Merge commit 'sound/master'

89 files changed, 2786 insertions, 1610 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 0bbee38acd26..e59569462cb9 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1091,7 +1091,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     This occurs when the access to non-existing or non-working codec slot
     (likely a modem one) causes a stall of the communication via HD-audio
     bus.  You can see which codec slots are probed by enabling
-    CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec
+    CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec
     proc files.  Then limit the slots to probe by probe_mask option.
     For example, probe_mask=1 means to probe only the first slot, and
     probe_mask=4 means only the third slot.
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index b03df4d4795c..e13c4e67029f 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -6127,8 +6127,8 @@ struct _snd_pcm_runtime {
 
       <para>
         <function>snd_printdd()</function> is compiled in only when
-      <constant>CONFIG_SND_DEBUG_DETECT</constant> is set. Please note
-      that <constant>DEBUG_DETECT</constant> is not set as default
+      <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is set. Please note
+      that <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is not set as default
       even if you configure the alsa-driver with
       <option>--with-debug=full</option> option. You need to give
       explicitly <option>--with-debug=detect</option> option instead. 
diff --git a/include/sound/core.h b/include/sound/core.h
index 695ee53488a3..558b96284bd2 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -412,13 +412,13 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 
 #endif /* CONFIG_SND_DEBUG */
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 /**
  * snd_printdd - debug printk
  * @format: format string
  *
  * Works like snd_printk() for debugging purposes.
- * Ignored when CONFIG_SND_DEBUG_DETECT is not set.
+ * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
  */
 #define snd_printdd(format, args...) snd_printk(format, ##args)
 #else
@@ -442,7 +442,7 @@ struct snd_pci_quirk {
 	unsigned short subvendor;	/* PCI subvendor ID */
 	unsigned short subdevice;	/* PCI subdevice ID */
 	int value;			/* value */
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	const char *name;		/* name of the device (optional) */
 #endif
 };
@@ -450,7 +450,7 @@ struct snd_pci_quirk {
 #define _SND_PCI_QUIRK_ID(vend,dev) \
 	.subvendor = (vend), .subdevice = (dev)
 #define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 #define SND_PCI_QUIRK(vend,dev,xname,val) \
 	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)}
 #else
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index a105b01e06d5..f8223fae5804 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -193,6 +193,7 @@ struct snd_soc_dapm_widget;
 enum snd_soc_dapm_type;
 struct snd_soc_dapm_path;
 struct snd_soc_dapm_pin;
+struct snd_soc_dapm_route;
 
 /* dapm controls */
 int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@@ -205,17 +206,23 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
 	const struct snd_soc_dapm_widget *widget);
+int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+	const struct snd_soc_dapm_widget *widget,
+	int num);
 
 /* dapm path setup */
-int snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
+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,
+			    const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
 	int event);
-int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event);
+int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+	enum snd_soc_bias_level level);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
@@ -223,6 +230,8 @@ int snd_soc_dapm_sys_add(struct device *dev);
 /* dapm audio endpoint control */
 int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
 	char *pin, int status);
+int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec,
+	char *pin);
 int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
 
 /* dapm widget types */
@@ -245,6 +254,18 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_post,			/* machine specific post widget - exec last */
 };
 
+/*
+ * DAPM audio route definition.
+ *
+ * Defines an audio route originating at source via control and finishing
+ * at sink.
+ */
+struct snd_soc_dapm_route {
+	const char *sink;
+	const char *control;
+	const char *source;
+};
+
 /* dapm audio path between two widgets */
 struct snd_soc_dapm_path {
 	char *name;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index d3c8c033dff8..bca9538d9e50 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -103,6 +103,24 @@
 	.private_value = (unsigned long)&xenum }
 
 /*
+ * Bias levels
+ *
+ * @ON:      Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ *           stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ *           in progress. NOTE: The transition time between STANDBY and ON
+ *           should be as fast as possible and no longer than 10ms.
+ * @OFF:     Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_PREPARE,
+	SND_SOC_BIAS_STANDBY,
+	SND_SOC_BIAS_OFF,
+};
+
+/*
  * Digital Audio Interface (DAI) types
  */
 #define SND_SOC_DAI_AC97	0x1
@@ -272,9 +290,9 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-/* ASoC codec DAI ops */
-struct snd_soc_codec_ops {
-	/* codec DAI clocking configuration */
+/* ASoC DAI ops */
+struct snd_soc_dai_ops {
+	/* DAI clocking configuration */
 	int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir);
 	int (*set_pll)(struct snd_soc_codec_dai *codec_dai,
@@ -282,7 +300,7 @@ struct snd_soc_codec_ops {
 	int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai,
 		int div_id, int div);
 
-	/* CPU DAI format configuration */
+	/* DAI format configuration */
 	int (*set_fmt)(struct snd_soc_codec_dai *codec_dai,
 		unsigned int fmt);
 	int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai,
@@ -293,24 +311,6 @@ struct snd_soc_codec_ops {
 	int (*digital_mute)(struct snd_soc_codec_dai *, int mute);
 };
 
-/* ASoC cpu DAI ops */
-struct snd_soc_cpu_ops {
-	/* CPU DAI clocking configuration */
-	int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai,
-		int clk_id, unsigned int freq, int dir);
-	int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai,
-		int div_id, int div);
-	int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out);
-
-	/* CPU DAI format configuration */
-	int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai,
-		unsigned int fmt);
-	int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai,
-		unsigned int mask, int slots);
-	int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate);
-};
-
 /* SoC Codec DAI */
 struct snd_soc_codec_dai {
 	char *name;
@@ -328,7 +328,7 @@ struct snd_soc_codec_dai {
 
 	/* ops */
 	struct snd_soc_ops ops;
-	struct snd_soc_codec_ops dai_ops;
+	struct snd_soc_dai_ops dai_ops;
 
 	/* DAI private data */
 	void *private_data;
@@ -352,7 +352,7 @@ struct snd_soc_cpu_dai {
 
 	/* ops */
 	struct snd_soc_ops ops;
-	struct snd_soc_cpu_ops dai_ops;
+	struct snd_soc_dai_ops dai_ops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
@@ -374,7 +374,8 @@ struct snd_soc_codec {
 	struct mutex mutex;
 
 	/* callbacks */
-	int (*dapm_event)(struct snd_soc_codec *codec, int event);
+	int (*set_bias_level)(struct snd_soc_codec *,
+			      enum snd_soc_bias_level level);
 
 	/* runtime */
 	struct snd_card *card;
@@ -396,8 +397,8 @@ struct snd_soc_codec {
 	/* dapm */
 	struct list_head dapm_widgets;
 	struct list_head dapm_paths;
-	unsigned int dapm_state;
-	unsigned int suspend_dapm_state;
+	enum snd_soc_bias_level bias_level;
+	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
 
 	/* codec DAI's */
@@ -467,7 +468,8 @@ struct snd_soc_machine {
 	int (*resume_post)(struct platform_device *pdev);
 
 	/* callbacks */
-	int (*dapm_event)(struct snd_soc_machine *, int event);
+	int (*set_bias_level)(struct snd_soc_machine *,
+			      enum snd_soc_bias_level level);
 
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;
diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h
index 2e564bfb37fe..110d5dc3a2be 100644
--- a/include/sound/uda1341.h
+++ b/include/sound/uda1341.h
@@ -15,8 +15,6 @@
  *                           features support
  */
 
-/* $Id: uda1341.h,v 1.8 2005/11/17 14:17:21 tiwai Exp $ */
-
 #define UDA1341_ALSA_NAME "snd-uda1341"
 
 /*
diff --git a/sound/Kconfig b/sound/Kconfig
index 4247406160e7..a37bee094eba 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -1,11 +1,9 @@
 # sound/Config.in
 #
 
-menu "Sound"
-	depends on HAS_IOMEM
-
-config SOUND
+menuconfig SOUND
 	tristate "Sound card support"
+	depends on HAS_IOMEM
 	help
 	  If you have a sound card in your computer, i.e. if it can say more
 	  than an occasional beep, say Y.  Be sure to have all the information
@@ -28,22 +26,22 @@ config SOUND
 	  and read <file:Documentation/sound/oss/README.modules>; the module
 	  will be called soundcore.
 
+if SOUND
+
 source "sound/oss/dmasound/Kconfig"
 
 if !M68K
 
-menu "Advanced Linux Sound Architecture"
-	depends on SOUND!=n
-
-config SND
+menuconfig SND
 	tristate "Advanced Linux Sound Architecture"
-	depends on SOUND
 	help
 	  Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture),
 	  the new base sound system.
 
 	  For more information, see <http://www.alsa-project.org/>
 
+if SND
+
 source "sound/core/Kconfig"
 
 source "sound/drivers/Kconfig"
@@ -58,9 +56,7 @@ source "sound/aoa/Kconfig"
 
 source "sound/arm/Kconfig"
 
-if SPI
 source "sound/spi/Kconfig"
-endif
 
 source "sound/mips/Kconfig"
 
@@ -80,22 +76,20 @@ source "sound/parisc/Kconfig"
 
 source "sound/soc/Kconfig"
 
-endmenu
+endif # SND
 
-menu "Open Sound System"
-	depends on SOUND!=n
-
-config SOUND_PRIME
+menuconfig SOUND_PRIME
 	tristate "Open Sound System (DEPRECATED)"
-	depends on SOUND
 	help
 	  Say 'Y' or 'M' to enable Open Sound System drivers.
 
+if SOUND_PRIME
+
 source "sound/oss/Kconfig"
 
-endmenu
+endif # SOUND_PRIME
 
-endif
+endif # !M68K
 
 config AC97_BUS
 	tristate
@@ -105,4 +99,4 @@ config AC97_BUS
 	  sound although they're sharing the AC97 bus. Concerned drivers
 	  should "select" this.
 
-endmenu
+endif # SOUND
diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig
index 5d5813cec4c8..c081e18b9540 100644
--- a/sound/aoa/Kconfig
+++ b/sound/aoa/Kconfig
@@ -1,18 +1,17 @@
-menu "Apple Onboard Audio driver"
-	depends on SND!=n && PPC_PMAC
-
-config SND_AOA
+menuconfig SND_AOA
 	tristate "Apple Onboard Audio driver"
-	depends on SND
+	depends on PPC_PMAC
 	select SND_PCM
 	---help---
 	This option enables the new driver for the various
 	Apple Onboard Audio components.
 
+if SND_AOA
+
 source "sound/aoa/fabrics/Kconfig"
 
 source "sound/aoa/codecs/Kconfig"
 
 source "sound/aoa/soundbus/Kconfig"
 
-endmenu
+endif	# SND_AOA
diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig
index d5fbd6016e93..808eb11ebacd 100644
--- a/sound/aoa/codecs/Kconfig
+++ b/sound/aoa/codecs/Kconfig
@@ -1,6 +1,5 @@
 config SND_AOA_ONYX
 	tristate "support Onyx chip"
-	depends on SND_AOA
 	select I2C
 	select I2C_POWERMAC
 	---help---
@@ -10,7 +9,6 @@ config SND_AOA_ONYX
 
 #config SND_AOA_TOPAZ
 #	tristate "support Topaz chips"
-#	depends on SND_AOA
 #	---help---
 #	This option enables support for the Topaz (CS84xx)
 #	codec chips found in the latest Apple machines,
@@ -19,7 +17,6 @@ config SND_AOA_ONYX
 
 config SND_AOA_TAS
 	tristate "support TAS chips"
-	depends on SND_AOA
 	select I2C
 	select I2C_POWERMAC
 	---help---
@@ -29,7 +26,6 @@ config SND_AOA_TAS
 
 config SND_AOA_TOONIE
 	tristate "support Toonie chip"
-	depends on SND_AOA
 	---help---
 	This option enables support for the toonie codec
 	found in the Mac Mini. If you have a Mac Mini and
diff --git a/sound/aoa/fabrics/Kconfig b/sound/aoa/fabrics/Kconfig
index 50d7021ff677..3ca475a886b1 100644
--- a/sound/aoa/fabrics/Kconfig
+++ b/sound/aoa/fabrics/Kconfig
@@ -1,6 +1,5 @@
 config SND_AOA_FABRIC_LAYOUT
 	tristate "layout-id fabric"
-	depends on SND_AOA
 	select SND_AOA_SOUNDBUS
 	select SND_AOA_SOUNDBUS_I2S
 	---help---
diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig
index 7368b7ddfe0d..839d1137b9b2 100644
--- a/sound/aoa/soundbus/Kconfig
+++ b/sound/aoa/soundbus/Kconfig
@@ -1,6 +1,5 @@
 config SND_AOA_SOUNDBUS
 	tristate "Apple Soundbus support"
-	depends on SOUND
 	select SND_PCM
 	---help---
 	This option enables the generic driver for the soundbus
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 2e4a5e0d16db..351e19ea3785 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -1,11 +1,19 @@
 # ALSA ARM drivers
 
-menu "ALSA ARM devices"
-	depends on SND!=n && ARM
+menuconfig SND_ARM
+	bool "ARM sound devices"
+	depends on ARM
+	default y
+	help
+	  Support for sound devices specific to ARM architectures.
+	  Drivers that are implemented on ASoC can be found in
+	  "ALSA for SoC audio support" section.
+
+if SND_ARM
 
 config SND_SA11XX_UDA1341
 	tristate "SA11xx UDA1341TS driver (iPaq H3600)"
-	depends on ARCH_SA1100 && SND && L3
+	depends on ARCH_SA1100 && L3
 	select SND_PCM
 	help
 	  Say Y here if you have a Compaq iPaq H3x00 handheld computer
@@ -16,7 +24,7 @@ config SND_SA11XX_UDA1341
 
 config SND_ARMAACI
 	tristate "ARM PrimeCell PL041 AC Link support"
-	depends on SND && ARM_AMBA
+	depends on ARM_AMBA
 	select SND_PCM
 	select SND_AC97_CODEC
 
@@ -26,11 +34,12 @@ config SND_PXA2XX_PCM
 
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
-	depends on ARCH_PXA && SND
+	depends on ARCH_PXA
 	select SND_PXA2XX_PCM
 	select SND_AC97_CODEC
 	help
 	  Say Y or M if you want to support any AC97 codec attached to
 	  the PXA2xx AC97 interface.
 
-endmenu
+endif	# SND_ARM
+
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
index 0eff33ca0f79..faeddf3ecedb 100644
--- a/sound/arm/sa11xx-uda1341.c
+++ b/sound/arm/sa11xx-uda1341.c
@@ -21,8 +21,6 @@
  *                              merged HAL layer (patches from Brian)
  */
 
-/* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */
-
 /***************************************************************************************************
 *
 * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index a8d71c6c8e75..335d45ecde6a 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -1,24 +1,19 @@
 # ALSA soundcard-configuration
 config SND_TIMER
 	tristate
-	depends on SND
 
 config SND_PCM
 	tristate
 	select SND_TIMER
-	depends on SND
 
 config SND_HWDEP
 	tristate
-	depends on SND
 
 config SND_RAWMIDI
 	tristate
-	depends on SND
 
 config SND_SEQUENCER
 	tristate "Sequencer support"
-	depends on SND
 	select SND_TIMER
 	help
 	  Say Y or M to enable MIDI sequencer and router support.  This
@@ -44,11 +39,9 @@ config SND_SEQ_DUMMY
 
 config SND_OSSEMUL
 	bool
-	depends on SND
 
 config SND_MIXER_OSS
 	tristate "OSS Mixer API"
-	depends on SND
 	select SND_OSSEMUL
 	help
 	  To enable OSS mixer API emulation (/dev/mixer*), say Y here
@@ -61,7 +54,6 @@ config SND_MIXER_OSS
 
 config SND_PCM_OSS
 	tristate "OSS PCM (digital audio) API"
-	depends on SND
 	select SND_OSSEMUL
 	select SND_PCM
 	help
@@ -84,7 +76,7 @@ config SND_PCM_OSS_PLUGINS
 
 config SND_SEQUENCER_OSS
 	bool "OSS Sequencer API"
-	depends on SND && SND_SEQUENCER
+	depends on SND_SEQUENCER
 	select SND_OSSEMUL
 	help
 	  Say Y here to enable OSS sequencer emulation (both
@@ -98,7 +90,7 @@ config SND_SEQUENCER_OSS
 
 config SND_RTCTIMER
 	tristate "RTC Timer support"
-	depends on SND && RTC
+	depends on RTC
 	select SND_TIMER
 	help
 	  Say Y here to enable RTC timer support for ALSA.  ALSA uses
@@ -123,7 +115,6 @@ config SND_SEQ_RTCTIMER_DEFAULT
 
 config SND_DYNAMIC_MINORS
 	bool "Dynamic device file minor numbers"
-	depends on SND
 	help
 	  If you say Y here, the minor numbers of ALSA device files in
 	  /dev/snd/ are allocated dynamically.  This allows you to have
@@ -134,7 +125,6 @@ config SND_DYNAMIC_MINORS
 
 config SND_SUPPORT_OLD_API
 	bool "Support old ALSA API"
-	depends on SND
 	default y
 	help
 	  Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3
@@ -142,7 +132,7 @@ config SND_SUPPORT_OLD_API
 
 config SND_VERBOSE_PROCFS
 	bool "Verbose procfs contents"
-	depends on SND && PROC_FS
+	depends on PROC_FS
 	default y
 	help
 	  Say Y here to include code for verbose procfs contents (provides
@@ -151,7 +141,6 @@ config SND_VERBOSE_PROCFS
 
 config SND_VERBOSE_PRINTK
 	bool "Verbose printk"
-	depends on SND
 	help
 	  Say Y here to enable verbose log messages.  These messages
 	  will help to identify source file and position containing
@@ -161,16 +150,17 @@ config SND_VERBOSE_PRINTK
 
 config SND_DEBUG
 	bool "Debug"
-	depends on SND
 	help
 	  Say Y here to enable ALSA debug code.
 
