUDIO:
+		priv->sysclk = 0;
+		ssp_set_scr(&priv->dev, 1);
+		sscr0 |= SSCR0_ADC;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	/* The SSP clock must be disabled when changing SSP clock mode
+	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
+	if (!cpu_is_pxa3xx())
+		clk_disable(priv->dev.ssp->clk);
+	val = ssp_read_reg(ssp, SSCR0) | sscr0;
+	ssp_write_reg(ssp, SSCR0, val);
+	if (!cpu_is_pxa3xx())
+		clk_enable(priv->dev.ssp->clk);
+
+	return 0;
+}
+
+/*
+ * Set the SSP clock dividers.
+ */
+static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (div_id) {
+	case PXA_SSP_AUDIO_DIV_ACDS:
+		val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_AUDIO_DIV_SCDB:
+		val = ssp_read_reg(ssp, SSACD);
+		val &= ~SSACD_SCDB;
+#if defined(CONFIG_PXA3xx)
+		if (cpu_is_pxa3xx())
+			val &= ~SSACD_SCDX8;
+#endif
+		switch (div) {
+		case PXA_SSP_CLK_SCDB_1:
+			val |= SSACD_SCDB;
+			break;
+		case PXA_SSP_CLK_SCDB_4:
+			break;
+#if defined(CONFIG_PXA3xx)
+		case PXA_SSP_CLK_SCDB_8:
+			if (cpu_is_pxa3xx())
+				val |= SSACD_SCDX8;
+			else
+				return -EINVAL;
+			break;
+#endif
+		default:
+			return -EINVAL;
+		}
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_DIV_SCR:
+		ssp_set_scr(&priv->dev, div);
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
+ */
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
+	int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70;
+
+#if defined(CONFIG_PXA3xx)
+	if (cpu_is_pxa3xx())
+		ssp_write_reg(ssp, SSACDD, 0);
+#endif
+
+	switch (freq_out) {
+	case 5622000:
+		break;
+	case 11345000:
+		ssacd |= (0x1 << 4);
+		break;
+	case 12235000:
+		ssacd |= (0x2 << 4);
+		break;
+	case 14857000:
+		ssacd |= (0x3 << 4);
+		break;
+	case 32842000:
+		ssacd |= (0x4 << 4);
+		break;
+	case 48000000:
+		ssacd |= (0x5 << 4);
+		break;
+	case 0:
+		/* Disable */
+		break;
+
+	default:
+#ifdef CONFIG_PXA3xx
+		/* PXA3xx has a clock ditherer which can be used to generate
+		 * a wider range of frequencies - calculate a value for it.
+		 */
+		if (cpu_is_pxa3xx()) {
+			u32 val;
+			u64 tmp = 19968;
+			tmp *= 1000000;
+			do_div(tmp, freq_out);
+			val = tmp;
+
+			val = (val << 16) | 64;;
+			ssp_write_reg(ssp, SSACDD, val);
+
+			ssacd |= (0x6 << 4);
+
+			dev_dbg(&ssp->pdev->dev,
+				"Using SSACDD %x to supply %dHz\n",
+				val, freq_out);
+			break;
+		}
+#endif
+
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSACD, ssacd);
+
+	return 0;
+}
+
+/*
+ * Set the active slots in TDM/Network mode
+ */
+static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+	unsigned int mask, int slots)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7);
+
+	/* set number of active slots */
+	sscr0 |= SSCR0_SlotsPerFrm(slots);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* set active slot mask */
+	ssp_write_reg(ssp, SSTSA, mask);
+	ssp_write_reg(ssp, SSRSA, mask);
+	return 0;
+}
+
+/*
+ * Tristate the SSP DAI lines
+ */
+static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
+	int tristate)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr1;
+
+	sscr1 = ssp_read_reg(ssp, SSCR1);
+	if (tristate)
+		sscr1 &= ~SSCR1_TTE;
+	else
+		sscr1 |= SSCR1_TTE;
+	ssp_write_reg(ssp, SSCR1, sscr1);
+
+	return 0;
+}
+
+/*
+ * Set up the SSP DAI format.
+ * The SSP Port must be inactive before calling this function as the
+ * physical interface format is changed.
+ */
+static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+	u32 sscr1;
+	u32 sspsp;
+
+	/* reset port settings */
+	sscr0 = ssp_read_reg(ssp, SSCR0) &
+		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
+	sspsp = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		sscr1 |= SSCR1_SCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+		sspsp |= SSPSP_FSRT;
+	case SND_SOC_DAIFMT_DSP_B:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	dump_registers(ssp);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	priv->dai_fmt = fmt;
+
+	return 0;
+}
+
+/*
+ * Set the SSP audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int dma = 0, chn = params_channels(params);
+	u32 sscr0;
+	u32 sspsp;
+	int width = snd_pcm_format_physical_width(params_format(params));
+
+	/* select correct DMA params */
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		dma = 1; /* capture DMA offset is 1,3 */
+	if (chn == 2)
+		dma += 2; /* stereo DMA offset is 2, mono is 0 */
+	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
+
+	dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+
+	/* we can only change the settings if the port is not in use */
+	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+		return 0;
+
+	/* clear selected SSP bits */
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* bit size */
+	sscr0 = ssp_read_reg(ssp, SSCR0);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+#ifdef CONFIG_PXA3xx
+		if (cpu_is_pxa3xx())
+			sscr0 |= SSCR0_FPCKE;
+#endif
+		sscr0 |= SSCR0_DataSize(16);
+		if (params_channels(params) > 1)
+			sscr0 |= SSCR0_EDSS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+		/* we must be in network mode (2 slots) for 24 bit stereo */
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+		/* we must be in network mode (2 slots) for 32 bit stereo */
+		break;
+	}
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Cleared when the DAI format is set */
+		sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+		ssp_write_reg(ssp, SSPSP, sspsp);
+		break;
+	default:
+		break;
+	}
+
+	/* We always use a network mode so we always require TDM slots
+	 * - complain loudly and fail if they've not been set up yet.
+	 */
+	if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+		return -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return 0;
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret = 0;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		val = ssp_read_reg(ssp, SSSR);
+		ssp_write_reg(ssp, SSSR, val);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ssp_disable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return ret;
+}
+
+static int pxa_ssp_probe(struct platform_device *pdev,
+			    struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv;
+	int ret;
+
+	priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+	if (priv->dev.ssp == NULL) {
+		ret = -ENODEV;
+		goto err_priv;
+	}
+
+	dai->private_data = priv;
+
+	return 0;
+
+err_priv:
+	kfree(priv);
+	return ret;
+}
+
+static void pxa_ssp_remove(struct platform_device *pdev,
+			      struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv = dai->private_data;
+	ssp_free(priv->dev.ssp);
+}
+
+#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+			  SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |	\
+			  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |	\
+			  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			    SNDRV_PCM_FMTBIT_S24_LE |	\
+			    SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai pxa_ssp_dai[] = {
+	{
+		.name = "pxa2xx-ssp1",
+		.id = 0,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			 .channels_min = 1,
+			 .channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{	.name = "pxa2xx-ssp2",
+		.id = 1,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp3",
+		.id = 2,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp4",
+		.id = 3,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+};
+EXPORT_SYMBOL_GPL(pxa_ssp_dai);
+
+static int __init pxa_ssp_init(void)
+{
+	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_init(pxa_ssp_init);
+
+static void __exit pxa_ssp_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_exit(pxa_ssp_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h
new file mode 100644
index 000000000000..91deadd55675
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,47 @@
+/*
+ * ASoC PXA SSP port support
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA_SSP_H
+#define _PXA_SSP_H
+
+/* pxa DAI SSP IDs */
+#define PXA_DAI_SSP1			0
+#define PXA_DAI_SSP2			1
+#define PXA_DAI_SSP3			2
+#define PXA_DAI_SSP4			3
+
+/* SSP clock sources */
+#define PXA_SSP_CLK_PLL	0
+#define PXA_SSP_CLK_EXT	1
+#define PXA_SSP_CLK_NET	2
+#define PXA_SSP_CLK_AUDIO	3
+#define PXA_SSP_CLK_NET_PLL	4
+
+/* SSP audio dividers */
+#define PXA_SSP_AUDIO_DIV_ACDS		0
+#define PXA_SSP_AUDIO_DIV_SCDB		1
+#define PXA_SSP_DIV_SCR				2
+
+/* SSP ACDS audio dividers values */
+#define PXA_SSP_CLK_AUDIO_DIV_1		0
+#define PXA_SSP_CLK_AUDIO_DIV_2		1
+#define PXA_SSP_CLK_AUDIO_DIV_4		2
+#define PXA_SSP_CLK_AUDIO_DIV_8		3
+#define PXA_SSP_CLK_AUDIO_DIV_16	4
+#define PXA_SSP_CLK_AUDIO_DIV_32	5
+
+/* SSP divider bypass */
+#define PXA_SSP_CLK_SCDB_4		0
+#define PXA_SSP_CLK_SCDB_1		1
+#define PXA_SSP_CLK_SCDB_8		2
+
+#define PXA_SSP_PLL_OUT  0
+
+extern struct snd_soc_dai pxa_ssp_dai[4];
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a7a3a9c5c6ff..780db6757ad2 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -87,14 +87,12 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
 };
 
 #ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_suspend();
 }
 
-static int pxa2xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_resume();
 }
@@ -117,7 +115,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev,
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -131,7 +130,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -145,7 +145,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -170,7 +171,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = pxa2xx_ac97_probe,
 	.remove = pxa2xx_ac97_remove,
 	.suspend = pxa2xx_ac97_suspend,
@@ -193,7 +194,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-aux",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Aux Playback",
 		.channels_min = 1,
@@ -212,7 +213,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 2,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
@@ -227,6 +228,18 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __init pxa_ac97_init(void)
+{
+	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_init(pxa_ac97_init);
+
+static void __exit pxa_ac97_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_exit(pxa_ac97_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index e758034db5c3..517991fb1099 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -121,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = {
 	},
 };
 
-static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -187,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 }
 
 static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -248,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			      struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -269,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
 	return ret;
 }
 
-static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
 {
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		SACR1 |= SACR1_DRPL;
@@ -289,8 +293,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
 }
 
 #ifdef CONFIG_PM
-static int pxa2xx_i2s_suspend(struct platform_device *dev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -307,8 +310,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
 	return 0;
 }
 
-static int pxa2xx_i2s_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -336,7 +338,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.suspend = pxa2xx_i2s_suspend,
 	.resume = pxa2xx_i2s_resume,
 	.playback = {
@@ -353,8 +354,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.startup = pxa2xx_i2s_startup,
 		.shutdown = pxa2xx_i2s_shutdown,
 		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = pxa2xx_i2s_hw_params,
 		.set_fmt = pxa2xx_i2s_set_dai_fmt,
 		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
 	},
@@ -364,12 +364,23 @@ EXPORT_SYMBOL_GPL(pxa_i2s_dai);
 
 static int pxa2xx_i2s_probe(struct platform_device *dev)
 {
+	int ret;
+
 	clk_i2s = clk_get(&dev->dev, "I2SCLK");
-	return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	pxa_i2s_dai.dev = &dev->dev;
+	ret = snd_soc_register_dai(&pxa_i2s_dai);
+	if (ret != 0)
+		clk_put(clk_i2s);
+
+	return ret;
 }
 
 static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
 {
+	snd_soc_unregister_dai(&pxa_i2s_dai);
 	clk_put(clk_i2s);
 	clk_i2s = ERR_PTR(-ENOENT);
 	return 0;
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index afcd892cd2fa..c670d08e7c9e 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -69,7 +69,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-struct snd_pcm_ops pxa2xx_pcm_ops = {
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
 	.open		= __pxa2xx_pcm_open,
 	.close		= __pxa2xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
@@ -118,6 +118,18 @@ struct snd_soc_platform pxa2xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
+static int __init pxa2xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&pxa2xx_soc_platform);
+}
+module_init(pxa2xx_soc_platform_init);
+
+static void __exit pxa2xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&pxa2xx_soc_platform);
+}
+module_exit(pxa2xx_soc_platform_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d307b6757e95..a3b9e6bdf979 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -319,8 +319,9 @@ static struct snd_soc_dai_link spitz_dai = {
 };
 
 /* spitz audio machine driver */
-static struct snd_soc_machine snd_soc_machine_spitz = {
+static struct snd_soc_card snd_soc_spitz = {
 	.name = "Spitz",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &spitz_dai,
 	.num_links = 1,
 };
@@ -333,8 +334,7 @@ static struct wm8750_setup_data spitz_wm8750_setup = {
 
 /* spitz audio subsystem */
 static struct snd_soc_device spitz_snd_devdata = {
-	.machine = &snd_soc_machine_spitz,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_spitz,
 	.codec_dev = &soc_codec_dev_wm8750,
 	.codec_data = &spitz_wm8750_setup,
 };
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index afefe41b8c46..c77194f74c9b 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -38,7 +38,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine tosa;
+static struct snd_soc_card tosa;
 
 #define TOSA_HP        0
 #define TOSA_MIC_INT   1
@@ -230,15 +230,37 @@ static struct snd_soc_dai_link tosa_dai[] = {
 },
 };
 
-static struct snd_soc_machine tosa = {
+static int tosa_probe(struct platform_device *dev)
+{
+	int ret;
+
+	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+	if (ret)
+		gpio_free(TOSA_GPIO_L_MUTE);
+
+	return ret;
+}
+
+static int tosa_remove(struct platform_device *dev)
+{
+	gpio_free(TOSA_GPIO_L_MUTE);
+	return 0;
+}
+
+static struct snd_soc_card tosa = {
 	.name = "Tosa",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
+	.probe = tosa_probe,
+	.remove = tosa_remove,
 };
 
 static struct snd_soc_device tosa_snd_devdata = {
-	.machine = &tosa,
-	.platform = &pxa2xx_soc_platform,
+	.card = &tosa,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
@@ -251,11 +273,6 @@ static int __init tosa_init(void)
 	if (!machine_is_tosa())
 		return -ENODEV;
 
-	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
-	if (ret)
-		return ret;
-	gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
-
 	tosa_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!tosa_snd_device) {
 		ret = -ENOMEM;
@@ -272,15 +289,12 @@ static int __init tosa_init(void)
 	platform_device_put(tosa_snd_device);
 
 err_alloc:
-	gpio_free(TOSA_GPIO_L_MUTE);
-
 	return ret;
 }
 
 static void __exit tosa_exit(void)
 {
 	platform_device_unregister(tosa_snd_device);
-	gpio_free(TOSA_GPIO_L_MUTE);
 }
 
 module_init(tosa_init);
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
new file mode 100644
index 000000000000..f8e9ecd589d3
--- /dev/null
+++ b/sound/soc/pxa/zylonite.c
@@ -0,0 +1,219 @@
+/*
+ * zylonite.c  --  SoC audio for Zylonite
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 of the
+ * License, or (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+static struct snd_soc_card zylonite;
+
+static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Microphone", NULL),
+	SND_SOC_DAPM_MIC("Handset Microphone", NULL),
+	SND_SOC_DAPM_SPK("Multiactor", NULL),
+	SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
+};
+
+/* Currently supported audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Headphone output connected to HPL/HPR */
+	{ "Headphone", NULL,  "HPL" },
+	{ "Headphone", NULL,  "HPR" },
+
+	/* On-board earpiece */
+	{ "Headset Earpiece", NULL, "OUT3" },
+
+	/* Headphone mic */
+	{ "MIC2A", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Headset Microphone" },
+
+	/* On-board mic */
+	{ "MIC1", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Handset Microphone" },
+
+	/* Multiactor differentially connected over SPKL/SPKR */
+	{ "Multiactor", NULL, "SPKL" },
+	{ "Multiactor", NULL, "SPKR" },
+};
+
+static int zylonite_wm9713_init(struct snd_soc_codec *codec)
+{
+	/* Currently we only support use of the AC97 clock here.  If
+	 * CLK_POUT is selected by SW15 then the clock API will need
+	 * to be used to request and enable it here.
+	 */
+
+	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+				  ARRAY_SIZE(zylonite_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Static setup for now */
+	snd_soc_dapm_enable_pin(codec, "Headphone");
+	snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int pll_out = 0;
+	unsigned int acds = 0;
+	unsigned int wm9713_div = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+		wm9713_div = 12;
+		pll_out = 2048000;
+		break;
+	case 16000:
+		wm9713_div = 6;
+		pll_out = 4096000;
+		break;
+	case 48000:
+	default:
+		wm9713_div = 2;
+		pll_out = 12288000;
+		acds = 1;
+		break;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai,
+				       params_channels(params),
+				       params_channels(params));
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	/* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
+	 * to be set instead.
+	 */
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+				     WM9713_PCMDIV(wm9713_div));
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops zylonite_voice_ops = {
+	.hw_params = zylonite_voice_hw_params,
+};
+
+static struct snd_soc_dai_link zylonite_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+	.init = zylonite_wm9713_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+},
+{
+	.name = "WM9713 Voice",
+	.stream_name = "WM9713 Voice",
+	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3],
+	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
+	.ops = &zylonite_voice_ops,
+},
+};
+
+static struct snd_soc_card zylonite = {
+	.name = "Zylonite",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = zylonite_dai,
+	.num_links = ARRAY_SIZE(zylonite_dai),
+};
+
+static struct snd_soc_device zylonite_snd_ac97_devdata = {
+	.card = &zylonite,
+	.codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *zylonite_snd_ac97_device;
+
+static int __init zylonite_init(void)
+{
+	int ret;
+
+	zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+	if (!zylonite_snd_ac97_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(zylonite_snd_ac97_device,
+			     &zylonite_snd_ac97_devdata);
+	zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev;
+
+	ret = platform_device_add(zylonite_snd_ac97_device);
+	if (ret != 0)
+		platform_device_put(zylonite_snd_ac97_device);
+
+	return ret;
+}
+
+static void __exit zylonite_exit(void)
+{
+	platform_device_unregister(zylonite_snd_ac97_device);
+}
+
+module_init(zylonite_init);
+module_exit(zylonite_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353effeb..fcd03acf10f6 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
 	  Say Y if you want to add support for SoC audio on ln2440sbc
 	  with the ALC650.
 