-config SND_DEBUG_DETECT
-	bool "Debug detection"
+config SND_DEBUG_VERBOSE
+	bool "More verbose debug"
 	depends on SND_DEBUG
 	help
-	  Say Y here to enable extra-verbose log messages printed when
-	  detecting devices.
+	  Say Y here to enable extra-verbose debugging messages.
+	  
+	  Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages.
+	  So, say Y only if you are ready to be annoyed.
 
 config SND_PCM_XRUN_DEBUG
 	bool "Enable PCM ring buffer overrun/underrun debugging"
@@ -184,4 +174,3 @@ config SND_PCM_XRUN_DEBUG
 
 config SND_VMASTER
 	bool
-	depends on SND
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 23b7bc02728b..f5d6d8d12979 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -80,68 +80,6 @@ struct snd_mem_list {
 #endif
 
 /*
- *  Hacks
- */
-
-#if defined(__i386__)
-/*
- * A hack to allocate large buffers via dma_alloc_coherent()
- *
- * since dma_alloc_coherent always tries GFP_DMA when the requested
- * pci memory region is below 32bit, it happens quite often that even
- * 2 order of pages cannot be allocated.
- *
- * so in the following, we allocate at first without dma_mask, so that
- * allocation will be done without GFP_DMA.  if the area doesn't match
- * with the requested region, then realloate with the original dma_mask
- * again.
- *
- * Really, we want to move this type of thing into dma_alloc_coherent()
- * so dma_mask doesn't have to be messed with.
- */
-
-static void *snd_dma_hack_alloc_coherent(struct device *dev, size_t size,
-					 dma_addr_t *dma_handle,
-					 gfp_t flags)
-{
-	void *ret;
-	u64 dma_mask, coherent_dma_mask;
-
-	if (dev == NULL || !dev->dma_mask)
-		return dma_alloc_coherent(dev, size, dma_handle, flags);
-	dma_mask = *dev->dma_mask;
-	coherent_dma_mask = dev->coherent_dma_mask;
-	*dev->dma_mask = 0xffffffff; 	/* do without masking */
-	dev->coherent_dma_mask = 0xffffffff; 	/* do without masking */
-	ret = dma_alloc_coherent(dev, size, dma_handle, flags);
-	*dev->dma_mask = dma_mask;	/* restore */
-	dev->coherent_dma_mask = coherent_dma_mask;	/* restore */
-	if (ret) {
-		/* obtained address is out of range? */
-		if (((unsigned long)*dma_handle + size - 1) & ~dma_mask) {
-			/* reallocate with the proper mask */
-			dma_free_coherent(dev, size, ret, *dma_handle);
-			ret = dma_alloc_coherent(dev, size, dma_handle, flags);
-		}
-	} else {
-		/* wish to success now with the proper mask... */
-		if (dma_mask != 0xffffffffUL) {
-			/* allocation with GFP_ATOMIC to avoid the long stall */
-			flags &= ~GFP_KERNEL;
-			flags |= GFP_ATOMIC;
-			ret = dma_alloc_coherent(dev, size, dma_handle, flags);
-		}
-	}
-	return ret;
-}
-
-/* redefine dma_alloc_coherent for some architectures */
-#undef dma_alloc_coherent
-#define dma_alloc_coherent snd_dma_hack_alloc_coherent
-
-#endif /* arch */
-
-/*
  *
  *  Generic memory allocators
  *
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index 602b58e3b55d..255fd18b9aec 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -1,15 +1,41 @@
-# ALSA generic drivers
+config SND_MPU401_UART
+        tristate
+        select SND_RAWMIDI
 
-menu "Generic devices"
-	depends on SND!=n
+config SND_OPL3_LIB
+	tristate
+	select SND_TIMER
+	select SND_HWDEP
 
+config SND_OPL4_LIB
+	tristate
+	select SND_TIMER
+	select SND_HWDEP
+
+config SND_VX_LIB
+	tristate
+	select SND_HWDEP
+	select SND_PCM
+
+config SND_AC97_CODEC
+	tristate
+	select SND_PCM
+	select AC97_BUS
+	select SND_VMASTER
+
+menuconfig SND_DRIVERS
+	bool "Generic sound devices"
+	default y
+	help
+	  Support for generic sound devices.
+  
+if SND_DRIVERS
 
 config SND_PCSP
 	tristate "PC-Speaker support (READ HELP!)"
 	depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS
 	depends on INPUT
 	depends on EXPERIMENTAL
-	depends on SND
 	select SND_PCM
 	help
 	  If you don't have a sound card in your computer, you can include a
@@ -35,33 +61,8 @@ config SND_PCSP
 	  Say M if you don't.
 	  Say Y only if you really know what you do.
 
-config SND_MPU401_UART
-        tristate
-        select SND_RAWMIDI
-
-config SND_OPL3_LIB
-	tristate
-	select SND_TIMER
-	select SND_HWDEP
-
-config SND_OPL4_LIB
-	tristate
-	select SND_TIMER
-	select SND_HWDEP
-
-config SND_VX_LIB
-	tristate
-	select SND_HWDEP
-	select SND_PCM
-
-config SND_AC97_CODEC
-	tristate
-	select SND_PCM
-	select AC97_BUS
-
 config SND_DUMMY
 	tristate "Dummy (/dev/null) soundcard"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include the dummy driver.  This driver does
@@ -90,7 +91,6 @@ config SND_VIRMIDI
 
 config SND_MTPAV
 	tristate "MOTU MidiTimePiece AV multiport MIDI"
-	depends on SND
 	select SND_RAWMIDI
 	help
 	  To use a MOTU MidiTimePiece AV multiport MIDI adapter
@@ -102,7 +102,7 @@ config SND_MTPAV
 
 config SND_MTS64
 	tristate "ESI Miditerminal 4140 driver"
-	depends on SND && PARPORT
+	depends on PARPORT
 	select SND_RAWMIDI
 	help
 	  The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with 
@@ -115,7 +115,6 @@ config SND_MTS64
 
 config SND_SERIAL_U16550
 	tristate "UART16550 serial MIDI driver"
-	depends on SND
 	select SND_RAWMIDI
 	help
 	  To include support for MIDI serial port interfaces, say Y here
@@ -131,7 +130,6 @@ config SND_SERIAL_U16550
 
 config SND_MPU401
 	tristate "Generic MPU-401 UART driver"
-	depends on SND
 	select SND_MPU401_UART
 	help
 	  Say Y here to include support for MIDI ports compatible with
@@ -142,7 +140,7 @@ config SND_MPU401
 
 config SND_PORTMAN2X4
 	tristate "Portman 2x4 driver"
-	depends on SND && PARPORT
+	depends on PARPORT
 	select SND_RAWMIDI
 	help
 	  Say Y here to include support for Midiman Portman 2x4 parallel
@@ -153,7 +151,7 @@ config SND_PORTMAN2X4
 
 config SND_ML403_AC97CR
 	tristate "Xilinx ML403 AC97 Controller Reference"
-	depends on SND && XILINX_VIRTEX
+	depends on XILINX_VIRTEX
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the
@@ -163,4 +161,25 @@ config SND_ML403_AC97CR
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-ml403_ac97cr.
 
-endmenu
+config SND_AC97_POWER_SAVE
+	bool "AC97 Power-Saving Mode"
+	depends on SND_AC97_CODEC && EXPERIMENTAL
+	default n
+	help
+	  Say Y here to enable the aggressive power-saving support of
+	  AC97 codecs.  In this mode, the power-mode is dynamically
+	  controlled at each open/close.
+
+	  The mode is activated by passing power_save=1 option to
+	  snd-ac97-codec driver.  You can toggle it dynamically over
+	  sysfs, too.
+
+config SND_AC97_POWER_SAVE_DEFAULT
+	int "Default time-out for AC97 power-save mode"
+	depends on SND_AC97_POWER_SAVE
+	default 0
+	help
+	  The default time-out value in seconds for AC97 automatic
+	  power-save mode.  0 means to disable the power-save mode.
+
+endif	# SND_DRIVERS
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index e57e9cbe6a0f..9c3d361accfb 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -23,6 +23,7 @@
 #include <linux/slab.h>
 #include <linux/delay.h>
 #include <linux/init.h>
+#include <asm/unaligned.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/pcm.h>
@@ -264,10 +265,7 @@ int snd_cs8427_create(struct snd_i2c_bus *bus,
 		goto __fail;
 	}
 	/* write default channel status bytes */
-	buf[0] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 0));
-	buf[1] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 8));
-	buf[2] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 16));
-	buf[3] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 24));
+	put_unaligned_le32(SNDRV_PCM_DEFAULT_CON_SPDIF, buf);
 	memset(buf + 4, 0, 24 - 4);
 	if (snd_cs8427_send_corudata(device, 0, buf, 24) < 0)
 		goto __fail;
diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c
index bfa5d2c3608b..1f4942ea1414 100644
--- a/sound/i2c/l3/uda1341.c
+++ b/sound/i2c/l3/uda1341.c
@@ -17,8 +17,6 @@
  * 2002-05-12   Tomas Kasparek  another code cleanup
  */
 
-/* $Id: uda1341.c,v 1.18 2005/11/17 14:17:21 tiwai Exp $ */
-
 #include <linux/module.h>
 #include <linux/init.h>
 #include <linux/types.h>
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 2639a6ab8f2e..4575ba865910 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -21,12 +21,17 @@ config SND_SB16_DSP
         select SND_PCM
         select SND_SB_COMMON
 
-menu "ISA devices"
-	depends on SND!=n && ISA && ISA_DMA_API
+menuconfig SND_ISA
+	bool "ISA sound devices"
+	depends on ISA && ISA_DMA_API
+	default y
+	help
+	  Support for sound devices connected via the ISA bus.
+
+if SND_ISA
 
 config SND_ADLIB
 	tristate "AdLib FM card"
-	depends on SND
 	select SND_OPL3_LIB
 	help
 	  Say Y here to include support for AdLib FM cards.
@@ -36,7 +41,7 @@ config SND_ADLIB
 
 config SND_AD1816A
 	tristate "Analog Devices SoundPort AD1816A"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -50,7 +55,6 @@ config SND_AD1816A
 
 config SND_AD1848
 	tristate "Generic AD1848/CS4248 driver"
-	depends on SND
 	select SND_AD1848_LIB
 	help
 	  Say Y here to include support for AD1848 (Analog Devices) or
@@ -64,7 +68,7 @@ config SND_AD1848
 
 config SND_ALS100
 	tristate "Avance Logic ALS100/ALS120"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -78,7 +82,7 @@ config SND_ALS100
 
 config SND_AZT2320
 	tristate "Aztech Systems AZT2320"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -92,7 +96,6 @@ config SND_AZT2320
 
 config SND_CMI8330
 	tristate "C-Media CMI8330"
-	depends on SND
 	select SND_AD1848_LIB
 	select SND_SB16_DSP
 	help
@@ -104,7 +107,6 @@ config SND_CMI8330
 
 config SND_CS4231
 	tristate "Generic Cirrus Logic CS4231 driver"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
 	help
@@ -116,7 +118,6 @@ config SND_CS4231
 
 config SND_CS4232
 	tristate "Generic Cirrus Logic CS4232 driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -129,7 +130,6 @@ config SND_CS4232
 
 config SND_CS4236
 	tristate "Generic Cirrus Logic CS4236+ driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -142,7 +142,7 @@ config SND_CS4236
 
 config SND_DT019X
 	tristate "Diamond Technologies DT-019X, Avance Logic ALS-007"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -156,7 +156,7 @@ config SND_DT019X
 
 config SND_ES968
 	tristate "Generic ESS ES968 driver"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_MPU401_UART
 	select SND_SB8_DSP
@@ -168,7 +168,6 @@ config SND_ES968
 
 config SND_ES1688
 	tristate "Generic ESS ES688/ES1688 driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -181,7 +180,6 @@ config SND_ES1688
 
 config SND_ES18XX
 	tristate "Generic ESS ES18xx driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -193,7 +191,7 @@ config SND_ES18XX
 
 config SND_SC6000
 	tristate "Gallant SC-6000, Audio Excel DSP 16"
-	depends on SND && HAS_IOPORT
+	depends on HAS_IOPORT
 	select SND_AD1848_LIB
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -209,7 +207,6 @@ config SND_GUS_SYNTH
 
 config SND_GUSCLASSIC
 	tristate "Gravis UltraSound Classic"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_PCM
 	select SND_GUS_SYNTH
@@ -222,7 +219,6 @@ config SND_GUSCLASSIC
 
 config SND_GUSEXTREME
 	tristate "Gravis UltraSound Extreme"
-	depends on SND
 	select SND_HWDEP
 	select SND_MPU401_UART
 	select SND_PCM
@@ -236,7 +232,6 @@ config SND_GUSEXTREME
 
 config SND_GUSMAX
 	tristate "Gravis UltraSound MAX"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_CS4231_LIB
 	select SND_GUS_SYNTH
@@ -249,7 +244,7 @@ config SND_GUSMAX
 
 config SND_INTERWAVE
 	tristate "AMD InterWave, Gravis UltraSound PnP"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select SND_RAWMIDI
 	select SND_CS4231_LIB
 	select SND_GUS_SYNTH
@@ -263,7 +258,7 @@ config SND_INTERWAVE
 
 config SND_INTERWAVE_STB
 	tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select SND_RAWMIDI
 	select SND_CS4231_LIB
 	select SND_GUS_SYNTH
@@ -277,7 +272,6 @@ config SND_INTERWAVE_STB
 
 config SND_OPL3SA2
 	tristate "Yamaha OPL3-SA2/SA3"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -290,7 +284,6 @@ config SND_OPL3SA2
 
 config SND_OPTI92X_AD1848
 	tristate "OPTi 82C92x - AD1848"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_OPL4_LIB
 	select SND_MPU401_UART
@@ -304,7 +297,6 @@ config SND_OPTI92X_AD1848
 
 config SND_OPTI92X_CS4231
 	tristate "OPTi 82C92x - CS4231"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_OPL4_LIB
 	select SND_MPU401_UART
@@ -318,7 +310,6 @@ config SND_OPTI92X_CS4231
 
 config SND_OPTI93X
 	tristate "OPTi 82C93x"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -331,7 +322,6 @@ config SND_OPTI93X
 
 config SND_MIRO
 	tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver"
-	depends on SND
 	select SND_OPL4_LIB
 	select SND_CS4231_LIB
 	select SND_MPU401_UART
@@ -345,7 +335,6 @@ config SND_MIRO
 
 config SND_SB8
 	tristate "Sound Blaster 1.0/2.0/Pro (8-bit)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_RAWMIDI
 	select SND_SB8_DSP
@@ -358,7 +347,6 @@ config SND_SB8
 
 config SND_SB16
 	tristate "Sound Blaster 16 (PnP)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_SB16_DSP
@@ -371,7 +359,6 @@ config SND_SB16
 
 config SND_SBAWE
 	tristate "Sound Blaster AWE (32,64) (PnP)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_SB16_DSP
@@ -402,7 +389,6 @@ config SND_SB16_CSP_FIRMWARE_IN_KERNEL
 
 config SND_SGALAXY
 	tristate "Aztech Sound Galaxy"
-	depends on SND
 	select SND_AD1848_LIB
 	help
 	  Say Y here to include support for Aztech Sound Galaxy
@@ -413,7 +399,6 @@ config SND_SGALAXY
 
 config SND_SSCAPE
 	tristate "Ensoniq SoundScape PnP driver"
-	depends on SND
 	select SND_HWDEP
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -426,7 +411,6 @@ config SND_SSCAPE
 
 config SND_WAVEFRONT
 	tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)"
-	depends on SND
 	select FW_LOADER
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -448,4 +432,5 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL
 	  you need to install the firmware files from the
 	  alsa-firmware package.
 
-endmenu
+endif	# SND_ISA
+
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
index 531f8ba96a71..bb26f6cf4c0a 100644
--- a/sound/mips/Kconfig
+++ b/sound/mips/Kconfig
@@ -1,15 +1,21 @@
 # ALSA MIPS drivers
 
-menu "ALSA MIPS devices"
-	depends on SND!=n && MIPS
+menuconfig SND_MIPS
+	bool "MIPS sound devices"
+	depends on MIPS
+	default y
+	help
+	  Support for sound devices of MIPS architectures.
+
+if SND_MIPS
 
 config SND_AU1X00
 	tristate "Au1x00 AC97 Port Driver"
-	depends on (SOC_AU1000 || SOC_AU1100 || SOC_AU1500) && SND
+	depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
 	  ALSA Sound driver for the Au1x00's AC97 port.
 
-endmenu
+endif	# SND_MIPS
 
diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c
index ba38d6200099..e4282d93a1aa 100644
--- a/sound/oss/msnd.c
+++ b/sound/oss/msnd.c
@@ -20,8 +20,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd.c,v 1.17 1999/03/21 16:50:09 andrewtv Exp $
- *
  ********************************************************************/
 
 #include <linux/module.h>
diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h
index d0ca582c4583..61b3955481c5 100644
--- a/sound/oss/msnd.h
+++ b/sound/oss/msnd.h
@@ -24,8 +24,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd.h,v 1.36 1999/03/21 17:05:42 andrewtv Exp $
- *
  ********************************************************************/
 #ifndef __MSND_H
 #define __MSND_H
diff --git a/sound/oss/msnd_classic.h b/sound/oss/msnd_classic.h
index 7ffea5267f96..1a17dde2f650 100644
--- a/sound/oss/msnd_classic.h
+++ b/sound/oss/msnd_classic.h
@@ -24,8 +24,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  * 
- * $Id: msnd_classic.h,v 1.10 1999/03/21 17:36:09 andrewtv Exp $
- *
  ********************************************************************/
 #ifndef __MSND_CLASSIC_H
 #define __MSND_CLASSIC_H
diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c
index f1f49ebf752e..bf27e008f465 100644
--- a/sound/oss/msnd_pinnacle.c
+++ b/sound/oss/msnd_pinnacle.c
@@ -29,13 +29,8 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd_pinnacle.c,v 1.8 2000/12/30 00:33:21 sycamore Exp $
- *
  * 12-3-2000  Modified IO port validation  Steve Sycamore
  *
- *
- * $$$: msnd_pinnacle.c,v 1.75 1999/03/21 16:50:09 andrewtv $$$ $
- *
  ********************************************************************/
 
 #include <linux/kernel.h>
diff --git a/sound/oss/msnd_pinnacle.h b/sound/oss/msnd_pinnacle.h
index cce911487004..c18d66cbbe3f 100644
--- a/sound/oss/msnd_pinnacle.h
+++ b/sound/oss/msnd_pinnacle.h
@@ -24,8 +24,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd_pinnacle.h,v 1.11 1999/03/21 17:36:09 andrewtv Exp $
- *
  ********************************************************************/
 #ifndef __MSND_PINNACLE_H
 #define __MSND_PINNACLE_H
diff --git a/sound/parisc/Kconfig b/sound/parisc/Kconfig
index a5a7f9d75d05..9b61d95010f0 100644
--- a/sound/parisc/Kconfig
+++ b/sound/parisc/Kconfig
@@ -1,15 +1,20 @@
 # ALSA PA-RISC drivers
 
-menu "GSC devices"
-	depends on SND!=n && GSC
+menuconfig SND_GSC
+	bool "GSC sound devices"
+	depends on GSC
+	default y
+	help
+	  Support for GSC sound devices on PA-RISC architectures.
+
+if SND_GSC
 
 config SND_HARMONY
 	tristate "Harmony/Vivace sound chip"
-	depends on SND
 	select SND_PCM
 	help
 	  Say 'Y' or 'M' to include support for the Harmony/Vivace sound
 	  chip found in most GSC-based PA-RISC workstations.  It's frequently
 	  provided as part of the Lasi multi-function IC.
 
-endmenu
+endif	# SND_GSC
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 7e4742109572..8fe5dac39428 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,11 +1,16 @@
 # ALSA PCI drivers
 
-menu "PCI devices"
-	depends on SND!=n && PCI
+menuconfig SND_PCI
+	bool "PCI sound devices"
+	depends on PCI
+	default y
+	help
+	  Support for sound devices connected via the PCI bus.
+
+if SND_PCI
 
 config SND_AD1889
 	tristate "Analog Devices AD1889"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated AC97 sound
@@ -17,7 +22,6 @@ config SND_AD1889
 
 config SND_ALS300
 	tristate "Avance Logic ALS300/ALS300+"
-	depends on SND
 	select SND_PCM
 	select SND_AC97_CODEC
 	select SND_OPL3_LIB
@@ -29,7 +33,7 @@ config SND_ALS300
 
 config SND_ALS4000
 	tristate "Avance Logic ALS4000"
-	depends on SND && ISA_DMA_API
+	depends on ISA_DMA_API
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -43,7 +47,6 @@ config SND_ALS4000
 
 config SND_ALI5451
 	tristate "ALi M5451 PCI Audio Controller"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -57,7 +60,6 @@ config SND_ALI5451
 
 config SND_ATIIXP
 	tristate "ATI IXP AC97 Controller"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated AC97 sound
@@ -69,7 +71,6 @@ config SND_ATIIXP
 
 config SND_ATIIXP_MODEM
 	tristate "ATI IXP Modem"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated MC97 modem on
@@ -80,7 +81,6 @@ config SND_ATIIXP_MODEM
 
 config SND_AU8810
 	tristate "Aureal Advantage"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -95,7 +95,6 @@ config SND_AU8810
 
 config SND_AU8820
 	tristate "Aureal Vortex"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -109,7 +108,6 @@ config SND_AU8820
 
 config SND_AU8830
 	tristate "Aureal Vortex 2"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -124,7 +122,6 @@ config SND_AU8830
 
 config SND_AW2
 	tristate "Emagic Audiowerk 2"
-	depends on SND
 	help
 	  Say Y here to include support for Emagic Audiowerk 2 soundcards.
 