+config SND_S3C24XX_SOC_S3C24XX_UDA134X
+	tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
+       	depends on SND_S3C24XX_SOC
+       	select SND_S3C24XX_SOC_I2S
+       	select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 0aa5fb0b9700..96b3f3f617d4 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
+obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index 4eab2c19c454..12c71482d258 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -27,7 +27,7 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
 
-static struct snd_soc_machine ln2440sbc;
+static struct snd_soc_card ln2440sbc;
 
 static struct snd_soc_dai_link ln2440sbc_dai[] = {
 {
@@ -38,15 +38,15 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = {
 },
 };
 
-static struct snd_soc_machine ln2440sbc = {
+static struct snd_soc_card ln2440sbc = {
 	.name = "LN2440SBC",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = ln2440sbc_dai,
 	.num_links = ARRAY_SIZE(ln2440sbc_dai),
 };
 
 static struct snd_soc_device ln2440sbc_snd_ac97_devdata = {
-	.machine = &ln2440sbc,
-	.platform = &s3c24xx_soc_platform,
+	.card = &ln2440sbc,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 87ddfefcc2fb..45bb12e8ea44 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -59,7 +59,7 @@
 #define NEO_CAPTURE_HEADSET		7
 #define NEO_CAPTURE_BLUETOOTH		8
 
-static struct snd_soc_machine neo1973;
+static struct snd_soc_card neo1973;
 static struct i2c_client *i2c;
 
 static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
@@ -548,7 +548,6 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai bt_dai = {
 	.name = "Bluetooth",
 	.id = 0,
-	.type = SND_SOC_DAI_PCM,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 1,
@@ -579,8 +578,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 },
 };
 
-static struct snd_soc_machine neo1973 = {
+static struct snd_soc_card neo1973 = {
 	.name = "neo1973",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = neo1973_dai,
 	.num_links = ARRAY_SIZE(neo1973_dai),
 };
@@ -591,8 +591,7 @@ static struct wm8753_setup_data neo1973_wm8753_setup = {
 };
 
 static struct snd_soc_device neo1973_snd_devdata = {
-	.machine = &neo1973,
-	.platform = &s3c24xx_soc_platform,
+	.card = &neo1973,
 	.codec_dev = &soc_codec_dev_wm8753,
 	.codec_data = &neo1973_wm8753_setup,
 };
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ded7d995a922..f3fc0aba0aaf 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -343,7 +343,8 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
@@ -373,7 +374,8 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
@@ -647,8 +649,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct platform_device *dev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 	u32 iismod;
@@ -663,25 +664,24 @@ static int s3c2412_i2s_suspend(struct platform_device *dev,
 		iismod = readl(i2s->regs + S3C2412_IISMOD);
 
 		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__);
+			pr_warning("%s: RXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__);
+			pr_warning("%s: TXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			dev_warn(&dev->dev, "%s: IIS active\n", __func__);
+			pr_warning("%s: IIS active\n", __func__);
 	}
 
 	return 0;
 }
 
-static int s3c2412_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 
-	dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n",
-		 dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
 
 	if (dai->active) {
 		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
@@ -711,7 +711,6 @@ static int s3c2412_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai s3c2412_i2s_dai = {
 	.name	= "s3c2412-i2s",
 	.id	= 0,
-	.type	= SND_SOC_DAI_I2S,
 	.probe	= s3c2412_i2s_probe,
 	.suspend = s3c2412_i2s_suspend,
 	.resume = s3c2412_i2s_resume,
@@ -730,8 +729,6 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 	.ops = {
 		.trigger	= s3c2412_i2s_trigger,
 		.hw_params	= s3c2412_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= s3c2412_i2s_set_fmt,
 		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
 		.set_sysclk	= s3c2412_i2s_set_sysclk,
@@ -739,6 +736,19 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
+static int __init s3c2412_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c2412_i2s_dai);
+}
+module_init(s3c2412_i2s_init);
+
+static void __exit s3c2412_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c2412_i2s_dai);
+}
+module_exit(s3c2412_i2s_exit);
+
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 19c5c3cf5d8c..1bfce40bb2e4 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -271,7 +271,8 @@ static void s3c2443_ac97_remove(struct platform_device *pdev,
 }
 
 static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -284,7 +285,8 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
 
@@ -313,7 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				      struct snd_pcm_hw_params *params,
+				      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -327,7 +330,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 }
 
 static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
-	int cmd)
+				    int cmd, struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
 
@@ -356,7 +359,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = s3c2443_ac97_probe,
 	.remove = s3c2443_ac97_remove,
 	.playback = {
@@ -378,7 +381,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
@@ -393,6 +396,21 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __init s3c2443_ac97_init(void)
+{
+	return snd_soc_register_dais(s3c2443_ac97_dai,
+				     ARRAY_SIZE(s3c2443_ac97_dai));
+}
+module_init(s3c2443_ac97_init);
+
+static void __exit s3c2443_ac97_exit(void)
+{
+	snd_soc_unregister_dais(s3c2443_ac97_dai,
+				ARRAY_SIZE(s3c2443_ac97_dai));
+}
+module_exit(s3c2443_ac97_exit);
+
+
 MODULE_AUTHOR("Graeme Gregory");
 MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index ba4476b55fbc..6f4d439b57aa 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -243,7 +243,8 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
@@ -261,10 +262,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
+		iismod &= ~S3C2410_IISMOD_16BIT;
+		((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->dma_size = 1;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		iismod |= S3C2410_IISMOD_16BIT;
+		((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->dma_size = 2;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -272,7 +280,8 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -410,8 +419,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c24xx_i2s_suspend(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -425,8 +433,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int s3c24xx_i2s_resume(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
@@ -452,7 +459,6 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = s3c24xx_i2s_probe,
 	.suspend = s3c24xx_i2s_suspend,
 	.resume = s3c24xx_i2s_resume,
@@ -468,8 +474,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
 		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = s3c24xx_i2s_hw_params,
 		.set_fmt = s3c24xx_i2s_set_fmt,
 		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
 		.set_sysclk = s3c24xx_i2s_set_sysclk,
@@ -477,6 +482,18 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
+static int __init s3c24xx_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c24xx_i2s_dai);
+}
+module_init(s3c24xx_i2s_init);
+
+static void __exit s3c24xx_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c24xx_i2s_dai);
+}
+module_exit(s3c24xx_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index e13e614bada9..7c64d31d067e 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -465,6 +465,18 @@ struct snd_soc_platform s3c24xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
+static int __init s3c24xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&s3c24xx_soc_platform);
+}
+module_init(s3c24xx_soc_platform_init);
+
+static void __exit s3c24xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&s3c24xx_soc_platform);
+}
+module_exit(s3c24xx_soc_platform_exit);
+
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
new file mode 100644
index 000000000000..a0a4d1832a14
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -0,0 +1,373 @@
+/*
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * s3c24xx_uda134x.c  --  S3C24XX_UDA134X ALSA SoC Audio board driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/s3c24xx_uda134x.h>
+#include <sound/uda134x.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda134x.h"
+
+
+/* #define ENFORCE_RATES 1 */
+/*
+  Unfortunately the S3C24XX in master mode has a limited capacity of
+  generating the clock for the codec. If you define this only rates
+  that are really available will be enforced. But be careful, most
+  user level application just want the usual sampling frequencies (8,
+  11.025, 22.050, 44.1 kHz) and anyway resampling is a costly
+  operation for embedded systems. So if you aren't very lucky or your
+  hardware engineer wasn't very forward-looking it's better to leave
+  this undefined. If you do so an approximate value for the requested
+  sampling rate in the range -/+ 5% will be chosen. If this in not
+  possible an error will be returned.
+*/
+
+static struct clk *xtal;
+static struct clk *pclk;
+/* this is need because we don't have a place where to keep the
+ * pointers to the clocks in each substream. We get the clocks only
+ * when we are actually using them so we don't block stuff like
+ * frequency change or oscillator power-off */
+static int clk_users;
+static DEFINE_MUTEX(clk_lock);
+
+static unsigned int rates[33 * 2];
+#ifdef ENFORCE_RATES
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+	.count	= ARRAY_SIZE(rates),
+	.list	= rates,
+	.mask	= 0,
+};
+#endif
+
+static struct platform_device *s3c24xx_uda134x_snd_device;
+
+static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+#ifdef ENFORCE_RATES
+	struct snd_pcm_runtime *runtime = substream->runtime;;
+#endif
+
+	mutex_lock(&clk_lock);
+	pr_debug("%s %d\n", __func__, clk_users);
+	if (clk_users == 0) {
+		xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
+		if (!xtal) {
+			printk(KERN_ERR "%s cannot get xtal\n", __func__);
+			ret = -EBUSY;
+		} else {
+			pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
+				       "pclk");
+			if (!pclk) {
+				printk(KERN_ERR "%s cannot get pclk\n",
+				       __func__);
+				clk_put(xtal);
+				ret = -EBUSY;
+			}
+		}
+		if (!ret) {
+			int i, j;
+
+			for (i = 0; i < 2; i++) {
+				int fs = i ? 256 : 384;
+
+				rates[i*33] = clk_get_rate(xtal) / fs;
+				for (j = 1; j < 33; j++)
+					rates[i*33 + j] = clk_get_rate(pclk) /
+						(j * fs);
+			}
+		}
+	}
+	clk_users += 1;
+	mutex_unlock(&clk_lock);
+	if (!ret) {
+#ifdef ENFORCE_RATES
+		ret = snd_pcm_hw_constraint_list(runtime, 0,
+						 SNDRV_PCM_HW_PARAM_RATE,
+						 &hw_constraints_rates);
+		if (ret < 0)
+			printk(KERN_ERR "%s cannot set constraints\n",
+			       __func__);
+#endif
+	}
+	return ret;
+}
+
+static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+	mutex_lock(&clk_lock);
+	pr_debug("%s %d\n", __func__, clk_users);
+	clk_users -= 1;
+	if (clk_users == 0) {
+		clk_put(xtal);
+		xtal = NULL;
+		clk_put(pclk);
+		pclk = NULL;
+	}
+	mutex_unlock(&clk_lock);
+}
+
+static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret = 0;
+	int clk_source, fs_mode;
+	unsigned long rate = params_rate(params);
+	long err, cerr;
+	unsigned int div;
+	int i, bi;
+
+	err = 999999;
+	bi = 0;
+	for (i = 0; i < 2*33; i++) {
+		cerr = rates[i] - rate;
+		if (cerr < 0)
+			cerr = -cerr;
+		if (cerr < err) {
+			err = cerr;
+			bi = i;
+		}
+	}
+	if (bi / 33 == 1)
+		fs_mode = S3C2410_IISMOD_256FS;
+	else
+		fs_mode = S3C2410_IISMOD_384FS;
+	if (bi % 33 == 0) {
+		clk_source = S3C24XX_CLKSRC_MPLL;
+		div = 1;
+	} else {
+		clk_source = S3C24XX_CLKSRC_PCLK;
+		div = bi % 33;
+	}
+	pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi);
+
+	clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
+	pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__,
+		 fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
+		 clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
+		 div, clk, err);
+
+	if ((err * 100 / rate) > 5) {
+		printk(KERN_ERR "S3C24XX_UDA134X: effective frequency "
+		       "too different from desired (%ld%%)\n",
+		       err * 100 / rate);
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk,
+			SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+			S3C2410_IISMOD_32FS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+			S3C24XX_PRESCALE(div, div));
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops s3c24xx_uda134x_ops = {
+	.startup = s3c24xx_uda134x_startup,
+	.shutdown = s3c24xx_uda134x_shutdown,
+	.hw_params = s3c24xx_uda134x_hw_params,
+};
+
+static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
+	.name = "UDA134X",
+	.stream_name = "UDA134X",
+	.codec_dai = &uda134x_dai,
+	.cpu_dai = &s3c24xx_i2s_dai,
+	.ops = &s3c24xx_uda134x_ops,
+};
+
+static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
+	.name = "S3C24XX_UDA134X",
+	.platform = &s3c24xx_soc_platform,
+	.dai_link = &s3c24xx_uda134x_dai_link,
+	.num_links = 1,
+};
+
+static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins;
+
+static void setdat(int v)
+{
+	gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0);
+}
+
+static void setclk(int v)
+{
+	gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0);
+}
+
+static void setmode(int v)
+{
+	gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0);
+}
+
+static struct uda134x_platform_data s3c24xx_uda134x = {
+	.l3 = {
+		.setdat = setdat,
+		.setclk = setclk,
+		.setmode = setmode,
+		.data_hold = 1,
+		.data_setup = 1,
+		.clock_high = 1,
+		.mode_hold = 1,
+		.mode = 1,
+		.mode_setup = 1,
+	},
+};
+
+static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
+	.card = &snd_soc_s3c24xx_uda134x,
+	.codec_dev = &soc_codec_dev_uda134x,
+	.codec_data = &s3c24xx_uda134x,
+};
+
+static int s3c24xx_uda134x_setup_pin(int pin, char *fun)
+{
+	if (gpio_request(pin, "s3c24xx_uda134x") < 0) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+		       "l3 %s pin already in use", fun);
+		return -EBUSY;
+	}
+	gpio_direction_output(pin, 0);
+	return 0;
+}
+
+static int s3c24xx_uda134x_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n");
+
+	s3c24xx_uda134x_l3_pins = pdev->dev.platform_data;
+	if (s3c24xx_uda134x_l3_pins == NULL) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+		       "unable to find platform data\n");
+		return -ENODEV;
+	}
+	s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power;
+	s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model;
+
+	if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data,
+				      "data") < 0)
+		return -EBUSY;
+	if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk,
+				      "clk") < 0) {
+		gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+		return -EBUSY;
+	}
+	if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode,
+				      "mode") < 0) {
+		gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+		gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+		return -EBUSY;
+	}
+
+	s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!s3c24xx_uda134x_snd_device) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+		       "Unable to register\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(s3c24xx_uda134x_snd_device,
+			     &s3c24xx_uda134x_snd_devdata);
+	s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev;
+	ret = platform_device_add(s3c24xx_uda134x_snd_device);
+	if (ret) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
+		platform_device_put(s3c24xx_uda134x_snd_device);
+	}
+
+	return ret;
+}
+
+static int s3c24xx_uda134x_remove(struct platform_device *pdev)
+{
+	platform_device_unregister(s3c24xx_uda134x_snd_device);
+	gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+	gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+	gpio_free(s3c24xx_uda134x_l3_pins->l3_mode);
+	return 0;
+}
+
+static struct platform_driver s3c24xx_uda134x_driver = {
+	.probe  = s3c24xx_uda134x_probe,
+	.remove = s3c24xx_uda134x_remove,
+	.driver = {
+		.name = "s3c24xx_uda134x",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init s3c24xx_uda134x_init(void)
+{
+	return platform_driver_register(&s3c24xx_uda134x_driver);
+}
+
+static void __exit s3c24xx_uda134x_exit(void)
+{
+	platform_driver_unregister(&s3c24xx_uda134x_driver);
+}
+
+
+module_init(s3c24xx_uda134x_init);
+module_exit(s3c24xx_uda134x_exit);
+
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index 8515d6ff03f2..a2a4f5323c17 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -23,7 +23,7 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
 
-static struct snd_soc_machine smdk2443;
+static struct snd_soc_card smdk2443;
 
 static struct snd_soc_dai_link smdk2443_dai[] = {
 {
@@ -34,15 +34,15 @@ static struct snd_soc_dai_link smdk2443_dai[] = {
 },
 };
 
-static struct snd_soc_machine smdk2443 = {
+static struct snd_soc_card smdk2443 = {
 	.name = "SMDK2443",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = smdk2443_dai,
 	.num_links = ARRAY_SIZE(smdk2443_dai),
 };
 
 static struct snd_soc_device smdk2443_snd_ac97_devdata = {
-	.machine = &smdk2443,
-	.platform = &s3c24xx_soc_platform,
+	.card = &smdk2443,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 9faa12622d09..0dad3a0bb920 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -348,6 +348,18 @@ struct snd_soc_platform sh7760_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(sh7760_soc_platform);
 
+static int __init sh7760_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&sh7760_soc_platform);
+}
+module_init(sh7760_soc_platform_init);
+
+static void __exit sh7760_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&sh7760_soc_platform);
+}
+module_exit(sh7760_soc_platform_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index df7bc345c320..eab31838badf 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -236,7 +236,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int hac_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
+			 struct snd_pcm_hw_params *params,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
@@ -270,7 +271,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 {
 	.name			= "HAC0",
 	.id			= 0,
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.playback = {
 		.rates		= AC97_RATES,
 		.formats	= AC97_FMTS,
@@ -290,8 +291,8 @@ struct snd_soc_dai sh4_hac_dai[] = {
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
 	.name			= "HAC1",
+	.ac97_control		= 1,
 	.id			= 1,
-	.type			= SND_SOC_DAI_AC97,
 	.playback = {
 		.rates		= AC97_RATES,
 		.formats	= AC97_FMTS,
@@ -313,6 +314,18 @@ struct snd_soc_dai sh4_hac_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_hac_dai);
 
+static int __init sh4_hac_init(void)
+{
+	return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_init(sh4_hac_init);
+
+static void __exit sh4_hac_exit(void)
+{
+	snd_soc_unregister_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_exit(sh4_hac_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 92bfaf4774a7..ce7f95b59de3 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -38,15 +38,15 @@ static struct snd_soc_dai_link sh7760_ac97_dai = {
 	.ops = NULL,
 };
 
-static struct snd_soc_machine sh7760_ac97_soc_machine  = {
+static struct snd_soc_card sh7760_ac97_soc_machine  = {
 	.name = "SH7760 AC97",
+	.platform = &sh7760_soc_platform,
 	.dai_link = &sh7760_ac97_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device sh7760_ac97_snd_devdata = {
-	.machine = &sh7760_ac97_soc_machine,
-	.platform = &sh7760_soc_platform,
+	.card = &sh7760_ac97_soc_machine,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 55c3464163ab..d1e5390fddeb 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -89,7 +89,8 @@ struct ssi_priv {
  * track usage of the SSI; it is simplex-only so prevent attempts of
  * concurrent playback + capture. FIXME: any locking required?
  */
-static int ssi_startup(struct snd_pcm_substream *substream)
+static int ssi_startup(struct snd_pcm_substream *substream,
+		       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -101,7 +102,8 @@ static int ssi_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void ssi_shutdown(struct snd_pcm_substream *substream)
+static void ssi_shutdown(struct snd_pcm_substream *substream,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -109,7 +111,8 @@ static void ssi_shutdown(struct snd_pcm_substream *substream)
 	ssi->inuse = 0;
 }
 
-static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+		       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -129,7 +132,8 @@ static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int ssi_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
+			 struct snd_pcm_hw_params *params,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -336,7 +340,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
 	.id			= 0,
-	.type			= SND_SOC_DAI_I2S,
 	.playback = {
 		.rates		= SSI_RATES,
 		.formats	= SSI_FMTS,
@@ -354,8 +357,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.shutdown	= ssi_shutdown,
 		.trigger	= ssi_trigger,
 		.hw_params	= ssi_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk	= ssi_set_sysclk,
 		.set_clkdiv	= ssi_set_clkdiv,
 		.set_fmt	= ssi_set_fmt,
@@ -365,7 +366,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI1",
 	.id			= 1,
-	.type			= SND_SOC_DAI_I2S,
 	.playback = {
 		.rates		= SSI_RATES,
 		.formats	= SSI_FMTS,
@@ -383,8 +383,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.shutdown	= ssi_shutdown,
 		.trigger	= ssi_trigger,
 		.hw_params	= ssi_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk	= ssi_set_sysclk,
 		.set_clkdiv	= ssi_set_clkdiv,
 		.set_fmt	= ssi_set_fmt,
@@ -394,6 +392,18 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_ssi_dai);
 
+static int __init sh4_ssi_init(void)
+{
+	return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_init(sh4_ssi_init);
+
+static void __exit sh4_ssi_exit(void)
+{
+	snd_soc_unregister_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_exit(sh4_ssi_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 16c7453f4946..b098c0b4c584 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -26,6 +26,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/bitops.h>
+#include <linux/debugfs.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -34,18 +35,23 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 
-/* debug */
-#define SOC_DEBUG 0
-#if SOC_DEBUG
-#define dbg(format, arg...) printk(format, ## arg)
-#else
-#define dbg(format, arg...)
-#endif
-
 static DEFINE_MUTEX(pcm_mutex);
 static DEFINE_MUTEX(io_mutex);
 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
 