@@ -139,7 +136,7 @@ config SND_AW2
 
 config SND_AZT3328
 	tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)"
-	depends on SND && EXPERIMENTAL
+	depends on EXPERIMENTAL
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -152,7 +149,6 @@ config SND_AZT3328
 
 config SND_BT87X
 	tristate "Bt87x Audio Capture"
-	depends on SND
 	select SND_PCM
 	help
 	  If you want to record audio from TV cards based on
@@ -174,7 +170,6 @@ config SND_BT87X_OVERCLOCK
 
 config SND_CA0106
 	tristate "SB Audigy LS / Live 24bit"
-	depends on SND
 	select SND_AC97_CODEC
 	select SND_RAWMIDI
 	select SND_VMASTER
@@ -187,7 +182,6 @@ config SND_CA0106
 
 config SND_CMIPCI
 	tristate "C-Media 8338, 8738, 8768, 8770"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -201,13 +195,11 @@ config SND_CMIPCI
 
 config SND_OXYGEN_LIB
         tristate
-	depends on SND
 	select SND_PCM
 	select SND_MPU401_UART
 
 config SND_OXYGEN
 	tristate "C-Media 8788 (Oxygen)"
-	depends on SND
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
@@ -225,7 +217,6 @@ config SND_OXYGEN
 
 config SND_CS4281
 	tristate "Cirrus Logic (Sound Fusion) CS4281"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
@@ -237,7 +228,6 @@ config SND_CS4281
 
 config SND_CS46XX
 	tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
 	help
@@ -258,7 +248,7 @@ config SND_CS46XX_NEW_DSP
 
 config SND_CS5530
 	tristate "CS5530 Audio"
-	depends on SND && ISA_DMA_API
+	depends on ISA_DMA_API
 	select SND_SB16_DSP
 	help
 	  Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips.
@@ -268,7 +258,7 @@ config SND_CS5530
 
 config SND_CS5535AUDIO
 	tristate "CS5535/CS5536 Audio"
-	depends on SND && X86 && !X86_64
+	depends on X86 && !X86_64
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
@@ -286,7 +276,6 @@ config SND_CS5535AUDIO
 
 config SND_DARLA20
 	tristate "(Echoaudio) Darla20"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -297,7 +286,6 @@ config SND_DARLA20
 
 config SND_GINA20
 	tristate "(Echoaudio) Gina20"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -308,7 +296,6 @@ config SND_GINA20
 
 config SND_LAYLA20
 	tristate "(Echoaudio) Layla20"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -320,7 +307,6 @@ config SND_LAYLA20
 
 config SND_DARLA24
 	tristate "(Echoaudio) Darla24"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -331,7 +317,6 @@ config SND_DARLA24
 
 config SND_GINA24
 	tristate "(Echoaudio) Gina24"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -342,7 +327,6 @@ config SND_GINA24
 
 config SND_LAYLA24
 	tristate "(Echoaudio) Layla24"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -354,7 +338,6 @@ config SND_LAYLA24
 
 config SND_MONA
 	tristate "(Echoaudio) Mona"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -366,7 +349,6 @@ config SND_MONA
 
 config SND_MIA
 	tristate "(Echoaudio) Mia"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -378,7 +360,6 @@ config SND_MIA
 
 config SND_ECHO3G
 	tristate "(Echoaudio) 3G cards"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -390,7 +371,6 @@ config SND_ECHO3G
 
 config SND_INDIGO
 	tristate "(Echoaudio) Indigo"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -401,7 +381,6 @@ config SND_INDIGO
 
 config SND_INDIGOIO
 	tristate "(Echoaudio) Indigo IO"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -412,7 +391,6 @@ config SND_INDIGOIO
 
 config SND_INDIGODJ
 	tristate "(Echoaudio) Indigo DJ"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -423,7 +401,6 @@ config SND_INDIGODJ
 
 config SND_EMU10K1
 	tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
-	depends on SND
 	select FW_LOADER
 	select SND_HWDEP
 	select SND_RAWMIDI
@@ -441,7 +418,6 @@ config SND_EMU10K1
 
 config SND_EMU10K1X
 	tristate "Emu10k1X (Dell OEM Version)"
-	depends on SND
 	select SND_AC97_CODEC
 	select SND_RAWMIDI
 	help
@@ -453,7 +429,6 @@ config SND_EMU10K1X
 
 config SND_ENS1370
 	tristate "(Creative) Ensoniq AudioPCI 1370"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_PCM
 	help
@@ -464,7 +439,6 @@ config SND_ENS1370
 
 config SND_ENS1371
 	tristate "(Creative) Ensoniq AudioPCI 1371/1373"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
 	help
@@ -476,7 +450,6 @@ config SND_ENS1371
 
 config SND_ES1938
 	tristate "ESS ES1938/1946/1969 (Solo-1)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
@@ -489,7 +462,6 @@ config SND_ES1938
 
 config SND_ES1968
 	tristate "ESS ES1968/1978 (Maestro-1/2/2E)"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -501,7 +473,6 @@ config SND_ES1968
 
 config SND_FM801
 	tristate "ForteMedia FM801"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
@@ -528,7 +499,6 @@ config SND_FM801_TEA575X
 
 config SND_HDA_INTEL
 	tristate "Intel HD Audio"
-	depends on SND
 	select SND_PCM
 	select SND_VMASTER
 	help
@@ -637,7 +607,6 @@ config SND_HDA_POWER_SAVE_DEFAULT
 
 config SND_HDSP
 	tristate "RME Hammerfall DSP Audio"
-	depends on SND
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -650,7 +619,6 @@ config SND_HDSP
 
 config SND_HDSPM
 	tristate "RME Hammerfall DSP MADI"
-	depends on SND
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -663,7 +631,6 @@ config SND_HDSPM
 
 config SND_HIFIER
 	tristate "TempoTec HiFier Fantasia"
-	depends on SND
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for the MediaTek/TempoTec HiFier
@@ -674,7 +641,6 @@ config SND_HIFIER
 
 config SND_ICE1712
 	tristate "ICEnsemble ICE1712 (Envy24)"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -691,8 +657,7 @@ config SND_ICE1712
 
 config SND_ICE1724
 	tristate "ICE/VT1724/1720 (Envy24HT/PT)"
-	depends on SND
-	select SND_MPU401_UART
+	select SND_RAWMIDI
 	select SND_AC97_CODEC
 	select SND_VMASTER
 	help
@@ -709,7 +674,6 @@ config SND_ICE1724
 
 config SND_INTEL8X0
 	tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated AC97 sound
@@ -722,7 +686,6 @@ config SND_INTEL8X0
 
 config SND_INTEL8X0M
 	tristate "Intel/SiS/nVidia/AMD MC97 Modem"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated MC97 modem on
@@ -733,7 +696,6 @@ config SND_INTEL8X0M
 
 config SND_KORG1212
 	tristate "Korg 1212 IO"
-	depends on SND
 	select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL
 	select SND_PCM
 	help
@@ -753,7 +715,6 @@ config SND_KORG1212_FIRMWARE_IN_KERNEL
 
 config SND_MAESTRO3
 	tristate "ESS Allegro/Maestro3"
-	depends on SND
 	select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL
 	select SND_AC97_CODEC
 	help
@@ -774,7 +735,6 @@ config SND_MAESTRO3_FIRMWARE_IN_KERNEL
 
 config SND_MIXART
 	tristate "Digigram miXart"
-	depends on SND
 	select SND_HWDEP
 	select SND_PCM
 	help
@@ -786,7 +746,6 @@ config SND_MIXART
 
 config SND_NM256
 	tristate "NeoMagic NM256AV/ZX"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for NeoMagic NM256AV/ZX chips.
@@ -796,7 +755,6 @@ config SND_NM256
 
 config SND_PCXHR
 	tristate "Digigram PCXHR"
-	depends on SND
 	select SND_PCM
 	select SND_HWDEP
 	help
@@ -807,7 +765,6 @@ config SND_PCXHR
 
 config SND_RIPTIDE
 	tristate "Conexant Riptide"
-	depends on SND
 	select FW_LOADER
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -820,7 +777,6 @@ config SND_RIPTIDE
 
 config SND_RME32
 	tristate "RME Digi32, 32/8, 32 PRO"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y to include support for RME Digi32, Digi32 PRO and
@@ -832,7 +788,6 @@ config SND_RME32
 
 config SND_RME96
 	tristate "RME Digi96, 96/8, 96/8 PRO"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include support for RME Digi96, Digi96/8 and
@@ -843,7 +798,6 @@ config SND_RME96
 
 config SND_RME9652
 	tristate "RME Digi9652 (Hammerfall)"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include support for RME Hammerfall (RME
@@ -854,7 +808,7 @@ config SND_RME9652
 
 config SND_SIS7019
 	tristate "SiS 7019 Audio Accelerator"
-	depends on SND && X86 && !X86_64
+	depends on X86 && !X86_64
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the SiS 7019 Audio Accelerator.
@@ -864,7 +818,6 @@ config SND_SIS7019
 
 config SND_SONICVIBES
 	tristate "S3 SonicVibes"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
@@ -877,7 +830,6 @@ config SND_SONICVIBES
 
 config SND_TRIDENT
 	tristate "Trident 4D-Wave DX/NX; SiS 7018"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -889,7 +841,6 @@ config SND_TRIDENT
 
 config SND_VIA82XX
 	tristate "VIA 82C686A/B, 8233/8235 AC97 Controller"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -901,7 +852,6 @@ config SND_VIA82XX
 
 config SND_VIA82XX_MODEM
 	tristate "VIA 82C686A/B, 8233 based Modems"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated MC97 modem on
@@ -912,7 +862,6 @@ config SND_VIA82XX_MODEM
 
 config SND_VIRTUOSO
 	tristate "Asus Virtuoso 100/200 (Xonar)"
-	depends on SND
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
@@ -923,7 +872,6 @@ config SND_VIRTUOSO
 
 config SND_VX222
 	tristate "Digigram VX222"
-	depends on SND
 	select SND_VX_LIB
 	help
 	  Say Y here to include support for Digigram VX222 soundcards.
@@ -933,7 +881,6 @@ config SND_VX222
 
 config SND_YMFPCI
 	tristate "Yamaha YMF724/740/744/754"
-	depends on SND
 	select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -954,25 +901,4 @@ config SND_YMFPCI_FIRMWARE_IN_KERNEL
 	  for the YMFPCI driver.  If you choose N here, you need to
 	  install the firmware files from the alsa-firmware package.
 
-config SND_AC97_POWER_SAVE
-	bool "AC97 Power-Saving Mode"
-	depends on SND_AC97_CODEC && EXPERIMENTAL
-	default n
-	help
-	  Say Y here to enable the aggressive power-saving support of
-	  AC97 codecs.  In this mode, the power-mode is dynamically
-	  controlled at each open/close.
-
-	  The mode is activated by passing power_save=1 option to
-	  snd-ac97-codec driver.  You can toggle it dynamically over
-	  sysfs, too.
-
-config SND_AC97_POWER_SAVE_DEFAULT
-	int "Default time-out for AC97 power-save mode"
-	depends on SND_AC97_POWER_SAVE
-	default 0
-	help
-	  The default time-out value in seconds for AC97 automatic
-	  power-save mode.  0 means to disable the power-save mode.
-
-endmenu
+endif	# SND_PCI
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 2da89810ca10..f668457e9908 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -3332,8 +3332,66 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0),
 AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0),
 };
 
+static const char *slave_vols_vt1616[] = {
+	"Front Playback Volume",
+	"Surround Playback Volume",
+	"Center Playback Volume",
+	"LFE Playback Volume",
+	NULL
+};
+
+static const char *slave_sws_vt1616[] = {
+	"Front Playback Switch",
+	"Surround Playback Switch",
+	"Center Playback Switch",
+	"LFE Playback Switch",
+	NULL
+};
+
+/* find a mixer control element with the given name */
+static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
+						    const char *name)
+{
+	struct snd_ctl_elem_id id;
+	memset(&id, 0, sizeof(id));
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	strcpy(id.name, name);
+	return snd_ctl_find_id(ac97->bus->card, &id);
+}
+
+/* create a virtual master control and add slaves */
+int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
+			 const unsigned int *tlv, const char **slaves)
+{
+	struct snd_kcontrol *kctl;
+	const char **s;
+	int err;
+
+	kctl = snd_ctl_make_virtual_master(name, tlv);
+	if (!kctl)
+		return -ENOMEM;
+	err = snd_ctl_add(ac97->bus->card, kctl);
+	if (err < 0)
+		return err;
+
+	for (s = slaves; *s; s++) {
+		struct snd_kcontrol *sctl;
+
+		sctl = snd_ac97_find_mixer_ctl(ac97, *s);
+		if (!sctl) {
+			snd_printdd("Cannot find slave %s, skipped\n", *s);
+			continue;
+		}
+		err = snd_ctl_add_slave(kctl, sctl);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
 static int patch_vt1616_specific(struct snd_ac97 * ac97)
 {
+	struct snd_kcontrol *kctl;
 	int err;
 
 	if (snd_ac97_try_bit(ac97, 0x5a, 9))
@@ -3341,6 +3399,24 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97)
 			return err;
 	if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[1], ARRAY_SIZE(snd_ac97_controls_vt1616) - 1)) < 0)
 		return err;
+
+	/* There is already a misnamed master switch.  Rename it.  */
+	kctl = snd_ac97_find_mixer_ctl(ac97, "Master Playback Volume");
+	if (!kctl)
+		return -EINVAL;
+
+	snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback");
+
+	err = snd_ac97_add_vmaster(ac97, "Master Playback Volume",
+				   kctl->tlv.p, slave_vols_vt1616);
+	if (err < 0)
+		return err;
+
+	err = snd_ac97_add_vmaster(ac97, "Master Playback Switch",
+				   NULL, slave_sws_vt1616);
+	if (err < 0)
+		return err;
+
 	return 0;
 }
 
diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c
index bc212f41a38a..e291aa59742e 100644
--- a/sound/pci/au88x0/au88x0_game.c
+++ b/sound/pci/au88x0/au88x0_game.c
@@ -1,6 +1,4 @@
 /*
- * $Id: au88x0_game.c,v 1.9 2003/09/22 03:51:28 mjander Exp $
- *
  *  Manuel Jander.
  *
  *  Based on the work of:
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 5f63af6b88a2..b832333c3023 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1,6 +1,6 @@
 /*
  *  azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168).
- *  Copyright (C) 2002, 2005, 2006, 2007 by Andreas Mohr <andi AT lisas.de>
+ *  Copyright (C) 2002, 2005 - 2008 by Andreas Mohr <andi AT lisas.de>
  *
  *  Framework borrowed from Bart Hartgers's als4000.c.
  *  Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801),
@@ -35,9 +35,20 @@
  *  (3 weeks' worth of evenings filled with driver work).
  *  (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros)
  *
+ *  It is quite likely that the AZF3328 chip is the PCI cousin of the
+ *  AZF3318 ("azt1020 pnp", "MM Pro 16") ISA chip, given very similar specs.
+ *
  *  The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name
- *  for compatibility reasons) has the following features:
+ *  for compatibility reasons) from Azfin (joint-venture of Aztech and Fincitec,
+ *  Fincitec acquired by National Semiconductor in 2002, together with the
+ *  Fincitec-related company ARSmikro) has the following features:
  *
+ *  - compatibility & compliance:
+ *    - Microsoft PC 97 ("PC 97 Hardware Design Guide",
+ *                       http://www.microsoft.com/whdc/archive/pcguides.mspx)
+ *    - Microsoft PC 98 Baseline Audio
+ *    - MPU401 UART
+ *    - Sound Blaster Emulation (DOS Box)
  *  - builtin AC97 conformant codec (SNR over 80dB)
  *    Note that "conformant" != "compliant"!! this chip's mixer register layout
  *    *differs* from the standard AC97 layout:
@@ -48,21 +59,28 @@
  *    addresses illegally. So far unfortunately it looks like the very flexible
  *    ALSA AC97 support is still not enough to easily compensate for such a
  *    grave layout violation despite all tweaks and quirks mechanisms it offers.
- *  - builtin genuine OPL3
+ *  - builtin genuine OPL3 - verified to work fine, 20080506
  *  - full duplex 16bit playback/record at independent sampling rate
- *  - MPU401 (+ legacy address support) FIXME: how to enable legacy addr??
+ *  - MPU401 (+ legacy address support, claimed by one official spec sheet)
+ *    FIXME: how to enable legacy addr??
  *  - game port (legacy address support)
- *  - builtin 3D enhancement (said to be YAMAHA Ymersion)
  *  - builtin DirectInput support, helps reduce CPU overhead (interrupt-driven
- *    features supported)
+ *    features supported). - See common term "Digital Enhanced Game Port"...
+ *    (probably DirectInput 3.0 spec - confirm)
+ *  - builtin 3D enhancement (said to be YAMAHA Ymersion)
  *  - built-in General DirectX timer having a 20 bits counter
  *    with 1us resolution (see below!)
- *  - I2S serial port for external DAC
+ *  - I2S serial output port for external DAC
  *  - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI
  *  - supports hardware volume control
  *  - single chip low cost solution (128 pin QFP)
  *  - supports programmable Sub-vendor and Sub-system ID
  *    required for Microsoft's logo compliance (FIXME: where?)
+ *    At least the Trident 4D Wave DX has one bit somewhere
+ *    to enable writes to PCI subsystem VID registers, that should be it.
+ *    This might easily be in extended PCI reg space, since PCI168 also has
+ *    some custom data starting at 0x80. What kind of config settings
+ *    are located in our extended PCI space anyway??
  *  - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms
  *
  *  Note that this driver now is actually *better* than the Windows driver,
@@ -74,6 +92,24 @@
  *  - "timidity -iAv -B2,8 -Os -EFreverb=0"
  *  - "pmidi -p 128:0 jazz.mid"
  *
+ *  OPL3 hardware playback testing, try something like:
+ *  cat /proc/asound/hwdep
+ *  and
+ *  aconnect -o
+ *  Then use
+ *  sbiload -Dhw:x,y --opl3 /usr/share/sounds/opl3/std.o3 ......./drums.o3
+ *  where x,y is the xx-yy number as given in hwdep.
+ *  Then try
+ *  pmidi -p a:b jazz.mid
+ *  where a:b is the client number plus 0 usually, as given by aconnect above.
+ *  Oh, and make sure to unmute the FM mixer control (doh!)
+ *  NOTE: power use during OPL3 playback is _VERY_ high (70W --> 90W!)
+ *  despite no CPU activity, possibly due to hindering ACPI idling somehow.
+ *  Shouldn't be a problem of the AZF3328 chip itself, I'd hope.
+ *  Higher PCM / FM mixer levels seem to conflict (causes crackling),
+ *  at least sometimes.   Maybe even use with hardware sequencer timer above :)
+ *  adplay/adplug-utils might soon offer hardware-based OPL3 playback, too.
+ *
  *  Certain PCI versions of this card are susceptible to DMA traffic underruns
  *  in some systems (resulting in sound crackling/clicking/popping),
  *  probably because they don't have a DMA FIFO buffer or so.
@@ -87,6 +123,8 @@
  *  better than a VIA, yet ironically I still get crackling, like many other
  *  people with the same chipset.
  *  Possible remedies:
+ *  - use speaker (amplifier) output instead of headphone output
+ *    (in case crackling is due to overloaded output clipping)
  *  - plug card into a different PCI slot, preferrably one that isn't shared
  *    too much (this helps a lot, but not completely!)
  *  - get rid of PCI VGA card, use AGP instead
@@ -94,18 +132,23 @@
  *  - fiddle with PCI latency settings (setpci -v -s BUSID latency_timer=XX)
  *    Not too helpful.
  *  - Disable ACPI/power management/"Auto Detect RAM/PCI Clk" in BIOS
- * 
+ *
  * BUGS
- *  - full-duplex might *still* be problematic, not fully tested recently
+ *  - full-duplex might *still* be problematic, however a recent test was fine
  *  - (non-bug) "Bass/Treble or 3D settings don't work" - they do get evaluated
  *    if you set PCM output switch to "pre 3D" instead of "post 3D".
  *    If this can't be set, then get a mixer application that Isn't Stupid (tm)
  *    (e.g. kmix, gamix) - unfortunately several are!!
- * 
+ *  - locking is not entirely clean, especially the audio stream activity
+ *    ints --> may be racy
+ *  - an _unconnected_ secondary joystick at the gameport will be reported
+ *    to be "active" (floating values, not precisely -1) due to the way we need
+ *    to read the Digital Enhanced Game Port. Not sure whether it is fixable.
+ *
  * TODO
  *  - test MPU401 MIDI playback etc.
- *  - add some power micro-management (disable various units of the card
- *    as long as they're unused). However this requires I/O ports which I
+ *  - add more power micro-management (disable various units of the card
+ *    as long as they're unused). However this requires more I/O ports which I
  *    haven't figured out yet and which thus might not even exist...
  *    The standard suspend/resume functionality could probably make use of
  *    some improvement, too...
@@ -113,6 +156,7 @@
  *  - figure out some cleverly evil scheme to possibly make ALSA AC97 code
  *    fully accept our quite incompatible ""AC97"" mixer and thus save some
  *    code (but I'm not too optimistic that doing this is possible at all)
+ *  - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport.
  */
 