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_root;
+#endif
+
+static DEFINE_MUTEX(client_mutex);
+static LIST_HEAD(card_list);
+static LIST_HEAD(dai_list);
+static LIST_HEAD(platform_list);
+static LIST_HEAD(codec_list);
+
+static int snd_soc_register_card(struct snd_soc_card *card);
+static int snd_soc_unregister_card(struct snd_soc_card *card);
+
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
  * It can be used to eliminate pops between different playback streams, e.g.
@@ -107,20 +113,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
-static inline const char *get_dai_name(int type)
-{
-	switch (type) {
-	case SND_SOC_DAI_AC97_BUS:
-	case SND_SOC_DAI_AC97:
-		return "AC97";
-	case SND_SOC_DAI_I2S:
-		return "I2S";
-	case SND_SOC_DAI_PCM:
-		return "PCM";
-	}
-	return NULL;
-}
-
 /*
  * Called by ALSA when a PCM substream is opened, the runtime->hw record is
  * then initialized and any private data can be allocated. This also calls
@@ -130,9 +122,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
@@ -141,7 +134,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 
 	/* startup the audio subsystem */
 	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream);
+		ret = cpu_dai->ops.startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -158,7 +151,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	}
 
 	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream);
+		ret = codec_dai->ops.startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -228,12 +221,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		goto machine_err;
 	}
 
-	dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
-	dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
-	dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
-		runtime->hw.channels_max);
-	dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
-		runtime->hw.rate_max);
+	pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
+	pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
+	pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+		 runtime->hw.channels_max);
+	pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+		 runtime->hw.rate_max);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		cpu_dai->playback.active = codec_dai->playback.active = 1;
@@ -255,7 +248,7 @@ codec_dai_err:
 
 platform_err:
 	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream);
+		cpu_dai->ops.shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -268,8 +261,9 @@ out:
  */
 static void close_delayed_work(struct work_struct *work)
 {
-	struct snd_soc_device *socdev =
-		container_of(work, struct snd_soc_device, delayed_work.work);
+	struct snd_soc_card *card = container_of(work, struct snd_soc_card,
+						 delayed_work.work);
+	struct snd_soc_device *socdev = card->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_dai *codec_dai;
 	int i;
@@ -278,18 +272,18 @@ static void close_delayed_work(struct work_struct *work)
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
 
-		dbg("pop wq checking: %s status: %s waiting: %s\n",
-			codec_dai->playback.stream_name,
-			codec_dai->playback.active ? "active" : "inactive",
-			codec_dai->pop_wait ? "yes" : "no");
+		pr_debug("pop wq checking: %s status: %s waiting: %s\n",
+			 codec_dai->playback.stream_name,
+			 codec_dai->playback.active ? "active" : "inactive",
+			 codec_dai->pop_wait ? "yes" : "no");
 
 		/* are we waiting on this codec DAI stream */
 		if (codec_dai->pop_wait == 1) {
 
 			/* Reduce power if no longer active */
 			if (codec->active == 0) {
-				dbg("pop wq D1 %s %s\n", codec->name,
-					codec_dai->playback.stream_name);
+				pr_debug("pop wq D1 %s %s\n", codec->name,
+					 codec_dai->playback.stream_name);
 				snd_soc_dapm_set_bias_level(socdev,
 					SND_SOC_BIAS_PREPARE);
 			}
@@ -301,8 +295,8 @@ static void close_delayed_work(struct work_struct *work)
 
 			/* Fall into standby if no longer active */
 			if (codec->active == 0) {
-				dbg("pop wq D3 %s %s\n", codec->name,
-					codec_dai->playback.stream_name);
+				pr_debug("pop wq D3 %s %s\n", codec->name,
+					 codec_dai->playback.stream_name);
 				snd_soc_dapm_set_bias_level(socdev,
 					SND_SOC_BIAS_STANDBY);
 			}
@@ -320,8 +314,9 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -346,10 +341,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
 	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream);
+		cpu_dai->ops.shutdown(substream, cpu_dai);
 
 	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream);
+		codec_dai->ops.shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -361,7 +356,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* start delayed pop wq here for playback streams */
 		codec_dai->pop_wait = 1;
-		schedule_delayed_work(&socdev->delayed_work,
+		schedule_delayed_work(&card->delayed_work,
 			msecs_to_jiffies(pmdown_time));
 	} else {
 		/* capture streams can be powered down now */
@@ -387,8 +382,9 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -413,7 +409,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream);
+		ret = codec_dai->ops.prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
@@ -421,58 +417,49 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream);
+		ret = cpu_dai->ops.prepare(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
 			goto out;
 		}
 	}
 
-	/* we only want to start a DAPM playback stream if we are not waiting
-	 * on an existing one stopping */
-	if (codec_dai->pop_wait) {
-		/* we are waiting for the delayed work to start */
-		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-				snd_soc_dapm_stream_event(socdev->codec,
+	/* cancel any delayed stream shutdown that is pending */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+	    codec_dai->pop_wait) {
+		codec_dai->pop_wait = 0;
+		cancel_delayed_work(&card->delayed_work);
+	}
+
+	/* do we need to power up codec */
+	if (codec->bias_level != SND_SOC_BIAS_ON) {
+		snd_soc_dapm_set_bias_level(socdev,
+					    SND_SOC_BIAS_PREPARE);
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			snd_soc_dapm_stream_event(codec,
+					codec_dai->playback.stream_name,
+					SND_SOC_DAPM_STREAM_START);
+		else
+			snd_soc_dapm_stream_event(codec,
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
-		else {
-			codec_dai->pop_wait = 0;
-			cancel_delayed_work(&socdev->delayed_work);
-			snd_soc_dai_digital_mute(codec_dai, 0);
-		}
+
+		snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
+		snd_soc_dai_digital_mute(codec_dai, 0);
+
 	} else {
-		/* no delayed work - do we need to power up codec */
-		if (codec->bias_level != SND_SOC_BIAS_ON) {
-
-			snd_soc_dapm_set_bias_level(socdev,
-						    SND_SOC_BIAS_PREPARE);
-
-			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-				snd_soc_dapm_stream_event(codec,
+		/* codec already powered - power on widgets */
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			snd_soc_dapm_stream_event(codec,
 					codec_dai->playback.stream_name,
 					SND_SOC_DAPM_STREAM_START);
-			else
-				snd_soc_dapm_stream_event(codec,
+		else
+			snd_soc_dapm_stream_event(codec,
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
 
-			snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
-			snd_soc_dai_digital_mute(codec_dai, 0);
-
-		} else {
-			/* codec already powered - power on widgets */
-			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-				snd_soc_dapm_stream_event(codec,
-					codec_dai->playback.stream_name,
-					SND_SOC_DAPM_STREAM_START);
-			else
-				snd_soc_dapm_stream_event(codec,
-					codec_dai->capture.stream_name,
-					SND_SOC_DAPM_STREAM_START);
-
-			snd_soc_dai_digital_mute(codec_dai, 0);
-		}
+		snd_soc_dai_digital_mute(codec_dai, 0);
 	}
 
 out:
@@ -491,7 +478,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
@@ -507,7 +495,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params);
+		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -516,7 +504,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params);
+		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -539,11 +527,11 @@ out:
 
 platform_err:
 	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream);
+		cpu_dai->ops.hw_free(substream, cpu_dai);
 
 interface_err:
 	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream);
+		codec_dai->ops.hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -561,7 +549,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -582,10 +571,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	/* now free hw params for the DAI's  */
 	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream);
+		codec_dai->ops.hw_free(substream, codec_dai);
 
 	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream);
+		cpu_dai->ops.hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -595,14 +584,15 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card= socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
 	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd);
+		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -614,7 +604,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	}
 
 	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd);
+		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -636,8 +626,8 @@ static struct snd_pcm_ops soc_pcm_ops = {
 static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	int i;
@@ -653,29 +643,29 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
 
 	/* mute any active DAC's */
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
-		if (dai->dai_ops.digital_mute && dai->playback.active)
-			dai->dai_ops.digital_mute(dai, 1);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+		if (dai->ops.digital_mute && dai->playback.active)
+			dai->ops.digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
-	for (i = 0; i < machine->num_links; i++)
-		snd_pcm_suspend_all(machine->dai_link[i].pcm);
+	for (i = 0; i < card->num_links; i++)
+		snd_pcm_suspend_all(card->dai_link[i].pcm);
 
-	if (machine->suspend_pre)
-		machine->suspend_pre(pdev, state);
+	if (card->suspend_pre)
+		card->suspend_pre(pdev, state);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
-		if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
-			cpu_dai->suspend(pdev, cpu_dai);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->suspend && !cpu_dai->ac97_control)
+			cpu_dai->suspend(cpu_dai);
 		if (platform->suspend)
-			platform->suspend(pdev, cpu_dai);
+			platform->suspend(cpu_dai);
 	}
 
 	/* close any waiting streams and save state */
-	run_delayed_work(&socdev->delayed_work);
+	run_delayed_work(&card->delayed_work);
 	codec->suspend_bias_level = codec->bias_level;
 
 	for (i = 0; i < codec->num_dai; i++) {
@@ -692,14 +682,14 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	if (codec_dev->suspend)
 		codec_dev->suspend(pdev, state);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
-		if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
-			cpu_dai->suspend(pdev, cpu_dai);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->suspend && cpu_dai->ac97_control)
+			cpu_dai->suspend(cpu_dai);
 	}
 
-	if (machine->suspend_post)
-		machine->suspend_post(pdev, state);
+	if (card->suspend_post)
+		card->suspend_post(pdev, state);
 
 	return 0;
 }
@@ -709,11 +699,11 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
  */
 static void soc_resume_deferred(struct work_struct *work)
 {
-	struct snd_soc_device *socdev = container_of(work,
-						     struct snd_soc_device,
-						     deferred_resume_work);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = container_of(work,
+						 struct snd_soc_card,
+						 deferred_resume_work);
+	struct snd_soc_device *socdev = card->socdev;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct platform_device *pdev = to_platform_device(socdev->dev);
@@ -723,15 +713,15 @@ static void soc_resume_deferred(struct work_struct *work)
 	 * so userspace apps are blocked from touching us
 	 */
 
-	dev_info(socdev->dev, "starting resume work\n");
+	dev_dbg(socdev->dev, "starting resume work\n");
 
-	if (machine->resume_pre)
-		machine->resume_pre(pdev);
+	if (card->resume_pre)
+		card->resume_pre(pdev);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
-		if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
-			cpu_dai->resume(pdev, cpu_dai);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->resume && cpu_dai->ac97_control)
+			cpu_dai->resume(cpu_dai);
 	}
 
 	if (codec_dev->resume)
@@ -749,24 +739,24 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	/* unmute any active DACs */
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
-		if (dai->dai_ops.digital_mute && dai->playback.active)
-			dai->dai_ops.digital_mute(dai, 0);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+		if (dai->ops.digital_mute && dai->playback.active)
+			dai->ops.digital_mute(dai, 0);
 	}
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
-		if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
-			cpu_dai->resume(pdev, cpu_dai);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->resume && !cpu_dai->ac97_control)
+			cpu_dai->resume(cpu_dai);
 		if (platform->resume)
-			platform->resume(pdev, cpu_dai);
+			platform->resume(cpu_dai);
 	}
 
-	if (machine->resume_post)
-		machine->resume_post(pdev);
+	if (card->resume_post)
+		card->resume_post(pdev);
 
-	dev_info(socdev->dev, "resume work completed\n");
+	dev_dbg(socdev->dev, "resume work completed\n");
 
 	/* userspace can access us now we are back as we were before */
 	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
@@ -776,11 +766,12 @@ static void soc_resume_deferred(struct work_struct *work)
 static int soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
 
-	dev_info(socdev->dev, "scheduling resume work\n");
+	dev_dbg(socdev->dev, "scheduling resume work\n");
 
-	if (!schedule_work(&socdev->deferred_resume_work))
-		dev_err(socdev->dev, "work item may be lost\n");
+	if (!schedule_work(&card->deferred_resume_work))
+		dev_err(socdev->dev, "resume work item may be lost\n");
 
 	return 0;
 }
@@ -790,23 +781,83 @@ static int soc_resume(struct platform_device *pdev)
 #define soc_resume	NULL
 #endif
 
-/* probes a new socdev */
-static int soc_probe(struct platform_device *pdev)
+static void snd_soc_instantiate_card(struct snd_soc_card *card)
 {
-	int ret = 0, i;
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+	struct platform_device *pdev = container_of(card->dev,
+						    struct platform_device,
+						    dev);
+	struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
+	struct snd_soc_platform *platform;
+	struct snd_soc_dai *dai;
+	int i, found, ret, ac97;
 
-	if (machine->probe) {
-		ret = machine->probe(pdev);
-		if (ret < 0)
-			return ret;
+	if (card->instantiated)
+		return;
+
+	found = 0;
+	list_for_each_entry(platform, &platform_list, list)
+		if (card->platform == platform) {
+			found = 1;
+			break;
+		}
+	if (!found) {
+		dev_dbg(card->dev, "Platform %s not registered\n",
+			card->platform->name);
+		return;
 	}
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	ac97 = 0;
+	for (i = 0; i < card->num_links; i++) {
+		found = 0;
+		list_for_each_entry(dai, &dai_list, list)
+			if (card->dai_link[i].cpu_dai == dai) {
+				found = 1;
+				break;
+			}
+		if (!found) {
+			dev_dbg(card->dev, "DAI %s not registered\n",
+				card->dai_link[i].cpu_dai->name);
+			return;
+		}
+
+		if (card->dai_link[i].cpu_dai->ac97_control)
+			ac97 = 1;
+	}
+
+	/* If we have AC97 in the system then don't wait for the
+	 * codec.  This will need revisiting if we have to handle
+	 * systems with mixed AC97 and non-AC97 parts.  Only check for
+	 * DAIs currently; we can't do this per link since some AC97
+	 * codecs have non-AC97 DAIs.
+	 */
+	if (!ac97)
+		for (i = 0; i < card->num_links; i++) {
+			found = 0;
+			list_for_each_entry(dai, &dai_list, list)
+				if (card->dai_link[i].codec_dai == dai) {
+					found = 1;
+					break;
+				}
+			if (!found) {
+				dev_dbg(card->dev, "DAI %s not registered\n",
+					card->dai_link[i].codec_dai->name);
+				return;
+			}
+		}
+
+	/* Note that we do not current check for codec components */
+
+	dev_dbg(card->dev, "All components present, instantiating\n");
+
+	/* Found everything, bring it up */
+	if (card->probe) {
+		ret = card->probe(pdev);
+		if (ret < 0)
+			return;
+	}
+
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->probe) {
 			ret = cpu_dai->probe(pdev, cpu_dai);
 			if (ret < 0)
@@ -827,13 +878,15 @@ static int soc_probe(struct platform_device *pdev)
 	}
 
 	/* DAPM stream work */
-	INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
 #ifdef CONFIG_PM
 	/* deferred resume work */
-	INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
+	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
 
-	return 0;
+	card->instantiated = 1;
+
+	return;
 
 platform_err:
 	if (codec_dev->remove)
@@ -841,15 +894,45 @@ platform_err:
 
 cpu_dai_err:
 	for (i--; i >= 0; i--) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
 			cpu_dai->remove(pdev, cpu_dai);
 	}
 
-	if (machine->remove)
-		machine->remove(pdev);
+	if (card->remove)
+		card->remove(pdev);
+}
 
-	return ret;
+/*
+ * Attempt to initialise any uninitalised cards.  Must be called with
+ * client_mutex.
+ */
+static void snd_soc_instantiate_cards(void)
+{
+	struct snd_soc_card *card;
+	list_for_each_entry(card, &card_list, list)
+		snd_soc_instantiate_card(card);
+}
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+	int ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
+
+	/* Bodge while we push things out of socdev */
+	card->socdev = socdev;
+
+	/* Bodge while we unpick instantiation */
+	card->dev = &pdev->dev;
+	ret = snd_soc_register_card(card);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to register card\n");
+		return ret;
+	}
+
+	return 0;
 }
 
 /* removes a socdev */
@@ -857,11 +940,11 @@ static int soc_remove(struct platform_device *pdev)
 {
 	int i;
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
-	run_delayed_work(&socdev->delayed_work);
+	run_delayed_work(&card->delayed_work);
 
 	if (platform->remove)
 		platform->remove(pdev);
@@ -869,14 +952,16 @@ static int soc_remove(struct platform_device *pdev)
 	if (codec_dev->remove)
 		codec_dev->remove(pdev);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
 			cpu_dai->remove(pdev, cpu_dai);
 	}
 
-	if (machine->remove)
-		machine->remove(pdev);
+	if (card->remove)
+		card->remove(pdev);
+
+	snd_soc_unregister_card(card);
 
 	return 0;
 }
@@ -898,6 +983,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	struct snd_soc_dai_link *dai_link, int num)
 {
 	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
 	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
 	struct snd_soc_pcm_runtime *rtd;
@@ -914,8 +1001,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	codec_dai->codec = socdev->codec;
 
 	/* check client and interface hw capabilities */
-	sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
-		get_dai_name(cpu_dai->type), num);
+	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
+		num);
 
 	if (codec_dai->playback.channels_min)
 		playback = 1;
@@ -933,13 +1020,13 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 
 	dai_link->pcm = pcm;
 	pcm->private_data = rtd;
-	soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
-	soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
-	soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
-	soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
-	soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
-	soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
-	soc_pcm_ops.page = socdev->platform->pcm_ops->page;
+	soc_pcm_ops.mmap = platform->pcm_ops->mmap;
+	soc_pcm_ops.pointer = platform->pcm_ops->pointer;
+	soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
+	soc_pcm_ops.copy = platform->pcm_ops->copy;
+	soc_pcm_ops.silence = platform->pcm_ops->silence;
+	soc_pcm_ops.ack = platform->pcm_ops->ack;
+	soc_pcm_ops.page = platform->pcm_ops->page;
 
 	if (playback)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
@@ -947,24 +1034,22 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	if (capture)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
 
-	ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+	ret = platform->pcm_new(codec->card, codec_dai, pcm);
 	if (ret < 0) {
 		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
 		kfree(rtd);
 		return ret;
 	}
 
-	pcm->private_free = socdev->platform->pcm_free;
+	pcm->private_free = platform->pcm_free;
 	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
 		cpu_dai->name);
 	return ret;
 }
 
 /* codec register dump */
-static ssize_t codec_reg_show(struct device *dev,
-	struct device_attribute *attr, char *buf)
+static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
 {
-	struct snd_soc_device *devdata = dev_get_drvdata(dev);
 	struct snd_soc_codec *codec = devdata->codec;
 	int i, step = 1, count = 0;
 
@@ -1001,8 +1086,110 @@ static ssize_t codec_reg_show(struct device *dev,
 
 	return count;
 }
+static ssize_t codec_reg_show(struct device *dev,
+	struct device_attribute *attr, char *buf)
+{
+	struct snd_soc_device *devdata = dev_get_drvdata(dev);
+	return soc_codec_reg_show(devdata, buf);
+}
+
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
 