 #include <asm/io.h>
@@ -138,7 +182,7 @@ MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 
 #if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
-#define SUPPORT_JOYSTICK 1
+#define SUPPORT_GAMEPORT 1
 #endif
 
 #define DEBUG_MISC	0
@@ -147,13 +191,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 #define DEBUG_PLAY_REC	0
 #define DEBUG_IO	0
 #define DEBUG_TIMER	0
+#define DEBUG_GAME	0
 #define MIXER_TESTING	0
 
 #if DEBUG_MISC
 #define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args)
 #else
 #define snd_azf3328_dbgmisc(format, args...)
-#endif		
+#endif
 
 #if DEBUG_CALLS
 #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args)
@@ -163,25 +208,31 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 #define snd_azf3328_dbgcalls(format, args...)
 #define snd_azf3328_dbgcallenter()
 #define snd_azf3328_dbgcallleave()
-#endif		
+#endif
 
 #if DEBUG_MIXER
 #define snd_azf3328_dbgmixer(format, args...) printk(format, ##args)
 #else
 #define snd_azf3328_dbgmixer(format, args...)
-#endif		
+#endif
 
 #if DEBUG_PLAY_REC
 #define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args)
 #else
 #define snd_azf3328_dbgplay(format, args...)
-#endif		
+#endif
 
 #if DEBUG_MISC
 #define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args)
 #else
 #define snd_azf3328_dbgtimer(format, args...)
-#endif		
+#endif
+
+#if DEBUG_GAME
+#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args)
+#else
+#define snd_azf3328_dbggame(format, args...)
+#endif
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 module_param_array(index, int, NULL, 0444);
@@ -195,39 +246,44 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card *
 module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard.");
 
-#ifdef SUPPORT_JOYSTICK
-static int joystick[SNDRV_CARDS];
-module_param_array(joystick, bool, NULL, 0444);
-MODULE_PARM_DESC(joystick, "Enable joystick for AZF3328 soundcard.");
-#endif
-
 static int seqtimer_scaling = 128;
 module_param(seqtimer_scaling, int, 0444);
 MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128.");
 
+struct snd_azf3328_audio_stream {
+	struct snd_pcm_substream *substream;
+	int enabled;
+	int running;
+	unsigned long portbase;
+};
+
+enum snd_azf3328_stream_index {
+  AZF_PLAYBACK = 0,
+  AZF_CAPTURE = 1,
+};
+
 struct snd_azf3328 {
 	/* often-used fields towards beginning, then grouped */
-	unsigned long codec_port;
-	unsigned long io2_port;
-	unsigned long mpu_port;
-	unsigned long synth_port;
-	unsigned long mixer_port;
+
+	unsigned long codec_io; /* usually 0xb000, size 128 */
+	unsigned long game_io;  /* usually 0xb400, size 8 */
+	unsigned long mpu_io;   /* usually 0xb800, size 4 */
+	unsigned long opl3_io; /* usually 0xbc00, size 8 */
+	unsigned long mixer_io; /* usually 0xc000, size 64 */
 
 	spinlock_t reg_lock;
 
 	struct snd_timer *timer;
-	
+
 	struct snd_pcm *pcm;
-	struct snd_pcm_substream *playback_substream;
-	struct snd_pcm_substream *capture_substream;
-	unsigned int is_playing;
-	unsigned int is_recording;
+	struct snd_azf3328_audio_stream audio_stream[2];
 
 	struct snd_card *card;
 	struct snd_rawmidi *rmidi;
 
-#ifdef SUPPORT_JOYSTICK
+#ifdef SUPPORT_GAMEPORT
 	struct gameport *gameport;
+	int axes[4];
 #endif
 
 	struct pci_dev *pci;
@@ -236,10 +292,10 @@ struct snd_azf3328 {
 #ifdef CONFIG_PM
 	/* register value containers for power management
 	 * Note: not always full I/O range preserved (just like Win driver!) */
-	u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2];
-	u16 saved_regs_io2   [AZF_IO_SIZE_IO2_PM / 2];
-	u16 saved_regs_mpu   [AZF_IO_SIZE_MPU_PM / 2];
-	u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2];
+	u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2];
+	u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2];
+	u16 saved_regs_mpu  [AZF_IO_SIZE_MPU_PM / 2];
+	u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2];
 	u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2];
 #endif
 };
@@ -252,126 +308,181 @@ static const struct pci_device_id snd_azf3328_ids[] = {
 
 MODULE_DEVICE_TABLE(pci, snd_azf3328_ids);
 
+
+static int
+snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set)
+{
+	u8 prev = inb(reg), new;
+
+	new = (do_set) ? (prev|mask) : (prev & ~mask);
+	/* we need to always write the new value no matter whether it differs
+	 * or not, since some register bits don't indicate their setting */
+	outb(new, reg);
+	if (new != prev)
+		return 1;
+
+	return 0;
+}
+
+static int
+snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set)
+{
+	u16 prev = inw(reg), new;
+
+	new = (do_set) ? (prev|mask) : (prev & ~mask);
+	/* we need to always write the new value no matter whether it differs
+	 * or not, since some register bits don't indicate their setting */
+	outw(new, reg);
+	if (new != prev)
+		return 1;
+
+	return 0;
+}
+
 static inline void
-snd_azf3328_codec_outb(const struct snd_azf3328 *chip, int reg, u8 value)
+snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value)
 {
-	outb(value, chip->codec_port + reg);
+	outb(value, chip->codec_io + reg);
 }
 
 static inline u8
-snd_azf3328_codec_inb(const struct snd_azf3328 *chip, int reg)
+snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg)
 {
-	return inb(chip->codec_port + reg);
+	return inb(chip->codec_io + reg);
 }
 
 static inline void
-snd_azf3328_codec_outw(const struct snd_azf3328 *chip, int reg, u16 value)
+snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
 {
-	outw(value, chip->codec_port + reg);
+	outw(value, chip->codec_io + reg);
 }
 
 static inline u16
-snd_azf3328_codec_inw(const struct snd_azf3328 *chip, int reg)
+snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg)
+{
+	return inw(chip->codec_io + reg);
+}
+
+static inline void
+snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value)
+{
+	outl(value, chip->codec_io + reg);
+}
+
+static inline u32
+snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg)
 {
-	return inw(chip->codec_port + reg);
+	return inl(chip->codec_io + reg);
 }
 
 static inline void
-snd_azf3328_codec_outl(const struct snd_azf3328 *chip, int reg, u32 value)
+snd_azf3328_game_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value)
 {
-	outl(value, chip->codec_port + reg);
+	outb(value, chip->game_io + reg);
 }
 
 static inline void
-snd_azf3328_io2_outb(const struct snd_azf3328 *chip, int reg, u8 value)
+snd_azf3328_game_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
 {
-	outb(value, chip->io2_port + reg);
+	outw(value, chip->game_io + reg);
 }
 
 static inline u8
-snd_azf3328_io2_inb(const struct snd_azf3328 *chip, int reg)
+snd_azf3328_game_inb(const struct snd_azf3328 *chip, unsigned reg)
+{
+	return inb(chip->game_io + reg);
+}
+
+static inline u16
+snd_azf3328_game_inw(const struct snd_azf3328 *chip, unsigned reg)
 {
-	return inb(chip->io2_port + reg);
+	return inw(chip->game_io + reg);
 }
 
 static inline void
-snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, int reg, u16 value)
+snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
 {
-	outw(value, chip->mixer_port + reg);
+	outw(value, chip->mixer_io + reg);
 }
 
 static inline u16
-snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, int reg)
+snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg)
 {
-	return inw(chip->mixer_port + reg);
+	return inw(chip->mixer_io + reg);
 }
 
-static void
-snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, int reg, int do_mute)
+#define AZF_MUTE_BIT 0x80
+
+static int
+snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip,
+			   unsigned reg, int do_mute
+)
 {
-	unsigned long portbase = chip->mixer_port + reg + 1;
-	unsigned char oldval;
+	unsigned long portbase = chip->mixer_io + reg + 1;
+	int updated;
 
 	/* the mute bit is on the *second* (i.e. right) register of a
 	 * left/right channel setting */
-	oldval = inb(portbase);
-	if (do_mute)
-		oldval |= 0x80;
-	else
-		oldval &= ~0x80;
-	outb(oldval, portbase);
+	updated = snd_azf3328_io_reg_setb(portbase, AZF_MUTE_BIT, do_mute);
+
+	/* indicate whether it was muted before */
+	return (do_mute) ? !updated : updated;
 }
 
 static void
-snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg, unsigned char dst_vol_left, unsigned char dst_vol_right, int chan_sel, int delay)
+snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip,
+					 unsigned reg,
+					 unsigned char dst_vol_left,
+					 unsigned char dst_vol_right,
+					 int chan_sel, int delay
+)
 {
-	unsigned long portbase = chip->mixer_port + reg;
+	unsigned long portbase = chip->mixer_io + reg;
 	unsigned char curr_vol_left = 0, curr_vol_right = 0;
-	int left_done = 0, right_done = 0;
-	
+	int left_change = 0, right_change = 0;
+
 	snd_azf3328_dbgcallenter();
-	if (chan_sel & SET_CHAN_LEFT)
+
+	if (chan_sel & SET_CHAN_LEFT) {
 		curr_vol_left  = inb(portbase + 1);
-	else
-		left_done = 1;
-	if (chan_sel & SET_CHAN_RIGHT)
+
+		/* take care of muting flag contained in left channel */
+		if (curr_vol_left & AZF_MUTE_BIT)
+			dst_vol_left |= AZF_MUTE_BIT;
+		else
+			dst_vol_left &= ~AZF_MUTE_BIT;
+
+		left_change = (curr_vol_left > dst_vol_left) ? -1 : 1;
+	}
+
+	if (chan_sel & SET_CHAN_RIGHT) {
 		curr_vol_right = inb(portbase + 0);
-	else
-		right_done = 1;
-	
-	/* take care of muting flag (0x80) contained in left channel */
-	if (curr_vol_left & 0x80)
-		dst_vol_left |= 0x80;
-	else
-		dst_vol_left &= ~0x80;
+
+		right_change = (curr_vol_right > dst_vol_right) ? -1 : 1;
+	}
 
 	do {
-		if (!left_done) {
-			if (curr_vol_left > dst_vol_left)
-				curr_vol_left--;
-			else
-			if (curr_vol_left < dst_vol_left)
-				curr_vol_left++;
-			else
-			    left_done = 1;
-			outb(curr_vol_left, portbase + 1);
+		if (left_change) {
+			if (curr_vol_left != dst_vol_left) {
+				curr_vol_left += left_change;
+				outb(curr_vol_left, portbase + 1);
+			} else
+			    left_change = 0;
 		}
-		if (!right_done) {
-			if (curr_vol_right > dst_vol_right)
-				curr_vol_right--;
-			else
-			if (curr_vol_right < dst_vol_right)
-				curr_vol_right++;
-			else
-			    right_done = 1;
+		if (right_change) {
+			if (curr_vol_right != dst_vol_right) {
+				curr_vol_right += right_change;
+
 			/* during volume change, the right channel is crackling
 			 * somewhat more than the left channel, unfortunately.
 			 * This seems to be a hardware issue. */
-			outb(curr_vol_right, portbase + 0);
+				outb(curr_vol_right, portbase + 0);
+			} else
+			    right_change = 0;
 		}
 		if (delay)
 			mdelay(delay);
-	} while ((!left_done) || (!right_done));
+	} while ((left_change) || (right_change));
 	snd_azf3328_dbgcallleave();
 }
 
@@ -379,7 +490,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg
  * general mixer element
  */
 struct azf3328_mixer_reg {
-	unsigned int reg;
+	unsigned reg;
 	unsigned int lchan_shift, rchan_shift;
 	unsigned int mask;
 	unsigned int invert: 1;
@@ -544,13 +655,14 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
 		"Mix", "Mic"
 	};
 	static const char * const texts3[] = {
-                "Mic", "CD", "Video", "Aux",
+		"Mic", "CD", "Video", "Aux",
 		"Line", "Mix", "Mix Mono", "Phone"
         };
 	static const char * const texts4[] = {
 		"pre 3D", "post 3D"
         };
 	struct azf3328_mixer_reg reg;
+	const char *p = NULL;
 
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
         uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
@@ -561,18 +673,20 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
 	if (reg.reg == IDX_MIXER_ADVCTL2) {
 		switch(reg.lchan_shift) {
 		case 8: /* modem out sel */
-			strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]);
+			p = texts1[uinfo->value.enumerated.item];
 			break;
 		case 9: /* mono sel source */
-			strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]);
+			p = texts2[uinfo->value.enumerated.item];
 			break;
 		case 15: /* PCM Out Path */
-			strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]);
+			p = texts4[uinfo->value.enumerated.item];
 			break;
 		}
 	} else
-        	strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item]
-);
+	if (reg.reg == IDX_MIXER_REC_SELECT)
+		p = texts3[uinfo->value.enumerated.item];
+
+	strcpy(uinfo->value.enumerated.name, p);
         return 0;
 }
 
@@ -583,7 +697,7 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol,
         struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
 	struct azf3328_mixer_reg reg;
         unsigned short val;
-        
+
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
 	val = snd_azf3328_mixer_inw(chip, reg.reg);
 	if (reg.reg == IDX_MIXER_REC_SELECT) {
@@ -605,7 +719,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol,
         struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
 	struct azf3328_mixer_reg reg;
 	unsigned int oreg, nreg, val;
-        
+
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
 	oreg = snd_azf3328_mixer_inw(chip, reg.reg);
 	val = oreg;
@@ -717,15 +831,16 @@ snd_azf3328_mixer_new(struct snd_azf3328 *chip)
 	snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000);
 
 	/* mute and zero volume channels */
-	for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); idx++) {
+	for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); ++idx) {
 		snd_azf3328_mixer_outw(chip,
 			snd_azf3328_init_values[idx][0],
 			snd_azf3328_init_values[idx][1]);
 	}
-	
+
 	/* add mixer controls */
 	sw = snd_azf3328_mixer_controls;
-	for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); idx++, sw++) {
+	for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls);
+			++idx, ++sw) {
 		if ((err = snd_ctl_add(chip->card, snd_ctl_new1(sw, chip))) < 0)
 			return err;
 	}
@@ -757,8 +872,8 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream)
 }
 
 static void
-snd_azf3328_setfmt(struct snd_azf3328 *chip,
-			       unsigned int reg,
+snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
+			       unsigned reg,
 			       unsigned int bitrate,
 			       unsigned int format_width,
 			       unsigned int channels
@@ -769,24 +884,25 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip,
 
 	snd_azf3328_dbgcallenter();
 	switch (bitrate) {
-	case  4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
-	case  4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
-	case  5512: val |= SOUNDFORMAT_FREQ_5510; break; /* the AZF3328 names it "5510" for some strange reason */
-	case  6620: val |= SOUNDFORMAT_FREQ_6620; break;
-	case  8000: val |= SOUNDFORMAT_FREQ_8000; break;
-	case  9600: val |= SOUNDFORMAT_FREQ_9600; break;
-	case 11025: val |= SOUNDFORMAT_FREQ_11025; break;
-	case 13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
-	case 16000: val |= SOUNDFORMAT_FREQ_16000; break;
-	case 22050: val |= SOUNDFORMAT_FREQ_22050; break;
-	case 32000: val |= SOUNDFORMAT_FREQ_32000; break;
-	case 44100: val |= SOUNDFORMAT_FREQ_44100; break;
-	case 48000: val |= SOUNDFORMAT_FREQ_48000; break;
-	case 66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
+	case AZF_FREQ_4000:  val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+	case AZF_FREQ_4800:  val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+	case AZF_FREQ_5512:
+		/* the AZF3328 names it "5510" for some strange reason */
+			     val |= SOUNDFORMAT_FREQ_5510; break;
+	case AZF_FREQ_6620:  val |= SOUNDFORMAT_FREQ_6620; break;
+	case AZF_FREQ_8000:  val |= SOUNDFORMAT_FREQ_8000; break;
+	case AZF_FREQ_9600:  val |= SOUNDFORMAT_FREQ_9600; break;
+	case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break;
+	case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+	case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break;
+	case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break;
+	case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break;
 	default:
 		snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
-		val |= SOUNDFORMAT_FREQ_44100;
-		break;
+		/* fall-through */
+	case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break;
+	case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break;
+	case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
 	}
 	/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
 	/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
@@ -805,10 +921,10 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip,
 		val |= SOUNDFORMAT_FLAG_16BIT;
 
 	spin_lock_irqsave(&chip->reg_lock, flags);
-	
+
 	/* set bitrate/format */
 	snd_azf3328_codec_outw(chip, reg, val);
-	
+
 	/* changing the bitrate/format settings switches off the
 	 * audio output with an annoying click in case of 8/16bit format change
 	 * (maybe shutting down DAC/ADC?), thus immediately
@@ -830,31 +946,81 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip,
 	snd_azf3328_dbgcallleave();
 }
 
+static inline void
+snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip,
+			    unsigned reg
+)
+{
+	/* choose lowest frequency for low power consumption.
+	 * While this will cause louder noise due to rather coarse frequency,
+	 * it should never matter since output should always
+	 * get disabled properly when idle anyway. */
+	snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1);
+}
+
+static inline void
+snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable)
+{
+	/* no idea what exactly is being done here, but I strongly assume it's
+	 * PM related */
+	snd_azf3328_io_reg_setw(
+		chip->codec_io+IDX_IO_6AH,
+		IO_6A_PAUSE_PLAYBACK_BIT8,
+		!enable
+	);
+}
+
+static void
+snd_azf3328_codec_activity(struct snd_azf3328 *chip,
+				enum snd_azf3328_stream_index stream_type,
+				int enable
+)
+{
+	int need_change = (chip->audio_stream[stream_type].running != enable);
+
+	snd_azf3328_dbgplay(
+		"codec_activity: type %d, enable %d, need_change %d\n",
+				stream_type, enable, need_change
+	);
+	if (need_change) {
+		enum snd_azf3328_stream_index other =
+			(stream_type == AZF_PLAYBACK) ?
+				AZF_CAPTURE : AZF_PLAYBACK;
+		/* small check to prevent shutting down the other party
+		 * in case it's active */
+		if ((enable) || !(chip->audio_stream[other].running))
+			snd_azf3328_codec_enable(chip, enable);
+
+		/* ...and adjust clock, too
+		 * (reduce noise and power consumption) */
+		if (!enable)
+			snd_azf3328_codec_setfmt_lowpower(
+				chip,
+				chip->audio_stream[stream_type].portbase
+					+ IDX_IO_PLAY_SOUNDFORMAT
+			);
+	}
+	chip->audio_stream[stream_type].running = enable;
+}
+
 static void
 snd_azf3328_setdmaa(struct snd_azf3328 *chip,
 				long unsigned int addr,
                                 unsigned int count,
                                 unsigned int size,
-				int do_recording)
+				enum snd_azf3328_stream_index stream_type
+)
 {
-	unsigned long flags, portbase;
-	unsigned int is_running;
-
 	snd_azf3328_dbgcallenter();
-	if (do_recording) {
-		/* access capture registers, i.e. skip playback reg section */
-		portbase = chip->codec_port + 0x20;
-		is_running = chip->is_recording;
-	} else {
-		/* access the playback register section */
-		portbase = chip->codec_port + 0x00;
-		is_running = chip->is_playing;
-	}
+	if (!chip->audio_stream[stream_type].running) {
+		/* AZF3328 uses a two buffer pointer DMA playback approach */
+
+		unsigned long flags, portbase, addr_area2;
+
+		/* width 32bit (prevent overflow): */
+		unsigned long count_areas, count_tmp;
 