+#ifdef CONFIG_DEBUG_FS
+static int codec_reg_open_file(struct inode *inode, struct file *file)
+{
+	file->private_data = inode->i_private;
+	return 0;
+}
+
+static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
+			       size_t count, loff_t *ppos)
+{
+	ssize_t ret;
+	struct snd_soc_codec *codec = file->private_data;
+	struct device *card_dev = codec->card->dev;
+	struct snd_soc_device *devdata = card_dev->driver_data;
+	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+	ret = soc_codec_reg_show(devdata, buf);
+	if (ret >= 0)
+		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+	kfree(buf);
+	return ret;
+}
+
+static ssize_t codec_reg_write_file(struct file *file,
+		const char __user *user_buf, size_t count, loff_t *ppos)
+{
+	char buf[32];
+	int buf_size;
+	char *start = buf;
+	unsigned long reg, value;
+	int step = 1;
+	struct snd_soc_codec *codec = file->private_data;
+
+	buf_size = min(count, (sizeof(buf)-1));
+	if (copy_from_user(buf, user_buf, buf_size))
+		return -EFAULT;
+	buf[buf_size] = 0;
+
+	if (codec->reg_cache_step)
+		step = codec->reg_cache_step;
+
+	while (*start == ' ')
+		start++;
+	reg = simple_strtoul(start, &start, 16);
+	if ((reg >= codec->reg_cache_size) || (reg % step))
+		return -EINVAL;
+	while (*start == ' ')
+		start++;
+	if (strict_strtoul(start, 16, &value))
+		return -EINVAL;
+	codec->write(codec, reg, value);
+	return buf_size;
+}
+
+static const struct file_operations codec_reg_fops = {
+	.open = codec_reg_open_file,
+	.read = codec_reg_read_file,
+	.write = codec_reg_write_file,
+};
+
+static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
+						 debugfs_root, codec,
+						 &codec_reg_fops);
+	if (!codec->debugfs_reg)
+		printk(KERN_WARNING
+		       "ASoC: Failed to create codec register debugfs file\n");
+
+	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+						     debugfs_root,
+						     &codec->pop_time);
+	if (!codec->debugfs_pop_time)
+		printk(KERN_WARNING
+		       "Failed to create pop time debugfs file\n");
+}
+
+static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+	debugfs_remove(codec->debugfs_pop_time);
+	debugfs_remove(codec->debugfs_reg);
+}
+
+#else
+
+static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+
+static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+#endif
+
 /**
  * snd_soc_new_ac97_codec - initailise AC97 device
  * @codec: audio codec
@@ -1121,7 +1308,7 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0, i;
 
 	mutex_lock(&codec->mutex);
@@ -1140,11 +1327,11 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
 
 	/* create the pcms */
-	for (i = 0; i < machine->num_links; i++) {
-		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
+	for (i = 0; i < card->num_links; i++) {
+		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't create pcm %s\n",
-				machine->dai_link[i].stream_name);
+				card->dai_link[i].stream_name);
 			mutex_unlock(&codec->mutex);
 			return ret;
 		}
@@ -1156,7 +1343,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
 
 /**
- * snd_soc_register_card - register sound card
+ * snd_soc_init_card - register sound card
  * @socdev: the SoC audio device
  *
  * Register a SoC sound card. Also registers an AC97 device if the
@@ -1164,29 +1351,28 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
  *
  * Returns 0 for success, else error.
  */
-int snd_soc_register_card(struct snd_soc_device *socdev)
+int snd_soc_init_card(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0, i, ac97 = 0, err = 0;
 
-	for (i = 0; i < machine->num_links; i++) {
-		if (socdev->machine->dai_link[i].init) {
-			err = socdev->machine->dai_link[i].init(codec);
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].init) {
+			err = card->dai_link[i].init(codec);
 			if (err < 0) {
 				printk(KERN_ERR "asoc: failed to init %s\n",
-					socdev->machine->dai_link[i].stream_name);
+					card->dai_link[i].stream_name);
 				continue;
 			}
 		}
-		if (socdev->machine->dai_link[i].codec_dai->type ==
-			SND_SOC_DAI_AC97_BUS)
+		if (card->dai_link[i].codec_dai->ac97_control)
 			ac97 = 1;
 	}
 	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
-		 "%s", machine->name);
+		 "%s",  card->name);
 	snprintf(codec->card->longname, sizeof(codec->card->longname),
-		 "%s (%s)", machine->name, codec->name);
+		 "%s (%s)", card->name, codec->name);
 
 	ret = snd_card_register(codec->card);
 	if (ret < 0) {
@@ -1216,12 +1402,13 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 	if (err < 0)
 		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
+	soc_init_codec_debugfs(socdev->codec);
 	mutex_unlock(&codec->mutex);
 
 out:
 	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
+EXPORT_SYMBOL_GPL(snd_soc_init_card);
 
 /**
  * snd_soc_free_pcms - free sound card and pcms
@@ -1239,10 +1426,11 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
 #endif
 
 	mutex_lock(&codec->mutex);
+	soc_cleanup_codec_debugfs(socdev->codec);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
-		if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
+		if (codec_dai->ac97_control && codec->ac97) {
 			soc_ac97_dev_unregister(codec);
 			goto free_card;
 		}
@@ -1756,8 +1944,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -1776,8 +1964,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->dai_ops.set_clkdiv)
-		return dai->dai_ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops.set_clkdiv)
+		return dai->ops.set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -1795,8 +1983,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->dai_ops.set_pll)
-		return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops.set_pll)
+		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -1805,15 +1993,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
 /**
  * snd_soc_dai_set_fmt - configure DAI hardware audio format.
  * @dai: DAI
- * @clk_id: DAI specific clock ID
  * @fmt: SND_SOC_DAIFMT_ format value.
  *
  * Configures the DAI hardware format and clocking.
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->dai_ops.set_fmt)
-		return dai->dai_ops.set_fmt(dai, fmt);
+	if (dai->ops.set_fmt)
+		return dai->ops.set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -1831,8 +2018,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -1847,8 +2034,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_tristate(dai, tristate);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -1863,21 +2050,242 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->dai_ops.digital_mute)
-		return dai->dai_ops.digital_mute(dai, mute);
+	if (dai->ops.digital_mute)
+		return dai->ops.digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
-static int __devinit snd_soc_init(void)
+/**
+ * snd_soc_register_card - Register a card with the ASoC core
+ *
+ * @param card Card to register
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_register_card(struct snd_soc_card *card)
 {
-	printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
+	if (!card->name || !card->dev)
+		return -EINVAL;
+
+	INIT_LIST_HEAD(&card->list);
+	card->instantiated = 0;
+
+	mutex_lock(&client_mutex);
+	list_add(&card->list, &card_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(card->dev, "Registered card '%s'\n", card->name);
+
+	return 0;
+}
+
+/**
+ * snd_soc_unregister_card - Unregister a card with the ASoC core
+ *
+ * @param card Card to unregister
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_unregister_card(struct snd_soc_card *card)
+{
+	mutex_lock(&client_mutex);
+	list_del(&card->list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
+
+	return 0;
+}
+
+/**
+ * snd_soc_register_dai - Register a DAI with the ASoC core
+ *
+ * @param dai DAI to register
+ */
+int snd_soc_register_dai(struct snd_soc_dai *dai)
+{
+	if (!dai->name)
+		return -EINVAL;
+
+	/* The device should become mandatory over time */
+	if (!dai->dev)
+		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
+
+	INIT_LIST_HEAD(&dai->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&dai->list, &dai_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered DAI '%s'\n", dai->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dai);
+
+/**
+ * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
+ *
+ * @param dai DAI to unregister
+ */
+void snd_soc_unregister_dai(struct snd_soc_dai *dai)
+{
+	mutex_lock(&client_mutex);
+	list_del(&dai->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered DAI '%s'\n", dai->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
+
+/**
+ * snd_soc_register_dais - Register multiple DAIs with the ASoC core
+ *
+ * @param dai Array of DAIs to register
+ * @param count Number of DAIs
+ */
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
+{
+	int i, ret;
+
+	for (i = 0; i < count; i++) {
+		ret = snd_soc_register_dai(&dai[i]);
+		if (ret != 0)
+			goto err;
+	}
+
+	return 0;
+
+err:
+	for (i--; i >= 0; i--)
+		snd_soc_unregister_dai(&dai[i]);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dais);
+
+/**
+ * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
+ *
+ * @param dai Array of DAIs to unregister
+ * @param count Number of DAIs
+ */
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
+{
+	int i;
+
+	for (i = 0; i < count; i++)
+		snd_soc_unregister_dai(&dai[i]);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
+
+/**
+ * snd_soc_register_platform - Register a platform with the ASoC core
+ *
+ * @param platform platform to register
+ */
+int snd_soc_register_platform(struct snd_soc_platform *platform)
+{
+	if (!platform->name)
+		return -EINVAL;
+
+	INIT_LIST_HEAD(&platform->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&platform->list, &platform_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered platform '%s'\n", platform->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_platform);
+
+/**
+ * snd_soc_unregister_platform - Unregister a platform from the ASoC core
+ *
+ * @param platform platform to unregister
+ */
+void snd_soc_unregister_platform(struct snd_soc_platform *platform)
+{
+	mutex_lock(&client_mutex);
+	list_del(&platform->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered platform '%s'\n", platform->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
+
+/**
+ * snd_soc_register_codec - Register a codec with the ASoC core
+ *
+ * @param codec codec to register
+ */
+int snd_soc_register_codec(struct snd_soc_codec *codec)
+{
+	if (!codec->name)
+		return -EINVAL;
+
+	/* The device should become mandatory over time */
+	if (!codec->dev)
+		printk(KERN_WARNING "No device for codec %s\n", codec->name);
+
+	INIT_LIST_HEAD(&codec->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&codec->list, &codec_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered codec '%s'\n", codec->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_codec);
+
+/**
+ * snd_soc_unregister_codec - Unregister a codec from the ASoC core
+ *
+ * @param codec codec to unregister
+ */
+void snd_soc_unregister_codec(struct snd_soc_codec *codec)
+{
+	mutex_lock(&client_mutex);
+	list_del(&codec->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered codec '%s'\n", codec->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
+
+static int __init snd_soc_init(void)
+{
+#ifdef CONFIG_DEBUG_FS
+	debugfs_root = debugfs_create_dir("asoc", NULL);
+	if (IS_ERR(debugfs_root) || !debugfs_root) {
+		printk(KERN_WARNING
+		       "ASoC: Failed to create debugfs directory\n");
+		debugfs_root = NULL;
+	}
+#endif
+
 	return platform_driver_register(&soc_driver);
 }
 