-	/* AZF3328 uses a two buffer pointer DMA playback approach */
-	if (!is_running) {
-		unsigned long addr_area2;
-		unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */
+		portbase = chip->audio_stream[stream_type].portbase;
 		count_areas = size/2;
 		addr_area2 = addr+count_areas;
 		count_areas--; /* max. index */
@@ -884,11 +1050,11 @@ snd_azf3328_playback_prepare(struct snd_pcm_substream *substream)
 
 	snd_azf3328_dbgcallenter();
 #if 0
-	snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
+	snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
 		runtime->rate,
 		snd_pcm_format_width(runtime->format),
 		runtime->channels);
-	snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 0);
+	snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK);
 #endif
 	snd_azf3328_dbgcallleave();
 	return 0;
@@ -906,11 +1072,11 @@ snd_azf3328_capture_prepare(struct snd_pcm_substream *substream)
 
 	snd_azf3328_dbgcallenter();
 #if 0
-	snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
+	snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
 		runtime->rate,
 		snd_pcm_format_width(runtime->format),
 		runtime->channels);
-	snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 1);
+	snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE);
 #endif
 	snd_azf3328_dbgcallleave();
 	return 0;
@@ -923,6 +1089,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int result = 0;
 	unsigned int status1;
+	int previously_muted;
 
 	snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd);
 
@@ -930,20 +1097,23 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 		snd_azf3328_dbgplay("START PLAYBACK\n");
 
-		/* mute WaveOut */
-		snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+		/* mute WaveOut (avoid clicking during setup) */
+		previously_muted =
+			snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
 
-		snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
+		snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
 			runtime->rate,
 			snd_pcm_format_width(runtime->format),
 			runtime->channels);
 
 		spin_lock(&chip->reg_lock);
-		/* stop playback */
+		/* first, remember current value: */
 		status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS);
+
+		/* stop playback */
 		status1 &= ~DMA_RESUME;
 		snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
-	    
+
 		/* FIXME: clear interrupts or what??? */
 		snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff);
 		spin_unlock(&chip->reg_lock);
@@ -951,7 +1121,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		snd_azf3328_setdmaa(chip, runtime->dma_addr,
 			snd_pcm_lib_period_bytes(substream),
 			snd_pcm_lib_buffer_bytes(substream),
-			0);
+			AZF_PLAYBACK);
 
 		spin_lock(&chip->reg_lock);
 #ifdef WIN9X
@@ -978,30 +1148,35 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 			DMA_SOMETHING_ELSE);
 #endif
 		spin_unlock(&chip->reg_lock);
+		snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1);
 
 		/* now unmute WaveOut */
-		snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
+		if (!previously_muted)
+			snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
 
-		chip->is_playing = 1;
 		snd_azf3328_dbgplay("STARTED PLAYBACK\n");
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
 		snd_azf3328_dbgplay("RESUME PLAYBACK\n");
 		/* resume playback if we were active */
-		if (chip->is_playing)
+		spin_lock(&chip->reg_lock);
+		if (chip->audio_stream[AZF_PLAYBACK].running)
 			snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
 				snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME);
+		spin_unlock(&chip->reg_lock);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		snd_azf3328_dbgplay("STOP PLAYBACK\n");
 
-		/* mute WaveOut */
-		snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+		/* mute WaveOut (avoid clicking during setup) */
+		previously_muted =
+			snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
 
 		spin_lock(&chip->reg_lock);
-		/* stop playback */
+		/* first, remember current value: */
 		status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS);
 
+		/* stop playback */
 		status1 &= ~DMA_RESUME;
 		snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
 
@@ -1013,10 +1188,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		status1 &= ~DMA_PLAY_SOMETHING1;
 		snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
 		spin_unlock(&chip->reg_lock);
-	    
+		snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0);
+
 		/* now unmute WaveOut */
-		snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
-		chip->is_playing = 0;
+		if (!previously_muted)
+			snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
+
 		snd_azf3328_dbgplay("STOPPED PLAYBACK\n");
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
@@ -1035,7 +1212,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		printk(KERN_ERR "FIXME: unknown trigger mode!\n");
                 return -EINVAL;
 	}
-	
+
 	snd_azf3328_dbgcallleave();
 	return result;
 }
@@ -1057,17 +1234,19 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 
 		snd_azf3328_dbgplay("START CAPTURE\n");
 
-		snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
+		snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
 			runtime->rate,
 			snd_pcm_format_width(runtime->format),
 			runtime->channels);
 
 		spin_lock(&chip->reg_lock);
-		/* stop recording */
+		/* first, remember current value: */
 		status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS);
+
+		/* stop recording */
 		status1 &= ~DMA_RESUME;
 		snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
-	    
+
 		/* FIXME: clear interrupts or what??? */
 		snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff);
 		spin_unlock(&chip->reg_lock);
@@ -1075,7 +1254,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 		snd_azf3328_setdmaa(chip, runtime->dma_addr,
 			snd_pcm_lib_period_bytes(substream),
 			snd_pcm_lib_buffer_bytes(substream),
-			1);
+			AZF_CAPTURE);
 
 		spin_lock(&chip->reg_lock);
 #ifdef WIN9X
@@ -1102,24 +1281,27 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 			DMA_SOMETHING_ELSE);
 #endif
 		spin_unlock(&chip->reg_lock);
+		snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1);
 
-		chip->is_recording = 1;
 		snd_azf3328_dbgplay("STARTED CAPTURE\n");
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
 		snd_azf3328_dbgplay("RESUME CAPTURE\n");
 		/* resume recording if we were active */
-		if (chip->is_recording)
+		spin_lock(&chip->reg_lock);
+		if (chip->audio_stream[AZF_CAPTURE].running)
 			snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
 				snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME);
+		spin_unlock(&chip->reg_lock);
 		break;
         case SNDRV_PCM_TRIGGER_STOP:
 		snd_azf3328_dbgplay("STOP CAPTURE\n");
 
 		spin_lock(&chip->reg_lock);
-		/* stop recording */
+		/* first, remember current value: */
 		status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS);
 
+		/* stop recording */
 		status1 &= ~DMA_RESUME;
 		snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
 
@@ -1129,8 +1311,8 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 		status1 &= ~DMA_PLAY_SOMETHING1;
 		snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
 		spin_unlock(&chip->reg_lock);
-	    
-		chip->is_recording = 0;
+		snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0);
+
 		snd_azf3328_dbgplay("STOPPED CAPTURE\n");
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
@@ -1149,7 +1331,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 		printk(KERN_ERR "FIXME: unknown trigger mode!\n");
                 return -EINVAL;
 	}
-	
+
 	snd_azf3328_dbgcallleave();
 	return result;
 }
@@ -1162,11 +1344,11 @@ snd_azf3328_playback_pointer(struct snd_pcm_substream *substream)
 	snd_pcm_uframes_t frmres;
 
 #ifdef QUERY_HARDWARE
-	bufptr = inl(chip->codec_port+IDX_IO_PLAY_DMA_START_1);
+	bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1);
 #else
 	bufptr = substream->runtime->dma_addr;
 #endif
-	result = inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS);
+	result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS);
 
 	/* calculate offset */
 	result -= bufptr;
@@ -1183,11 +1365,11 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream)
 	snd_pcm_uframes_t frmres;
 
 #ifdef QUERY_HARDWARE
-	bufptr = inl(chip->codec_port+IDX_IO_REC_DMA_START_1);
+	bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1);
 #else
 	bufptr = substream->runtime->dma_addr;
 #endif
-	result = inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS);
+	result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS);
 
 	/* calculate offset */
 	result -= bufptr;
@@ -1196,27 +1378,233 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream)
 	return frmres;
 }
 
+/******************************************************************/
+
+#ifdef SUPPORT_GAMEPORT
+static inline void
+snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable)
+{
+	snd_azf3328_io_reg_setb(
+		chip->game_io+IDX_GAME_HWCONFIG,
+		GAME_HWCFG_IRQ_ENABLE,
+		enable
+	);
+}
+
+static inline void
+snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable)
+{
+	snd_azf3328_io_reg_setb(
+		chip->game_io+IDX_GAME_HWCONFIG,
+		GAME_HWCFG_LEGACY_ADDRESS_ENABLE,
+		enable
+	);
+}
+
+static inline void
+snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable)
+{
+	snd_azf3328_io_reg_setw(
+		chip->codec_io+IDX_IO_6AH,
+		IO_6A_SOMETHING2_GAMEPORT,
+		!enable
+	);
+}
+
+static inline void
+snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip)
+{
+	/*
+	 * skeleton handler only
+	 * (we do not want axis reading in interrupt handler - too much load!)
+	 */
+	snd_azf3328_dbggame("gameport irq\n");
+
+	 /* this should ACK the gameport IRQ properly, hopefully. */
+	snd_azf3328_game_inw(chip, IDX_GAME_AXIS_VALUE);
+}
+
+static int
+snd_azf3328_gameport_open(struct gameport *gameport, int mode)
+{
+	struct snd_azf3328 *chip = gameport_get_port_data(gameport);
+	int res;
+
+	snd_azf3328_dbggame("gameport_open, mode %d\n", mode);
+	switch (mode) {
+	case GAMEPORT_MODE_COOKED:
+	case GAMEPORT_MODE_RAW:
+		res = 0;
+		break;
+	default:
+		res = -1;
+		break;
+	}
+
+	snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0));
+
+	return res;
+}
+
+static void
+snd_azf3328_gameport_close(struct gameport *gameport)
+{
+	struct snd_azf3328 *chip = gameport_get_port_data(gameport);
+
+	snd_azf3328_dbggame("gameport_close\n");
+	snd_azf3328_gameport_axis_circuit_enable(chip, 0);
+}
+
+static int
+snd_azf3328_gameport_cooked_read(struct gameport *gameport,
+				 int *axes,
+				 int *buttons
+)
+{
+	struct snd_azf3328 *chip = gameport_get_port_data(gameport);
+	int i;
+	u8 val;
+	unsigned long flags;
+
+	snd_assert(chip, return 0);
+
+	spin_lock_irqsave(&chip->reg_lock, flags);
+	val = snd_azf3328_game_inb(chip, IDX_GAME_LEGACY_COMPATIBLE);
+	*buttons = (~(val) >> 4) & 0xf;
+
+	/* ok, this one is a bit dirty: cooked_read is being polled by a timer,
+	 * thus we're atomic and cannot actively wait in here
+	 * (which would be useful for us since it probably would be better
+	 * to trigger a measurement in here, then wait a short amount of
+	 * time until it's finished, then read values of _this_ measurement).
+	 *
+	 * Thus we simply resort to reading values if they're available already
+	 * and trigger the next measurement.
+	 */
+
+	val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG);
+	if (val & GAME_AXES_SAMPLING_READY) {
+		for (i = 0; i < 4; ++i) {
+			/* configure the axis to read */
+			val = (i << 4) | 0x0f;
+			snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val);
+
+			chip->axes[i] = snd_azf3328_game_inw(
+						chip, IDX_GAME_AXIS_VALUE
+					);
+		}
+	}
+
+	/* trigger next axes sampling, to be evaluated the next time we
+	 * enter this function */
+
+	/* for some very, very strange reason we cannot enable
+	 * Measurement Ready monitoring for all axes here,
+	 * at least not when only one joystick connected */
+	val = 0x03; /* we're able to monitor axes 1 and 2 only */
+	snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val);
+
+	snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff);
+	spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+	for (i = 0; i < 4; i++) {
+		axes[i] = chip->axes[i];
+		if (axes[i] == 0xffff)
+			axes[i] = -1;
+	}
+
+	snd_azf3328_dbggame("cooked_read: axes %d %d %d %d buttons %d\n",
+		axes[0], axes[1], axes[2], axes[3], *buttons
+	);
+
+	return 0;
+}
+
+static int __devinit
+snd_azf3328_gameport(struct snd_azf3328 *chip, int dev)
+{
+	struct gameport *gp;
+
+	int io_port = chip->game_io;
+
+	chip->gameport = gp = gameport_allocate_port();
+	if (!gp) {
+		printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n");
+		return -ENOMEM;
+	}
+
+	gameport_set_name(gp, "AZF3328 Gameport");
+	gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+	gameport_set_dev_parent(gp, &chip->pci->dev);
+	gp->io = io_port;
+	gameport_set_port_data(gp, chip);
+
+	gp->open = snd_azf3328_gameport_open;
+	gp->close = snd_azf3328_gameport_close;
+	gp->fuzz = 16; /* seems ok */
+	gp->cooked_read = snd_azf3328_gameport_cooked_read;
+
+	/* DISABLE legacy address: we don't need it! */
+	snd_azf3328_gameport_legacy_address_enable(chip, 0);
+
+	snd_azf3328_gameport_axis_circuit_enable(chip, 0);
+
+	gameport_register_port(chip->gameport);
+
+	return 0;
+}
+
+static void
+snd_azf3328_gameport_free(struct snd_azf3328 *chip)
+{
+	if (chip->gameport) {
+		gameport_unregister_port(chip->gameport);
+		chip->gameport = NULL;
+	}
+	snd_azf3328_gameport_irq_enable(chip, 0);
+}
+#else
+static inline int
+snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { return -ENOSYS; }
+static inline void
+snd_azf3328_gameport_free(struct snd_azf3328 *chip) { }
+static inline void
+snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip)
+{
+	printk(KERN_WARNING "huh, game port IRQ occurred!?\n");
+}
+#endif /* SUPPORT_GAMEPORT */
+
+/******************************************************************/
+
 static irqreturn_t
 snd_azf3328_interrupt(int irq, void *dev_id)
 {
 	struct snd_azf3328 *chip = dev_id;
 	u8 status, which;
+#if DEBUG_PLAY_REC
 	static unsigned long irq_count;
+#endif
 
 	status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS);
 
         /* fast path out, to ease interrupt sharing */
-	if (!(status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_MPU401|IRQ_TIMER)))
+	if (!(status &
+		(IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER)
+	))
 		return IRQ_NONE; /* must be interrupt for another device */
 
 	snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n",
-		irq_count,
+		irq_count++ /* debug-only */,
 		snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS),
 		snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE),
 		status);
-		
+
 	if (status & IRQ_TIMER) {
-		/* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */
+		/* snd_azf3328_dbgplay("timer %ld\n",
+			snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE)
+				& TIMER_VALUE_MASK
+		); */
 		if (chip->timer)
 			snd_timer_interrupt(chip->timer, chip->timer->sticks);
 		/* ACK timer */
@@ -1232,11 +1620,16 @@ snd_azf3328_interrupt(int irq, void *dev_id)
 		snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which);
                	spin_unlock(&chip->reg_lock);
 
-		if (chip->pcm && chip->playback_substream) {
-			snd_pcm_period_elapsed(chip->playback_substream);
+		if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) {
+			snd_pcm_period_elapsed(
+				chip->audio_stream[AZF_PLAYBACK].substream
+			);
 			snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n",
 				which,
-				inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS));
+				snd_azf3328_codec_inl(
+					chip, IDX_IO_PLAY_DMA_CURRPOS
+				)
+			);
 		} else
 			snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n");
 		if (which & IRQ_PLAY_SOMETHING)
@@ -1249,16 +1642,23 @@ snd_azf3328_interrupt(int irq, void *dev_id)
 		snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which);
 		spin_unlock(&chip->reg_lock);
 
-		if (chip->pcm && chip->capture_substream) {
-			snd_pcm_period_elapsed(chip->capture_substream);
+		if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) {
+			snd_pcm_period_elapsed(
+				chip->audio_stream[AZF_CAPTURE].substream
+			);
 			snd_azf3328_dbgplay("REC  period done (#%x), @ %x\n",
 				which,
-				inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS));
+				snd_azf3328_codec_inl(
+					chip, IDX_IO_REC_DMA_CURRPOS
+				)
+			);
 		} else
 			snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n");
 		if (which & IRQ_REC_SOMETHING)
 			snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n");
 	}
+	if (status & IRQ_GAMEPORT)
+		snd_azf3328_gameport_interrupt(chip);
 	/* MPU401 has less critical IRQ requirements
 	 * than timer and playback/recording, right? */
 	if (status & IRQ_MPU401) {
@@ -1268,7 +1668,6 @@ snd_azf3328_interrupt(int irq, void *dev_id)
 		 * If so, then I don't know how... */
 		snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n");
 	}
-	irq_count++;
 	return IRQ_HANDLED;
 }
 
@@ -1287,8 +1686,8 @@ static const struct snd_pcm_hardware snd_azf3328_playback =
 	.rates =		SNDRV_PCM_RATE_5512 |
 				SNDRV_PCM_RATE_8000_48000 |
 				SNDRV_PCM_RATE_KNOT,
-	.rate_min =		4000,
-	.rate_max =		66200,
+	.rate_min =		AZF_FREQ_4000,
+	.rate_max =		AZF_FREQ_66200,
 	.channels_min =		1,
 	.channels_max =		2,
 	.buffer_bytes_max =	65536,
@@ -1315,8 +1714,8 @@ static const struct snd_pcm_hardware snd_azf3328_capture =
 	.rates =		SNDRV_PCM_RATE_5512 |
 				SNDRV_PCM_RATE_8000_48000 |
 				SNDRV_PCM_RATE_KNOT,
-	.rate_min =		4000,
-	.rate_max =		66200,
+	.rate_min =		AZF_FREQ_4000,
+	.rate_max =		AZF_FREQ_66200,
 	.channels_min =		1,
 	.channels_max =		2,
 	.buffer_bytes_max =	65536,
@@ -1329,10 +1728,24 @@ static const struct snd_pcm_hardware snd_azf3328_capture =
 
 
 static unsigned int snd_azf3328_fixed_rates[] = {
-	4000, 4800, 5512, 6620, 8000, 9600, 11025, 13240, 16000, 22050, 32000,
-	44100, 48000, 66200 };
+	AZF_FREQ_4000,
+	AZF_FREQ_4800,
+	AZF_FREQ_5512,
+	AZF_FREQ_6620,
+	AZF_FREQ_8000,
+	AZF_FREQ_9600,
+	AZF_FREQ_11025,
+	AZF_FREQ_13240,
+	AZF_FREQ_16000,
+	AZF_FREQ_22050,
+	AZF_FREQ_32000,
+	AZF_FREQ_44100,
+	AZF_FREQ_48000,
+	AZF_FREQ_66200
+};
+
 static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = {
-	.count = ARRAY_SIZE(snd_azf3328_fixed_rates), 
+	.count = ARRAY_SIZE(snd_azf3328_fixed_rates),
 	.list = snd_azf3328_fixed_rates,
 	.mask = 0,
 };
@@ -1346,7 +1759,7 @@ snd_azf3328_playback_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
 	snd_azf3328_dbgcallenter();
-	chip->playback_substream = substream;
+	chip->audio_stream[AZF_PLAYBACK].substream = substream;
 	runtime->hw = snd_azf3328_playback;
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &snd_azf3328_hw_constraints_rates);
@@ -1361,7 +1774,7 @@ snd_azf3328_capture_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
 	snd_azf3328_dbgcallenter();
-	chip->capture_substream = substream;
+	chip->audio_stream[AZF_CAPTURE].substream = substream;
 	runtime->hw = snd_azf3328_capture;
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &snd_azf3328_hw_constraints_rates);
@@ -1375,7 +1788,7 @@ snd_azf3328_playback_close(struct snd_pcm_substream *substream)
 	struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
 
 	snd_azf3328_dbgcallenter();
-	chip->playback_substream = NULL;
+	chip->audio_stream[AZF_PLAYBACK].substream = NULL;
 	snd_azf3328_dbgcallleave();
 	return 0;
 }
@@ -1386,7 +1799,7 @@ snd_azf3328_capture_close(struct snd_pcm_substream *substream)
 	struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
 
 	snd_azf3328_dbgcallenter();
-	chip->capture_substream = NULL;
+	chip->audio_stream[AZF_CAPTURE].substream = NULL;
 	snd_azf3328_dbgcallleave();
 	return 0;
 }
@@ -1441,102 +1854,8 @@ snd_azf3328_pcm(struct snd_azf3328 *chip, int device)
 
 /******************************************************************/
 
-#ifdef SUPPORT_JOYSTICK
-static int __devinit
-snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev)
-{
-	struct gameport *gp;
-	struct resource *r;
-
-	if (!joystick[dev])
-		return -ENODEV;
-
-	if (!(r = request_region(0x200, 8, "AZF3328 gameport"))) {
-		printk(KERN_WARNING "azt3328: cannot reserve joystick ports\n");
-		return -EBUSY;
-	}
-
-	chip->gameport = gp = gameport_allocate_port();
-	if (!gp) {
-		printk(KERN_ERR "azt3328: cannot allocate memory for gameport\n");
-		release_and_free_resource(r);
-		return -ENOMEM;
-	}
-
-	gameport_set_name(gp, "AZF3328 Gameport");
-	gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
-	gameport_set_dev_parent(gp, &chip->pci->dev);
-	gp->io = 0x200;
-	gameport_set_port_data(gp, r);
-
-	snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR,
-			      snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) | LEGACY_JOY);
-
-	gameport_register_port(chip->gameport);
-
-	return 0;
-}
-
-static void
-snd_azf3328_free_joystick(struct snd_azf3328 *chip)
-{
-	if (chip->gameport) {
-		struct resource *r = gameport_get_port_data(chip->gameport);
-
-		gameport_unregister_port(chip->gameport);
-		chip->gameport = NULL;
-		/* disable gameport */
-		snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR,
-				      snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY);
-		release_and_free_resource(r);
-	}
-}
-#else
-static inline int
-snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) { return -ENOSYS; }
-static inline void
-snd_azf3328_free_joystick(struct snd_azf3328 *chip) { }
-#endif
-
-/******************************************************************/
-
-static int
-snd_azf3328_free(struct snd_azf3328 *chip)
-{
-        if (chip->irq < 0)
-                goto __end_hw;
-
-	/* reset (close) mixer */
-	snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); /* first mute master volume */
-	snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000);
-
-        /* interrupt setup - mask everything (FIXME!) */
-	/* well, at least we know how to disable the timer IRQ */
-	snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00);
-
-	if (chip->irq >= 0)
-        	synchronize_irq(chip->irq);
-__end_hw:
-	snd_azf3328_free_joystick(chip);
-        if (chip->irq >= 0)
-		free_irq(chip->irq, chip);
-	pci_release_regions(chip->pci);
-	pci_disable_device(chip->pci);
-
-        kfree(chip);
-        return 0;
-}
-
-static int
-snd_azf3328_dev_free(struct snd_device *device)
-{
-	struct snd_azf3328 *chip = device->device_data;
-	return snd_azf3328_free(chip);
-}
-
-/******************************************************************/
-
-/*** NOTE: the physical timer resolution actually is 1024000 ticks per second,
+/*** NOTE: the physical timer resolution actually is 1024000 ticks per second
+ *** (probably derived from main crystal via a divider of 24),
  *** but announcing those attributes to user-space would make programs
  *** configure the timer to a 1 tick value, resulting in an absolutely fatal
  *** timer IRQ storm.
@@ -1564,7 +1883,7 @@ snd_azf3328_timer_start(struct snd_timer *timer)
 		delay = 49; /* minimum time is 49 ticks */
 	}
 	snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay);
-	delay |= TIMER_ENABLE_COUNTDOWN | TIMER_ENABLE_IRQ;
+	delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE;
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
@@ -1582,7 +1901,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer)
 	chip = snd_timer_chip(timer);
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	/* disable timer countdown and interrupt */
-	/* FIXME: should we write TIMER_ACK_IRQ here? */
+	/* FIXME: should we write TIMER_IRQ_ACK here? */
 	snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	snd_azf3328_dbgcallleave();
@@ -1626,9 +1945,10 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device)
 
 	snd_azf3328_timer_hw.resolution *= seqtimer_scaling;
 	snd_azf3328_timer_hw.ticks /= seqtimer_scaling;
-	if ((err = snd_timer_new(chip->card, "AZF3328", &tid, &timer)) < 0) {
+
+	err = snd_timer_new(chip->card, "AZF3328", &tid, &timer);
+	if (err < 0)
 		goto out;
-	}
 
 	strcpy(timer->name, "AZF3328 timer");
 	timer->private_data = chip;
@@ -1636,6 +1956,8 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device)
 
 	chip->timer = timer;
 
+	snd_azf3328_timer_stop(timer);
+
 	err = 0;
 
 out:
@@ -1645,10 +1967,44 @@ out:
 
 /******************************************************************/
 
+static int
+snd_azf3328_free(struct snd_azf3328 *chip)
+{
+	if (chip->irq < 0)
+		goto __end_hw;
+
+	/* reset (close) mixer:
+	 * first mute master volume, then reset
+	 */
+	snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1);
+	snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000);
+
+	snd_azf3328_timer_stop(chip->timer);
+	snd_azf3328_gameport_free(chip);
+
+	if (chip->irq >= 0)
+		synchronize_irq(chip->irq);
+__end_hw:
+	if (chip->irq >= 0)
+		free_irq(chip->irq, chip);
+	pci_release_regions(chip->pci);
+	pci_disable_device(chip->pci);
+
+	kfree(chip);
+	return 0;
+}
+
+static int
+snd_azf3328_dev_free(struct snd_device *device)
+{
+	struct snd_azf3328 *chip = device->device_data;
+	return snd_azf3328_free(chip);
+}
+
 #if 0
 /* check whether a bit can be modified */
 static void
-snd_azf3328_test_bit(unsigned int reg, int bit)
+snd_azf3328_test_bit(unsigned unsigned reg, int bit)
 {
 	unsigned char val, valoff, valon;
 
@@ -1659,42 +2015,74 @@ snd_azf3328_test_bit(unsigned int reg, int bit)
 
 	outb(val|(1 << bit), reg);
 	valon = inb(reg);
-	
+
 	outb(val, reg);
 
-	printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", reg, bit, val, valoff, valon);
+	printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n",
+				reg, bit, val, valoff, valon
+	);
 }
 #endif
 
-#if DEBUG_MISC
-static void
+static inline void
 snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip)
 {
+#if DEBUG_MISC
 	u16 tmp;
 
-	snd_azf3328_dbgmisc("codec_port 0x%lx, io2_port 0x%lx, mpu_port 0x%lx, synth_port 0x%lx, mixer_port 0x%lx, irq %d\n", chip->codec_port, chip->io2_port, chip->mpu_port, chip->synth_port, chip->mixer_port, chip->irq);
-
-	snd_azf3328_dbgmisc("io2 %02x %02x %02x %02x %02x %02x\n", snd_azf3328_io2_inb(chip, 0), snd_azf3328_io2_inb(chip, 1), snd_azf3328_io2_inb(chip, 2), snd_azf3328_io2_inb(chip, 3), snd_azf3328_io2_inb(chip, 4), snd_azf3328_io2_inb(chip, 5));
-
-	for (tmp=0; tmp <= 0x01; tmp += 1)
-		snd_azf3328_dbgmisc("0x%02x: opl 0x%04x, mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, mpu330 0x%04x\n", tmp, inb(0x388 + tmp), inb(0x300 + tmp), inb(0x310 + tmp), inb(0x320 + tmp), inb(0x330 + tmp));
+	snd_azf3328_dbgmisc(
+		"codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, "
+		"opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n",
+		chip->codec_io, chip->game_io, chip->mpu_io,
+		chip->opl3_io, chip->mixer_io, chip->irq
+	);
+
+	snd_azf3328_dbgmisc("game %02x %02x %02x %02x %02x %02x\n",
+		snd_azf3328_game_inb(chip, 0),
+		snd_azf3328_game_inb(chip, 1),
+		snd_azf3328_game_inb(chip, 2),
+		snd_azf3328_game_inb(chip, 3),
+		snd_azf3328_game_inb(chip, 4),
+		snd_azf3328_game_inb(chip, 5)
+	);
+
+	for (tmp = 0; tmp < 0x07; tmp += 1)
+		snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp));
+
+	for (tmp = 0; tmp <= 0x07; tmp += 1)
+		snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n",
+			tmp, inb(0x200 + tmp), inb(0x208 + tmp));
+
+	for (tmp = 0; tmp <= 0x01; tmp += 1)
+		snd_azf3328_dbgmisc(
+			"0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, "
+			"mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n",
+				tmp,
+				inb(0x300 + tmp),
+				inb(0x310 + tmp),
+				inb(0x320 + tmp),
+				inb(0x330 + tmp),
+				inb(0x388 + tmp),
+				inb(0x38c + tmp)
+		);
 
 	for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2)
-		snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", tmp, snd_azf3328_codec_inw(chip, tmp));
+		snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n",
+			tmp, snd_azf3328_codec_inw(chip, tmp)
+		);
 
 	for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2)
-		snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", tmp, snd_azf3328_mixer_inw(chip, tmp));
+		snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n",
+			tmp, snd_azf3328_mixer_inw(chip, tmp)
+		);
+#endif /* DEBUG_MISC */
 }
-#else
-static inline void
-snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) {}
-#endif
 
 static int __devinit
 snd_azf3328_create(struct snd_card *card,
-                                         struct pci_dev *pci,
-                                         unsigned long device_type,
-                                         struct snd_azf3328 ** rchip)
+		   struct pci_dev *pci,
+		   unsigned long device_type,
+		   struct snd_azf3328 **rchip)
 {
 	struct snd_azf3328 *chip;
 	int err;
@@ -1705,7 +2093,8 @@ snd_azf3328_create(struct snd_card *card,
 
 	*rchip = NULL;
 
-	if ((err = pci_enable_device(pci)) < 0)
+	err = pci_enable_device(pci);
+	if (err < 0)
 		return err;
 
 	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
@@ -1721,20 +2110,25 @@ snd_azf3328_create(struct snd_card *card,
 	/* check if we can restrict PCI DMA transfers to 24 bits */
 	if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 ||
 	    pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) {
-		snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n");
+		snd_printk(KERN_ERR "architecture does not support "
+					"24bit PCI busmaster DMA\n"
+		);
 		err = -ENXIO;
 		goto out_err;
 	}
 
-	if ((err = pci_request_regions(pci, "Aztech AZF3328")) < 0) {
+	err = pci_request_regions(pci, "Aztech AZF3328");
+	if (err < 0)
 		goto out_err;
-	}
 
-	chip->codec_port = pci_resource_start(pci, 0);
-	chip->io2_port   = pci_resource_start(pci, 1);
-	chip->mpu_port   = pci_resource_start(pci, 2);
-	chip->synth_port = pci_resource_start(pci, 3);
-	chip->mixer_port = pci_resource_start(pci, 4);
+	chip->codec_io = pci_resource_start(pci, 0);
+	chip->game_io  = pci_resource_start(pci, 1);
+	chip->mpu_io   = pci_resource_start(pci, 2);
+	chip->opl3_io = pci_resource_start(pci, 3);
+	chip->mixer_io = pci_resource_start(pci, 4);
+
+	chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00;
+	chip->audio_stream[AZF_CAPTURE].portbase   = chip->codec_io + 0x20;
 
 	if (request_irq(pci->irq, snd_azf3328_interrupt,
 			IRQF_SHARED, card->shortname, chip)) {
@@ -1747,29 +2141,29 @@ snd_azf3328_create(struct snd_card *card,
 	synchronize_irq(chip->irq);
 
 	snd_azf3328_debug_show_ports(chip);
-	
-	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0)
 		goto out_err;
-	}
 
 	/* create mixer interface & switches */
-	if ((err = snd_azf3328_mixer_new(chip)) < 0)
+	err = snd_azf3328_mixer_new(chip);
+	if (err < 0)
 		goto out_err;
 
-#if 0
-	/* set very low bitrate to reduce noise and power consumption? */
-	snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, 5512, 8, 1);
-#endif
+	/* shutdown codecs to save power */
+		/* have snd_azf3328_codec_activity() act properly */
+	chip->audio_stream[AZF_PLAYBACK].running = 1;
+	snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0);
 
 	/* standard chip init stuff */
-	/* default IRQ init value */
+		/* default IRQ init value */
 	tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE;
 
 	spin_lock_irq(&chip->reg_lock);
 	snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp);
 	snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp);
 	snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp);
-	snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); /* disable timer */
 	spin_unlock_irq(&chip->reg_lock);
 
 	snd_card_set_dev(card, &pci->dev);
@@ -1805,52 +2199,61 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0 );
+	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
 	if (card == NULL)
 		return -ENOMEM;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
 
-        if ((err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip)) < 0) {
+	err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip);
+	if (err < 0)
 		goto out_err;
-	}
 
 	card->private_data = chip;
 
-	if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401,
-				        chip->mpu_port, MPU401_INFO_INTEGRATED,
-					pci->irq, 0, &chip->rmidi)) < 0) {
-		snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port);
+	err = snd_mpu401_uart_new(
+		card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED,
+		pci->irq, 0, &chip->rmidi
+	);
+	if (err < 0) {
+		snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n",
+				chip->mpu_io
+		);
 		goto out_err;
 	}
 
-	if ((err = snd_azf3328_timer(chip, 0)) < 0) {
+	err = snd_azf3328_timer(chip, 0);
+	if (err < 0)
 		goto out_err;
-	}
 
-	if ((err = snd_azf3328_pcm(chip, 0)) < 0) {
+	err = snd_azf3328_pcm(chip, 0);
+	if (err < 0)
 		goto out_err;
-	}
 
-	if (snd_opl3_create(card, chip->synth_port, chip->synth_port+2,
+	if (snd_opl3_create(card, chip->opl3_io, chip->opl3_io+2,
 			    OPL3_HW_AUTO, 1, &opl3) < 0) {
 		snd_printk(KERN_ERR "azf3328: no OPL3 device at 0x%lx-0x%lx?\n",
-			   chip->synth_port, chip->synth_port+2 );
+			   chip->opl3_io, chip->opl3_io+2
+		);
 	} else {
-		if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+		/* need to use IDs 1, 2 since ID 0 is snd_azf3328_timer above */
+		err = snd_opl3_timer_new(opl3, 1, 2);
+		if (err < 0)
+			goto out_err;
+		err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+		if (err < 0)
 			goto out_err;
-		}
 	}
 
 	opl3->private_data = chip;
 
 	sprintf(card->longname, "%s at 0x%lx, irq %i",
-		card->shortname, chip->codec_port, chip->irq);
+		card->shortname, chip->codec_io, chip->irq);
 
-	if ((err = snd_card_register(card)) < 0) {
+	err = snd_card_register(card);
+	if (err < 0)
 		goto out_err;
-	}
 
 #ifdef MODULE
 	printk(
@@ -1861,19 +2264,18 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	1024000 / seqtimer_scaling, seqtimer_scaling);
 #endif
 
-	if (snd_azf3328_config_joystick(chip, dev) < 0)
-		snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR,
-			      snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY);
+	snd_azf3328_gameport(chip, dev);
 
 	pci_set_drvdata(pci, card);
 	dev++;
 
 	err = 0;
 	goto out;
-	
+
 out_err:
+	snd_printk(KERN_ERR "azf3328: something failed, exiting\n");
 	snd_card_free(card);
-	
+
 out:
 	snd_azf3328_dbgcallleave();
 	return err;
@@ -1894,27 +2296,27 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct snd_azf3328 *chip = card->private_data;
-	int reg;
+	unsigned reg;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-	
+
 	snd_pcm_suspend_all(chip->pcm);
 
-	for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++)
-		chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg)
+		chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2);
 
 	/* make sure to disable master volume etc. to prevent looping sound */
 	snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1);
 	snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
-	
-	for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++)
-		chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++)
-		chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++)
-		chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++)
-		chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2);
+
+	for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg)
+		chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg)
+		chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg)
+		chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg)
+		chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2);
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
@@ -1927,7 +2329,7 @@ snd_azf3328_resume(struct pci_dev *pci)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct snd_azf3328 *chip = card->private_data;
-	int reg;
+	unsigned reg;
 
 	pci_set_power_state(pci, PCI_D0);
 	pci_restore_state(pci);
@@ -1939,23 +2341,21 @@ snd_azf3328_resume(struct pci_dev *pci)
 	}
 	pci_set_master(pci);
 
-	for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++)
-		outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++)
-		outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++)
-		outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++)
-		outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2);
-	for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++)
-		outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg)
+		outw(chip->saved_regs_game[reg], chip->game_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg)
+		outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg)
+		outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg)
+		outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg)
+		outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
-
-
+#endif /* CONFIG_PM */
 
 
 static struct pci_driver driver = {
diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h
index 679fa992e2bc..3448fd626f80 100644
--- a/sound/pci/azt3328.h
+++ b/sound/pci/azt3328.h
@@ -54,7 +54,10 @@
   #define SOUNDFORMAT_XTAL1		0x00
   #define SOUNDFORMAT_XTAL2		0x01
     /* all _SUSPECTED_ values are not used by Windows drivers, so we don't
-     * have any hard facts, only rough measurements */
+     * have any hard facts, only rough measurements.
+     * All we know is that the crystal used on the board has 24.576MHz,
+     * like many soundcards (which results in the frequencies below when
+     * using certain divider values selected by the values below) */
     #define SOUNDFORMAT_FREQ_SUSPECTED_4000	0x0c | SOUNDFORMAT_XTAL1
     #define SOUNDFORMAT_FREQ_SUSPECTED_4800	0x0a | SOUNDFORMAT_XTAL1
     #define SOUNDFORMAT_FREQ_5510		0x0c | SOUNDFORMAT_XTAL2
@@ -72,6 +75,26 @@
   #define SOUNDFORMAT_FLAG_16BIT	0x0010
   #define SOUNDFORMAT_FLAG_2CHANNELS	0x0020
 
+/* define frequency helpers, for maximum value safety */
+enum {
+#define AZF_FREQ(rate) AZF_FREQ_##rate = rate
+  AZF_FREQ(4000),
+  AZF_FREQ(4800),
+  AZF_FREQ(5512),
+  AZF_FREQ(6620),
+  AZF_FREQ(8000),
+  AZF_FREQ(9600),
+  AZF_FREQ(11025),
+  AZF_FREQ(13240),
+  AZF_FREQ(16000),
+  AZF_FREQ(22050),
+  AZF_FREQ(32000),
+  AZF_FREQ(44100),
+  AZF_FREQ(48000),
+  AZF_FREQ(66200),
+#undef AZF_FREQ
+} AZF_FREQUENCIES;
+
 /** recording area (see also: playback bit flag definitions) **/
 #define IDX_IO_REC_FLAGS	0x20 /* ??, PU:0x0000 */
 #define IDX_IO_REC_IRQTYPE	0x22 /* ??, PU:0x0000 */
@@ -97,40 +120,164 @@
 