-static void snd_soc_exit(void)
+static void __exit snd_soc_exit(void)
 {
+#ifdef CONFIG_DEBUG_FS
+	debugfs_remove_recursive(debugfs_root);
+#endif
 	platform_driver_unregister(&soc_driver);
 }
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7351db9606e4..8863eddbac02 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -37,7 +37,6 @@
 #include <linux/bitops.h>
 #include <linux/platform_device.h>
 #include <linux/jiffies.h>
-#include <linux/debugfs.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -67,17 +66,13 @@ static int dapm_status = 1;
 module_param(dapm_status, int, 0);
 MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
 
-static struct dentry *asoc_debugfs;
-
-static u32 pop_time;
-
-static void pop_wait(void)
+static void pop_wait(u32 pop_time)
 {
 	if (pop_time)
 		schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time));
 }
 
-static void pop_dbg(const char *fmt, ...)
+static void pop_dbg(u32 pop_time, const char *fmt, ...)
 {
 	va_list args;
 
@@ -85,7 +80,7 @@ static void pop_dbg(const char *fmt, ...)
 
 	if (pop_time) {
 		vprintk(fmt, args);
-		pop_wait();
+		pop_wait(pop_time);
 	}
 
 	va_end(args);
@@ -230,10 +225,11 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
 
 	change = old != new;
 	if (change) {
-		pop_dbg("pop test %s : %s in %d ms\n", widget->name,
-			widget->power ? "on" : "off", pop_time);
+		pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
+			widget->name, widget->power ? "on" : "off",
+			codec->pop_time);
 		snd_soc_write(codec, widget->reg, new);
-		pop_wait();
+		pop_wait(codec->pop_time);
 	}
 	pr_debug("reg %x old %x new %x change %d\n", widget->reg,
 		 old, new, change);
@@ -293,7 +289,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 	struct snd_soc_dapm_widget *w)
 {
 	int i, ret = 0;
-	char name[32];
+	size_t name_len;
 	struct snd_soc_dapm_path *path;
 
 	/* add kcontrol */
@@ -307,11 +303,16 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 				continue;
 
 			/* add dapm control with long name */
-			snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
-			path->long_name = kstrdup (name, GFP_KERNEL);
+			name_len = 2 + strlen(w->name)
+				+ strlen(w->kcontrols[i].name);
+			path->long_name = kmalloc(name_len, GFP_KERNEL);
 			if (path->long_name == NULL)
 				return -ENOMEM;
 
+			snprintf(path->long_name, name_len, "%s %s",
+				 w->name, w->kcontrols[i].name);
+			path->long_name[name_len - 1] = '\0';
+
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
 				path->long_name);
 			ret = snd_ctl_add(codec->card, path->kcontrol);
@@ -821,23 +822,9 @@ static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
 
 int snd_soc_dapm_sys_add(struct device *dev)
 {
-	int ret = 0;
-
 	if (!dapm_status)
 		return 0;
-
-	ret = device_create_file(dev, &dev_attr_dapm_widget);
-	if (ret != 0)
-		return ret;
-
-	asoc_debugfs = debugfs_create_dir("asoc", NULL);
-	if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
-		debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
-				   &pop_time);
-	else
-		asoc_debugfs = NULL;
-
-	return 0;
+	return device_create_file(dev, &dev_attr_dapm_widget);
 }
 
 static void snd_soc_dapm_sys_remove(struct device *dev)
@@ -845,9 +832,6 @@ static void snd_soc_dapm_sys_remove(struct device *dev)
 	if (dapm_status) {
 		device_remove_file(dev, &dev_attr_dapm_widget);
 	}
-
-	if (asoc_debugfs)
-		debugfs_remove_recursive(asoc_debugfs);
 }
 
 /* free all dapm widgets and resources */
@@ -1006,28 +990,6 @@ err:
 	return ret;
 }
 
-/**
- * snd_soc_dapm_connect_input - connect dapm widgets
- * @codec: audio codec
- * @sink: name of target widget
- * @control: mixer control name
- * @source: name of source name
- *
- * Connects 2 dapm widgets together via a named audio path. The sink is
- * the widget receiving the audio signal, whilst the source is the sender
- * of the audio signal.
- *
- * This function has been deprecated in favour of snd_soc_dapm_add_routes().
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
-	const char *control, const char *source)
-{
-	return snd_soc_dapm_add_route(codec, sink, control, source);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
-
 /**
  * snd_soc_dapm_add_routes - Add routes between DAPM widgets
  * @codec: codec
@@ -1358,8 +1320,12 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
 
 	for (i = 0; i < num; i++) {
 		ret = snd_soc_dapm_new_control(codec, widget);
-		if (ret < 0)
+		if (ret < 0) {
+			printk(KERN_ERR
+			       "ASoC: Failed to create DAPM control %s: %d\n",
+			       widget->name, ret);
 			return ret;
+		}
 		widget++;
 	}
 	return 0;
@@ -1440,11 +1406,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 				enum snd_soc_bias_level level)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0;
 
-	if (machine->set_bias_level)
-		ret = machine->set_bias_level(machine, level);
+	if (card->set_bias_level)
+		ret = card->set_bias_level(card, level);
 	if (ret == 0 && codec->set_bias_level)
 		ret = codec->set_bias_level(codec, level);
 
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 10ba4218161b..2b302bbffe73 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -10,6 +10,7 @@
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/err.h>
+#include <sound/core.h>
 
 #ifdef CONFIG_SOUND_OSS_CORE
 static int __init init_oss_soundcore(void);
diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c
index 798ca124da58..ccd763dd7167 100644
--- a/sound/usb/caiaq/caiaq-control.c
+++ b/sound/usb/caiaq/caiaq-control.c
@@ -247,69 +247,56 @@ static struct caiaq_controller a8dj_controller[] = {
 	{ "Software lock", 			40 		}
 };
 
-int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev)
+static int __devinit add_controls(struct caiaq_controller *c, int num,
+				  struct snd_usb_caiaqdev *dev)
 {
-	int i;
+	int i, ret;
 	struct snd_kcontrol *kc;
 
-	switch (dev->chip.usb_id) {
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
-		for (i = 0; i < ARRAY_SIZE(ak1_controller); i++) {
-			struct caiaq_controller *c = ak1_controller + i;
-			kcontrol_template.name = c->name;
-			kcontrol_template.private_value = c->index;
-			kc = snd_ctl_new1(&kcontrol_template, dev);
-			snd_ctl_add(dev->chip.card, kc);
-		}
-
-		break;
-
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
-		for (i = 0; i < ARRAY_SIZE(rk2_controller); i++) {
-			struct caiaq_controller *c = rk2_controller + i;
-			kcontrol_template.name = c->name;
-			kcontrol_template.private_value = c->index;
-			kc = snd_ctl_new1(&kcontrol_template, dev);
-			snd_ctl_add(dev->chip.card, kc);
-		}
-
-		break;
-
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
-		for (i = 0; i < ARRAY_SIZE(rk3_controller); i++) {
-			struct caiaq_controller *c = rk3_controller + i;
-			kcontrol_template.name = c->name;
-			kcontrol_template.private_value = c->index;
-			kc = snd_ctl_new1(&kcontrol_template, dev);
-			snd_ctl_add(dev->chip.card, kc);
-		}
-
-		break;
-
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER):
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2):
-		for (i = 0; i < ARRAY_SIZE(kore_controller); i++) {
-			struct caiaq_controller *c = kore_controller + i;
-			kcontrol_template.name = c->name;
-			kcontrol_template.private_value = c->index;
-			kc = snd_ctl_new1(&kcontrol_template, dev);
-			snd_ctl_add(dev->chip.card, kc);
-		}
-
-		break;
-
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
-		for (i = 0; i < ARRAY_SIZE(a8dj_controller); i++) {
-			struct caiaq_controller *c = a8dj_controller + i;
-			kcontrol_template.name = c->name;
-			kcontrol_template.private_value = c->index;
-			kc = snd_ctl_new1(&kcontrol_template, dev);
-			snd_ctl_add(dev->chip.card, kc);
-		}
-
-		break;
+	for (i = 0; i < num; i++, c++) {
+		kcontrol_template.name = c->name;
+		kcontrol_template.private_value = c->index;
+		kc = snd_ctl_new1(&kcontrol_template, dev);
+		ret = snd_ctl_add(dev->chip.card, kc);
+		if (ret < 0)
+			return ret;
 	}
 
 	return 0;
 }
 
+int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev)
+{
+	int ret = 0;
+
+	switch (dev->chip.usb_id) {
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
+		ret = add_controls(ak1_controller,
+			ARRAY_SIZE(ak1_controller), dev);
+		break;
+
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
+		ret = add_controls(rk2_controller,
+			ARRAY_SIZE(rk2_controller), dev);
+		break;
+
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+		ret = add_controls(rk3_controller,
+			ARRAY_SIZE(rk3_controller), dev);
+		break;
+
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER):
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2):
+		ret = add_controls(kore_controller,
+			ARRAY_SIZE(kore_controller), dev);
+		break;
+
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
+		ret = add_controls(a8dj_controller,
+			ARRAY_SIZE(a8dj_controller), dev);
+		break;
+	}
+
+	return ret;
+}
+
diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index 83175083e50f..b143ef7152f7 100644
--- a/sound/usb/caiaq/caiaq-device.c
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -42,7 +42,7 @@
 #endif
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.9");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 5962e4b84423..6d9f9b135c62 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -880,7 +880,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
 				snd_rawmidi_transmit_ack(substream, 1);
 			return;
 		}
-		tasklet_hi_schedule(&port->ep->tasklet);
+		tasklet_schedule(&port->ep->tasklet);
 	}
 }
 
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index ff23cc1ce3b9..70b96355ca4c 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -276,7 +276,8 @@ static void subs_set_complete(struct urb **urbs, void (*complete)(struct urb *))
 	}
 }
 
-int usb_stream_prepare_playback(struct usb_stream_kernel *sk, struct urb *inurb)
+static int usb_stream_prepare_playback(struct usb_stream_kernel *sk,
+		struct urb *inurb)
 {
 	struct usb_stream *s = sk->s;
 	struct urb *io;