 /** DirectX timer, main interrupt area (FIXME: and something else?) **/ 
 #define IDX_IO_TIMER_VALUE	0x60 /* found this timer area by pure luck :-) */
-  #define TIMER_VALUE_MASK		0x000fffffUL /* timer countdown value; triggers IRQ when timer is finished */
-  #define TIMER_ENABLE_COUNTDOWN	0x01000000UL /* activate the timer countdown */
-  #define TIMER_ENABLE_IRQ		0x02000000UL /* trigger timer IRQ on zero transition */
-  #define TIMER_ACK_IRQ			0x04000000UL /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) had 0x0020 set upon IRQ handler */
+  /* timer countdown value; triggers IRQ when timer is finished */
+  #define TIMER_VALUE_MASK		0x000fffffUL
+  /* activate timer countdown */
+  #define TIMER_COUNTDOWN_ENABLE	0x01000000UL
+  /* trigger timer IRQ on zero transition */
+  #define TIMER_IRQ_ENABLE		0x02000000UL
+  /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?)
+   * had 0x0020 set upon IRQ handler */
+  #define TIMER_IRQ_ACK			0x04000000UL
 #define IDX_IO_IRQSTATUS        0x64
-  #define IRQ_PLAYBACK			0x0001
-  #define IRQ_RECORDING			0x0002
-  #define IRQ_MPU401			0x0010
-  #define IRQ_TIMER			0x0020 /* DirectX timer */
-  #define IRQ_UNKNOWN1			0x0040 /* probably unused, or possibly I2S port? or gameport IRQ? */
-  #define IRQ_UNKNOWN2			0x0080 /* probably unused, or possibly I2S port? or gameport IRQ? */
+  /* some IRQ bit in here might also be used to signal a power-management timer
+   * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing).
+   * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which
+   * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */
+
+  #define IRQ_PLAYBACK	0x0001
+  #define IRQ_RECORDING	0x0002
+  #define IRQ_UNKNOWN1	0x0004 /* most probably I2S port */
+  #define IRQ_GAMEPORT	0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */
+  #define IRQ_MPU401	0x0010
+  #define IRQ_TIMER	0x0020 /* DirectX timer */
+  #define IRQ_UNKNOWN2	0x0040 /* probably unused, or possibly I2S port? */
+  #define IRQ_UNKNOWN3	0x0080 /* probably unused, or possibly I2S port? */
 #define IDX_IO_66H		0x66    /* writing 0xffff returns 0x0000 */
-#define IDX_IO_SOME_VALUE	0x68	/* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */
-#define IDX_IO_6AH		0x6A	/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */
-  #define IO_6A_PAUSE_PLAYBACK		0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */
-#define IDX_IO_6CH		0x6C
-#define IDX_IO_6EH		0x6E	/* writing 0xffff returns 0x83fe */
-/* further I/O indices not saved/restored, so probably not used */
+  /* this is set to e.g. 0x3ff or 0x300, and writable;
+   * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */
+#define IDX_IO_SOME_VALUE	0x68
+  #define IO_68_RANDOM_TOGGLE1	0x0100	/* toggles randomly */
+  #define IO_68_RANDOM_TOGGLE2	0x0200	/* toggles randomly */
+  /* umm, nope, behaviour of these bits changes depending on what we wrote
+   * to 0x6b!! */
+
+/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??);
+ * actually inhibits PCM playback!!! maybe power management??: */
+#define IDX_IO_6AH		0x6A
+  /* bit 5: enabling this will activate permanent counting of bytes 2/3
+   * at gameport I/O (0xb402/3) (equal values each) and cause
+   * gameport legacy I/O at 0x0200 to be _DISABLED_!
+   * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode
+   * for Enhanced Digital Gameport (see 4D Wave DX card): */
+  #define IO_6A_SOMETHING1_GAMEPORT	0x0020
+  /* bit 8; sure, this _pauses_ playback (later resumes at same spot!),
+   * but what the heck is this really about??: */
+  #define IO_6A_PAUSE_PLAYBACK_BIT8	0x0100
+  /* bit 9; sure, this _pauses_ playback (later resumes at same spot!),
+   * but what the heck is this really about??: */
+  #define IO_6A_PAUSE_PLAYBACK_BIT9	0x0200
+	/* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback,
+	 * thus it suggests influence on PCM only!!
+	 * However OTOH there seems to be no bit anywhere around here
+	 * which is able to disable OPL3... */
+  /* bit 10: enabling this actually changes values at legacy gameport
+   * I/O address (0x200); is this enabling of the Digital Enhanced Game Port???
+   * Or maybe this simply switches off the NE558 circuit, since enabling this
+   * still lets us evaluate button states, but not axis states */
+  #define IO_6A_SOMETHING2_GAMEPORT      0x0400
+	/* writing 0x0300: causes quite some crackling during
+	 * PC activity such as switching windows (PCI traffic??
+	 * --> FIFO/timing settings???) */
+	/* writing 0x0100 plus/or 0x0200 inhibits playback */
+	/* since the Windows .INF file has Flag_Enable_JoyStick and
+	 * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason
+	 * that some other bit in this same register might be responsible
+	 * for SB DOS Emulation activation (note that the file did NOT define
+	 * a switch for OPL3!) */
+#define IDX_IO_6CH		0x6C	/* unknown; fully read-writable */
+#define IDX_IO_6EH		0x6E
+	/* writing 0xffff returns 0x83fe (or 0x03fe only).
+	 * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch
+	 * from 0000 to ffff. */
 
+/* further I/O indices not saved/restored and not readable after writing,
+ * so probably not used */
 
-/*** I/O 2 area port indices ***/
+
+/*** Gameport area port indices ***/
 /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ 
-#define AZF_IO_SIZE_IO2		0x08
-#define AZF_IO_SIZE_IO2_PM	0x06
+#define AZF_IO_SIZE_GAME		0x08
+#define AZF_IO_SIZE_GAME_PM	0x06
+
+enum {
+	AZF_GAME_LEGACY_IO_PORT = 0x200
+} AZF_GAME_CONFIGS;
+
+#define IDX_GAME_LEGACY_COMPATIBLE	0x00
+	/* in some operation mode, writing anything to this port
+	 * triggers an interrupt:
+	 * yup, that's in case IDX_GAME_01H has one of the
+	 * axis measurement bits enabled
+	 * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */
+
+#define IDX_GAME_AXES_CONFIG            0x01
+	/* NOTE: layout of this register awfully similar (read: "identical??")
+	 * to AD1815JS.pdf (p.29) */
+
+  /* enables axis 1 (X axis) measurement: */
+  #define GAME_AXES_ENABLE_1		0x01
+  /* enables axis 2 (Y axis) measurement: */
+  #define GAME_AXES_ENABLE_2		0x02
+  /* enables axis 3 (X axis) measurement: */
+  #define GAME_AXES_ENABLE_3		0x04
+  /* enables axis 4 (Y axis) measurement: */
+  #define GAME_AXES_ENABLE_4		0x08
+  /* selects the current axis to read the measured value of
+   * (at IDX_GAME_AXIS_VALUE):
+   * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */
+  #define GAME_AXES_READ_MASK		0x30
+  /* enable to have the latch continuously accept ADC values
+   * (and continuously cause interrupts in case interrupts are enabled);
+   * AD1815JS.pdf says it's ~16ms interval there: */
+  #define GAME_AXES_LATCH_ENABLE	0x40
+  /* joystick data (measured axes) ready for reading: */
+  #define GAME_AXES_SAMPLING_READY	0x80
+
+  /* NOTE: other card specs (SiS960 and others!) state that the
+   * game position latches should be frozen when reading and be freed
+   * (== reset?) after reading!!!
+   * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE),
+   *  but how to free the value? */
+  /* An internet search for "gameport latch ADC" should provide some insight
+   * into how to program such a gameport system. */
+
+  /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!?
+   * yup, in case 6AH 0x20 is not enabled
+   * (and 0x40 is sufficient, 0xf0 is not needed) */
+
+#define IDX_GAME_AXIS_VALUE	0x02
+	/* R: value of currently configured axis (word value!);
+	 * W: trigger axis measurement */
+
+#define IDX_GAME_HWCONFIG	0x04
+	/* note: bits 4 to 7 are never set (== 0) when reading!
+	 * --> reserved bits? */
+  /* enables IRQ notification upon axes measurement ready: */
+  #define GAME_HWCFG_IRQ_ENABLE			0x01
+  /* these bits choose a different frequency for the
+   *  internal ADC counter increment.
+   * hmm, seems to be a combo of bits:
+   * 00 --> standard frequency
+   * 10 --> 1/2
+   * 01 --> 1/20
+   * 11 --> 1/200: */
+  #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK	0x06
 
-#define IDX_IO2_LEGACY_ADDR	0x04
-  #define LEGACY_SOMETHING		0x01 /* OPL3?? */
-  #define LEGACY_JOY			0x08
+  /* enable gameport legacy I/O address (0x200)
+   * I was unable to locate any configurability for a different address: */
+  #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE	0x08
 
+/*** MPU401 ***/
 #define AZF_IO_SIZE_MPU		0x04
 #define AZF_IO_SIZE_MPU_PM	0x04
 
-#define AZF_IO_SIZE_SYNTH	0x08
-#define AZF_IO_SIZE_SYNTH_PM	0x06
+/*** OPL3 synth ***/
+#define AZF_IO_SIZE_OPL3	0x08
+#define AZF_IO_SIZE_OPL3_PM	0x06
+/* hmm, given that a standard OPL3 has 4 registers only,
+ * there might be some enhanced functionality lurking at the end
+ * (especially since register 0x04 has a "non-empty" value 0xfe) */
 
 /*** mixer I/O area port indices ***/
 /* (only 0x22 of 0x40 bytes saved/restored by Windows driver)
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a6be6e3e8716..d2e1093f8e97 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2335,7 +2335,7 @@ int snd_hda_check_board_config(struct hda_codec *codec,
 	if (!tbl)
 		return -1;
 	if (tbl->value >= 0 && tbl->value < num_configs) {
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 		char tmp[10];
 		const char *model = NULL;
 		if (models)
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 2177d9af5334..6e18a422d993 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -88,7 +88,7 @@ static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file,
 
 static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
 {
-#ifndef CONFIG_SND_DEBUG_DETECT
+#ifndef CONFIG_SND_DEBUG_VERBOSE
 	if (!capable(CAP_SYS_RAWIO))
 		return -EACCES;
 #endif
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b3a618eb42cd..6ba7ac01d9f6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -285,6 +285,7 @@ struct azx_dev {
 	u32 *posbuf;		/* position buffer pointer */
 
 	unsigned int bufsize;	/* size of the play buffer in bytes */
+	unsigned int period_bytes; /* size of the period in bytes */
 	unsigned int frags;	/* number for period in the play buffer */
 	unsigned int fifo_size;	/* FIFO size */
 
@@ -301,11 +302,10 @@ struct azx_dev {
 					 */
 	unsigned char stream_tag;	/* assigned stream */
 	unsigned char index;		/* stream index */
-	/* for sanity check of position buffer */
-	unsigned int period_intr;
 
 	unsigned int opened :1;
 	unsigned int running :1;
+	unsigned int irq_pending: 1;
 };
 
 /* CORB/RIRB */
@@ -369,6 +369,9 @@ struct azx {
 
 	/* for debugging */
 	unsigned int last_cmd;	/* last issued command (to sync) */
+
+	/* for pending irqs */
+	struct work_struct irq_pending_work;
 };
 
 /* driver types */
@@ -908,6 +911,8 @@ static void azx_init_pci(struct azx *chip)
 }
 
 
+static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev);
+
 /*
  * interrupt handler
  */
@@ -930,11 +935,18 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 		azx_dev = &chip->azx_dev[i];
 		if (status & azx_dev->sd_int_sta_mask) {
 			azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
-			if (azx_dev->substream && azx_dev->running) {
-				azx_dev->period_intr++;
+			if (!azx_dev->substream || !azx_dev->running)
+				continue;
+			/* check whether this IRQ is really acceptable */
+			if (azx_position_ok(chip, azx_dev)) {
+				azx_dev->irq_pending = 0;
 				spin_unlock(&chip->reg_lock);
 				snd_pcm_period_elapsed(azx_dev->substream);
 				spin_lock(&chip->reg_lock);
+			} else {
+				/* bogus IRQ, process it later */
+				azx_dev->irq_pending = 1;
+				schedule_work(&chip->irq_pending_work);
 			}
 		}
 	}
@@ -973,6 +985,7 @@ static int azx_setup_periods(struct snd_pcm_substream *substream,
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 
 	period_bytes = snd_pcm_lib_period_bytes(substream);
+	azx_dev->period_bytes = period_bytes;
 	periods = azx_dev->bufsize / period_bytes;
 
 	/* program the initial BDL entries */
@@ -1421,27 +1434,16 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	return 0;
 }
 
-static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
+static unsigned int azx_get_position(struct azx *chip,
+				     struct azx_dev *azx_dev)
 {
-	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
-	struct azx *chip = apcm->chip;
-	struct azx_dev *azx_dev = get_azx_dev(substream);
 	unsigned int pos;
 
 	if (chip->position_fix == POS_FIX_POSBUF ||
 	    chip->position_fix == POS_FIX_AUTO) {
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
-		if (chip->position_fix == POS_FIX_AUTO &&
-		    azx_dev->period_intr == 1 && !pos) {
-			printk(KERN_WARNING
-			       "hda-intel: Invalid position buffer, "
-			       "using LPIB read method instead.\n");
-			chip->position_fix = POS_FIX_NONE;
-			goto read_lpib;
-		}
 	} else {
-	read_lpib:
 		/* read LPIB */
 		pos = azx_sd_readl(azx_dev, SD_LPIB);
 		if (chip->position_fix == POS_FIX_FIFO)
@@ -1449,7 +1451,90 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
 	}
 	if (pos >= azx_dev->bufsize)
 		pos = 0;
-	return bytes_to_frames(substream->runtime, pos);
+	return pos;
+}
+
+static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	struct azx *chip = apcm->chip;
+	struct azx_dev *azx_dev = get_azx_dev(substream);
+	return bytes_to_frames(substream->runtime,
+			       azx_get_position(chip, azx_dev));
+}
+
+/*
+ * Check whether the current DMA position is acceptable for updating
+ * periods.  Returns non-zero if it's OK.
+ *
+ * Many HD-audio controllers appear pretty inaccurate about
+ * the update-IRQ timing.  The IRQ is issued before actually the
+ * data is processed.  So, we need to process it afterwords in a
+ * workqueue.
+ */
+static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
+{
+	unsigned int pos;
+
+	pos = azx_get_position(chip, azx_dev);
+	if (chip->position_fix == POS_FIX_AUTO) {
+		if (!pos) {
+			printk(KERN_WARNING
+			       "hda-intel: Invalid position buffer, "
+			       "using LPIB read method instead.\n");
+			chip->position_fix = POS_FIX_NONE;
+			pos = azx_get_position(chip, azx_dev);
+		} else
+			chip->position_fix = POS_FIX_POSBUF;
+	}
+
+	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
+		return 0; /* NG - it's below the period boundary */
+	return 1; /* OK, it's fine */
+}
+
+/*
+ * The work for pending PCM period updates.
+ */
+static void azx_irq_pending_work(struct work_struct *work)
+{
+	struct azx *chip = container_of(work, struct azx, irq_pending_work);
+	int i, pending;
+
+	for (;;) {
+		pending = 0;
+		spin_lock_irq(&chip->reg_lock);
+		for (i = 0; i < chip->num_streams; i++) {
+			struct azx_dev *azx_dev = &chip->azx_dev[i];
+			if (!azx_dev->irq_pending ||
+			    !azx_dev->substream ||
+			    !azx_dev->running)
+				continue;
+			if (azx_position_ok(chip, azx_dev)) {
+				azx_dev->irq_pending = 0;
+				spin_unlock(&chip->reg_lock);
+				snd_pcm_period_elapsed(azx_dev->substream);
+				spin_lock(&chip->reg_lock);
+			} else
+				pending++;
+		}
+		spin_unlock_irq(&chip->reg_lock);
+		if (!pending)
+			return;
+		cond_resched();
+	}
+}
+
+/* clear irq_pending flags and assure no on-going workq */
+static void azx_clear_irq_pending(struct azx *chip)
+{
+	int i;
+
+	spin_lock_irq(&chip->reg_lock);
+	for (i = 0; i < chip->num_streams; i++)
+		chip->azx_dev[i].irq_pending = 0;
+	spin_unlock_irq(&chip->reg_lock);
+	flush_scheduled_work();
 }
 
 static struct snd_pcm_ops azx_pcm_ops = {
@@ -1676,6 +1761,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 	int i;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	azx_clear_irq_pending(chip);
 	for (i = 0; i < AZX_MAX_PCMS; i++)
 		snd_pcm_suspend_all(chip->pcm[i]);
 	if (chip->initialized)
@@ -1732,6 +1818,7 @@ static int azx_free(struct azx *chip)
 	int i;
 
 	if (chip->initialized) {
+		azx_clear_irq_pending(chip);
 		for (i = 0; i < chip->num_streams; i++)
 			azx_stream_stop(chip, &chip->azx_dev[i]);
 		azx_stop_chip(chip);
@@ -1857,6 +1944,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	chip->irq = -1;
 	chip->driver_type = driver_type;
 	chip->msi = enable_msi;
+	INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
 
 	chip->position_fix = check_position_fix(chip, position_fix[dev]);
 	check_probe_mask(chip, dev);
diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h
index 43b9e3e858be..a0c5e009bb4a 100644
--- a/sound/pci/ice1712/envy24ht.h
+++ b/sound/pci/ice1712/envy24ht.h
@@ -93,9 +93,13 @@ enum {
 #define VT1724_REG_MPU_TXFIFO		0x0a	/*byte ro. number of bytes in TX fifo*/
 #define VT1724_REG_MPU_RXFIFO		0x0b	/*byte ro. number of bytes in RX fifo*/
 
-//are these 2 the wrong way around? they don't seem to be used yet anyway
-#define VT1724_REG_MPU_CTRL		0x0c	/* byte */
-#define VT1724_REG_MPU_DATA		0x0d	/* byte */
+#define VT1724_REG_MPU_DATA		0x0c	/* byte */
+#define VT1724_REG_MPU_CTRL		0x0d	/* byte */
+#define   VT1724_MPU_UART	0x01
+#define   VT1724_MPU_TX_EMPTY	0x02
+#define   VT1724_MPU_TX_FULL	0x04
+#define   VT1724_MPU_RX_EMPTY	0x08
+#define   VT1724_MPU_RX_FULL	0x10
 
 #define VT1724_REG_MPU_FIFO_WM	0x0e	/*byte set the high/low watermarks for RX/TX fifos*/
 #define   VT1724_MPU_RX_FIFO	0x20	//1=rx fifo watermark 0=tx fifo watermark
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 3208901c740e..762fbd7a7507 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -333,6 +333,8 @@ struct snd_ice1712 {
 	unsigned int has_spdif: 1;	/* VT1720/4 - has SPDIF I/O */
 	unsigned int force_pdma4: 1;	/* VT1720/4 - PDMA4 as non-spdif */
 	unsigned int force_rdma1: 1;	/* VT1720/4 - RDMA1 as non-spdif */
+	unsigned int midi_output: 1;	/* VT1720/4: MIDI output triggered */
+	unsigned int midi_input: 1;	/* VT1720/4: MIDI input triggered */
 	unsigned int num_total_dacs;	/* total DACs */
 	unsigned int num_total_adcs;	/* total ADCs */
 	unsigned int cur_rate;		/* current rate */
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 67350901772c..e596d777d9dd 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -32,7 +32,7 @@
 #include <linux/mutex.h>
 #include <sound/core.h>
 #include <sound/info.h>
-#include <sound/mpu401.h>
+#include <sound/rawmidi.h>
 #include <sound/initval.h>
 
 #include <sound/asoundef.h>
@@ -223,30 +223,153 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice)
 }
 
 /*
- * MPU401 accessor
+ * MIDI
  */
-static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu,
-					    unsigned long addr)
+
+static void vt1724_midi_clear_rx(struct snd_ice1712 *ice)
+{
+	unsigned int count;
+
+	for (count = inb(ICEREG1724(ice, MPU_RXFIFO)); count > 0; --count)
+		inb(ICEREG1724(ice, MPU_DATA));
+}
+
+static inline struct snd_rawmidi_substream *
+get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream)
 {
-	/* fix status bits to the standard position */
-	/* only RX_EMPTY and TX_FULL are checked */
-	if (addr == MPU401C(mpu))
-		return (inb(addr) & 0x0c) << 4;
+	return list_first_entry(&ice->rmidi[0]->streams[stream].substreams,
+				struct snd_rawmidi_substream, list);
+}
+
+static void vt1724_midi_write(struct snd_ice1712 *ice)
+{
+	struct snd_rawmidi_substream *s;
+	int count, i;
+	u8 buffer[32];
+
+	s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_OUTPUT);
+	count = 31 - inb(ICEREG1724(ice, MPU_TXFIFO));
+	if (count > 0) {
+		count = snd_rawmidi_transmit(s, buffer, count);
+		for (i = 0; i < count; ++i)
+			outb(buffer[i], ICEREG1724(ice, MPU_DATA));
+	}
+}
+
+static void vt1724_midi_read(struct snd_ice1712 *ice)
+{
+	struct snd_rawmidi_substream *s;
+	int count, i;
+	u8 buffer[32];
+
+	s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_INPUT);
+	count = inb(ICEREG1724(ice, MPU_RXFIFO));
+	if (count > 0) {
+		count = min(count, 32);
+		for (i = 0; i < count; ++i)
+			buffer[i] = inb(ICEREG1724(ice, MPU_DATA));
+		snd_rawmidi_receive(s, buffer, count);
+	}
+}
+
+static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
+				   u8 flag, int enable)
+{
+	struct snd_ice1712 *ice = substream->rmidi->private_data;
+	u8 mask;
+
+	spin_lock_irq(&ice->reg_lock);
+	mask = inb(ICEREG1724(ice, IRQMASK));
+	if (enable)
+		mask &= ~flag;
 	else
-		return inb(addr);
+		mask |= flag;
+	outb(mask, ICEREG1724(ice, IRQMASK));
+	spin_unlock_irq(&ice->reg_lock);
 }
 
-static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu,
-				    unsigned char data, unsigned long addr)
+static int vt1724_midi_output_open(struct snd_rawmidi_substream *s)
 {
-	if (addr == MPU401C(mpu)) {
-		if (data == MPU401_ENTER_UART)
-			outb(0x01, addr);
-		/* what else? */
-	} else
-		outb(data, addr);
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1);
+	return 0;
+}
+
+static int vt1724_midi_output_close(struct snd_rawmidi_substream *s)
+{
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
+	return 0;
 }
 
+static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up)
+{
+	struct snd_ice1712 *ice = s->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&ice->reg_lock, flags);
+	if (up) {
+		ice->midi_output = 1;
+		vt1724_midi_write(ice);
+	} else {
+		ice->midi_output = 0;
+	}
+	spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s)
+{
+	struct snd_ice1712 *ice = s->rmidi->private_data;
+	unsigned long timeout;
+
+	/* 32 bytes should be transmitted in less than about 12 ms */
+	timeout = jiffies + msecs_to_jiffies(15);
+	do {
+		if (inb(ICEREG1724(ice, MPU_CTRL)) & VT1724_MPU_TX_EMPTY)
+			break;
+		schedule_timeout_uninterruptible(1);
+	} while (time_after(timeout, jiffies));
+}
+
+static struct snd_rawmidi_ops vt1724_midi_output_ops = {
+	.open = vt1724_midi_output_open,
+	.close = vt1724_midi_output_close,
+	.trigger = vt1724_midi_output_trigger,
+	.drain = vt1724_midi_output_drain,
+};
+
+static int vt1724_midi_input_open(struct snd_rawmidi_substream *s)
+{
+	vt1724_midi_clear_rx(s->rmidi->private_data);
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 1);
+	return 0;
+}
+
+static int vt1724_midi_input_close(struct snd_rawmidi_substream *s)
+{
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 0);
+	return 0;
+}
+
+static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up)
+{
+	struct snd_ice1712 *ice = s->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&ice->reg_lock, flags);
+	if (up) {
+		ice->midi_input = 1;
+		vt1724_midi_read(ice);
+	} else {
+		ice->midi_input = 0;
+	}
+	spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops vt1724_midi_input_ops = {
+	.open = vt1724_midi_input_open,
+	.close = vt1724_midi_input_close,
+	.trigger = vt1724_midi_input_trigger,
+};
+
 
 /*
  *  Interrupt handler
@@ -278,13 +401,10 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 #endif
 		handled = 1;		
 		if (status & VT1724_IRQ_MPU_TX) {
-			if (ice->rmidi[0])
-				snd_mpu401_uart_interrupt_tx(irq,
-					ice->rmidi[0]->private_data);
-			else /* disable TX to be sure */
-				outb(inb(ICEREG1724(ice, IRQMASK)) |
-				     VT1724_IRQ_MPU_TX,
-				     ICEREG1724(ice, IRQMASK));
+			spin_lock(&ice->reg_lock);
+			if (ice->midi_output)
+				vt1724_midi_write(ice);
+			spin_unlock(&ice->reg_lock);
 			/* Due to mysterical reasons, MPU_TX is always
 			 * generated (and can't be cleared) when a PCM
 			 * playback is going.  So let's ignore at the
@@ -293,13 +413,12 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 			status_mask &= ~VT1724_IRQ_MPU_TX;
 		}
 		if (status & VT1724_IRQ_MPU_RX) {
-			if (ice->rmidi[0])
-				snd_mpu401_uart_interrupt(irq,
-					ice->rmidi[0]->private_data);
-			else /* disable RX to be sure */
-				outb(inb(ICEREG1724(ice, IRQMASK)) |
-				     VT1724_IRQ_MPU_RX,
-				     ICEREG1724(ice, IRQMASK));
+			spin_lock(&ice->reg_lock);
+			if (ice->midi_input)
+				vt1724_midi_read(ice);
+			else
+				vt1724_midi_clear_rx(ice);
+			spin_unlock(&ice->reg_lock);
 		}
 		/* ack MPU irq */
 		outb(status, ICEREG1724(ice, IRQSTAT));
@@ -2425,28 +2544,30 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 
 	if (! c->no_mpu401) {
 		if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) {
-			struct snd_mpu401 *mpu;
-			if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
-						       ICEREG1724(ice, MPU_CTRL),
-						       (MPU401_INFO_INTEGRATED |
-							MPU401_INFO_NO_ACK |
-							MPU401_INFO_TX_IRQ),
-						       ice->irq, 0,
-						       &ice->rmidi[0])) < 0) {
+			struct snd_rawmidi *rmidi;
+
+			err = snd_rawmidi_new(card, "MIDI", 0, 1, 1, &rmidi);
+			if (err < 0) {
 				snd_card_free(card);
 				return err;
 			}
-			mpu = ice->rmidi[0]->private_data;
-			mpu->read = snd_vt1724_mpu401_read;
-			mpu->write = snd_vt1724_mpu401_write;
-			/* unmask MPU RX/TX irqs */
-			outb(inb(ICEREG1724(ice, IRQMASK)) &
-			     ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX),
-			     ICEREG1724(ice, IRQMASK));
+			ice->rmidi[0] = rmidi;
+			rmidi->private_data = ice;
+			strcpy(rmidi->name, "ICE1724 MIDI");
+			rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+					    SNDRV_RAWMIDI_INFO_INPUT |
+					    SNDRV_RAWMIDI_INFO_DUPLEX;
+			snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+					    &vt1724_midi_output_ops);
+			snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+					    &vt1724_midi_input_ops);
+
 			/* set watermarks */
 			outb(VT1724_MPU_RX_FIFO | 0x1,
 			     ICEREG1724(ice, MPU_FIFO_WM));
 			outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
+			/* set UART mode */
+			outb(VT1724_MPU_UART, ICEREG1724(ice, MPU_CTRL));
 		}
 	}
 
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 090dd4354a28..7442460583dd 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -28,7 +28,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("TempoTec HiFier driver");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
@@ -62,16 +62,28 @@ static void ak4396_write(struct oxygen *chip, u8 reg, u8 value)
 			 AK4396_WRITE | (reg << 8) | value);
 }
 
-static void hifier_init(struct oxygen *chip)
+static void update_ak4396_volume(struct oxygen *chip)
+{
+	ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]);
+	ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]);
+}
+
+static void hifier_registers_init(struct oxygen *chip)
 {
 	struct hifier_data *data = chip->model_data;
 
-	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
 	ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
 	ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2);
 	ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM);
-	ak4396_write(chip, AK4396_LCH_ATT, 0);
-	ak4396_write(chip, AK4396_RCH_ATT, 0);
+	update_ak4396_volume(chip);
+}
+
+static void hifier_init(struct oxygen *chip)
+{
+	struct hifier_data *data = chip->model_data;
+
+	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+	hifier_registers_init(chip);
 
 	snd_component_add(chip->card, "AK4396");
 	snd_component_add(chip->card, "CS5340");
@@ -100,12 +112,6 @@ static void set_ak4396_params(struct oxygen *chip,
 	ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
 }
 
-static void update_ak4396_volume(struct oxygen *chip)
-{
-	ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]);
-	ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]);
-}
-
 static void update_ak4396_mute(struct oxygen *chip)
 {
 	struct hifier_data *data = chip->model_data;
@@ -140,6 +146,7 @@ static const struct oxygen_model model_hifier = {
 	.init = hifier_init,
 	.control_filter = hifier_control_filter,
 	.cleanup = hifier_cleanup,
+	.resume = hifier_registers_init,
 	.set_dac_params = set_ak4396_params,
 	.set_adc_params = set_cs5340_params,
 	.update_dac_volume = update_ak4396_volume,
@@ -180,6 +187,10 @@ static struct pci_driver hifier_driver = {
 	.id_table = hifier_ids,
 	.probe = hifier_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+	.suspend = oxygen_pci_suspend,
+	.resume = oxygen_pci_resume,
+#endif
 };
 
 static int __init alsa_card_hifier_init(void)
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 63f185c1ed1e..7c8ae31eb468 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -43,7 +43,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 driver");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
@@ -80,6 +80,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids);
 
 struct generic_data {
 	u8 ak4396_ctl2;
+	u16 saved_wm8785_registers[2];
 };
 
 static void ak4396_write(struct oxygen *chip, unsigned int codec,
@@ -99,20 +100,35 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec,
 
 static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value)
 {
+	struct generic_data *data = chip->model_data;
+
 	oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER |
 			 OXYGEN_SPI_DATA_LENGTH_2 |
 			 OXYGEN_SPI_CLOCK_160 |
 			 (3 << OXYGEN_SPI_CODEC_SHIFT) |
 			 OXYGEN_SPI_CEN_LATCH_CLOCK_LO,
 			 (reg << 9) | value);
+	if (reg < ARRAY_SIZE(data->saved_wm8785_registers))
+		data->saved_wm8785_registers[reg] = value;
 }
 
-static void ak4396_init(struct oxygen *chip)
+static void update_ak4396_volume(struct oxygen *chip)
+{
+	unsigned int i;
+
+	for (i = 0; i < 4; ++i) {
+		ak4396_write(chip, i,
+			     AK4396_LCH_ATT, chip->dac_volume[i * 2]);
+		ak4396_write(chip, i,
+			     AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]);
+	}
+}
+
+static void ak4396_registers_init(struct oxygen *chip)
 {
 	struct generic_data *data = chip->model_data;
 	unsigned int i;
 
-	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
 	for (i = 0; i < 4; ++i) {
 		ak4396_write(chip, i,
 			     AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
@@ -120,9 +136,16 @@ static void ak4396_init(struct oxygen *chip)
 			     AK4396_CONTROL_2, data->ak4396_ctl2);
 		ak4396_write(chip, i,
 			     AK4396_CONTROL_3, AK4396_PCM);
-		ak4396_write(chip, i, AK4396_LCH_ATT, 0);
-		ak4396_write(chip, i, AK4396_RCH_ATT, 0);
 	}
+	update_ak4396_volume(chip);
+}
+
+static void ak4396_init(struct oxygen *chip)
+{
+	struct generic_data *data = chip->model_data;
+
+	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+	ak4396_registers_init(chip);
 	snd_component_add(chip->card, "AK4396");
 }
 
@@ -133,12 +156,23 @@ static void ak5385_init(struct oxygen *chip)
 	snd_component_add(chip->card, "AK5385");
 }
 
-static void wm8785_init(struct oxygen *chip)
+static void wm8785_registers_init(struct oxygen *chip)
 {
+	struct generic_data *data = chip->model_data;
+
 	wm8785_write(chip, WM8785_R7, 0);
-	wm8785_write(chip, WM8785_R0, WM8785_MCR_SLAVE |
-		     WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST);
-	wm8785_write(chip, WM8785_R1, WM8785_WL_24);
+	wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]);
+	wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]);
+}
+
+static void wm8785_init(struct oxygen *chip)
+{
+	struct generic_data *data = chip->model_data;
+
+	data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE |
+		WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST;
+	data->saved_wm8785_registers[1] = WM8785_WL_24;
+	wm8785_registers_init(chip);
 	snd_component_add(chip->card, "WM8785");
 }
 
@@ -158,6 +192,12 @@ static void generic_cleanup(struct oxygen *chip)
 {
 }
 
+static void generic_resume(struct oxygen *chip)
+{
+	ak4396_registers_init(chip);
+	wm8785_registers_init(chip);
+}
+
 static void set_ak4396_params(struct oxygen *chip,
 			      struct snd_pcm_hw_params *params)
 {
@@ -183,18 +223,6 @@ static void set_ak4396_params(struct oxygen *chip,
 	}
 }
 
-static void update_ak4396_volume(struct oxygen *chip)
-{
-	unsigned int i;
-
-	for (i = 0; i < 4; ++i) {
-		ak4396_write(chip, i,
-			     AK4396_LCH_ATT, chip->dac_volume[i * 2]);
-		ak4396_write(chip, i,
-			     AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]);
-	}
-}
-
 static void update_ak4396_mute(struct oxygen *chip)
 {
 	struct generic_data *data = chip->model_data;
@@ -256,6 +284,7 @@ static const struct oxygen_model model_generic = {
 	.owner = THIS_MODULE,
 	.init = generic_init,
 	.cleanup = generic_cleanup,
+	.resume = generic_resume,
 	.set_dac_params = set_ak4396_params,
 	.set_adc_params = set_wm8785_params,
 	.update_dac_volume = update_ak4396_volume,
@@ -283,6 +312,7 @@ static const struct oxygen_model model_meridian = {
 	.owner = THIS_MODULE,
 	.init = meridian_init,
 	.cleanup = generic_cleanup,
+	.resume = ak4396_registers_init,
 	.set_dac_params = set_ak4396_params,
 	.set_adc_params = set_ak5385_params,
 	.update_dac_volume = update_ak4396_volume,
@@ -331,6 +361,10 @@ static struct pci_driver oxygen_driver = {
 	.id_table = oxygen_ids,
 	.probe = generic_oxygen_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+	.suspend = oxygen_pci_suspend,
+	.resume = oxygen_pci_resume,
+#endif
 };
 
 static int __init alsa_card_oxygen_init(void)
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index a71c6e059260..74a644880074 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -16,6 +16,8 @@
 #define PCM_AC97	5
 #define PCM_COUNT	6
 
+#define OXYGEN_IO_SIZE	0x100
+
 /* model-specific configuration of outputs/inputs */
 #define PLAYBACK_0_TO_I2S	0x001
 #define PLAYBACK_1_TO_SPDIF	0x004
@@ -78,6 +80,12 @@ struct oxygen {
 	struct work_struct spdif_input_bits_work;
 	struct work_struct gpio_work;
 	wait_queue_head_t ac97_waitqueue;
+	union {
+		u8 _8[OXYGEN_IO_SIZE];
+		__le16 _16[OXYGEN_IO_SIZE / 2];
+		__le32 _32[OXYGEN_IO_SIZE / 4];
+	} saved_registers;
+	u16 saved_ac97_registers[2][0x40];
 };
 
 struct oxygen_model {
@@ -89,6 +97,8 @@ struct oxygen_model {
 	int (*control_filter)(struct snd_kcontrol_new *template);
 	int (*mixer_init)(struct oxygen *chip);
 	void (*cleanup)(struct oxygen *chip);
+	void (*suspend)(struct oxygen *chip);
+	void (*resume)(struct oxygen *chip);
 	void (*pcm_hardware_filter)(unsigned int channel,
 				    struct snd_pcm_hardware *hardware);
 	void (*set_dac_params)(struct oxygen *chip,
@@ -117,6 +127,10 @@ struct oxygen_model {
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		     const struct oxygen_model *model);
 void oxygen_pci_remove(struct pci_dev *pci);
+#ifdef CONFIG_PM
+int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
+int oxygen_pci_resume(struct pci_dev *pci);
+#endif
 
 /* oxygen_mixer.c */
 
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
index 5569606ee87f..83f135f80df4 100644
--- a/sound/pci/oxygen/oxygen_io.c
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -44,18 +44,21 @@ EXPORT_SYMBOL(oxygen_read32);
 void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value)
 {
 	outb(value, chip->addr + reg);
+	chip->saved_registers._8[reg] = value;
 }
 EXPORT_SYMBOL(oxygen_write8);
 
 void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value)
 {
 	outw(value, chip->addr + reg);
+	chip->saved_registers._16[reg / 2] = cpu_to_le16(value);
 }
 EXPORT_SYMBOL(oxygen_write16);
 
 void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value)
 {
 	outl(value, chip->addr + reg);
+	chip->saved_registers._32[reg / 4] = cpu_to_le32(value);
 }
 EXPORT_SYMBOL(oxygen_write32);
 
@@ -63,7 +66,10 @@ void oxygen_write8_masked(struct oxygen *chip, unsigned int reg,
 			  u8 value, u8 mask)
 {
 	u8 tmp = inb(chip->addr + reg);
-	outb((tmp & ~mask) | (value & mask), chip->addr + reg);
+	tmp &= ~mask;
+	tmp |= value & mask;
+	outb(tmp, chip->addr + reg);
+	chip->saved_registers._8[reg] = tmp;
 }
 EXPORT_SYMBOL(oxygen_write8_masked);
 
@@ -71,7 +77,10 @@ void oxygen_write16_masked(struct oxygen *chip, unsigned int reg,
 			   u16 value, u16 mask)
 {
 	u16 tmp = inw(chip->addr + reg);
-	outw((tmp & ~mask) | (value & mask), chip->addr + reg);
+	tmp &= ~mask;
+	tmp |= value & mask;
+	outw(tmp, chip->addr + reg);
+	chip->saved_registers._16[reg / 2] = cpu_to_le16(tmp);
 }
 EXPORT_SYMBOL(oxygen_write16_masked);
 
@@ -79,7 +88,10 @@ void oxygen_write32_masked(struct oxygen *chip, unsigned int reg,
 			   u32 value, u32 mask)
 {
 	u32 tmp = inl(chip->addr + reg);
-	outl((tmp & ~mask) | (value & mask), chip->addr + reg);
+	tmp &= ~mask;
+	tmp |= value & mask;
+	outl(tmp, chip->addr + reg);
+	chip->saved_registers._32[reg / 4] = cpu_to_le32(tmp);
 }
 EXPORT_SYMBOL(oxygen_write32_masked);
 
@@ -128,8 +140,10 @@ void oxygen_write_ac97(struct oxygen *chip, unsigned int codec,
 		oxygen_write32(chip, OXYGEN_AC97_REGS, reg);
 		/* require two "completed" writes, just to be sure */
 		if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 &&
-		    ++succeeded >= 2)
+		    ++succeeded >= 2) {
+			chip->saved_ac97_registers[codec][index / 2] = data;
 			return;
+		}
 	}
 	snd_printk(KERN_ERR "AC'97 write timeout\n");
 }
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 897697d43506..22f37851045e 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -32,7 +32,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 helper library");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 
 
 static irqreturn_t oxygen_interrupt(int dummy, void *dev_id)
@@ -173,7 +173,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry,
 	int i, j;
 
 	snd_iprintf(buffer, "CMI8788\n\n");
-	for (i = 0; i < 0x100; i += 0x10) {
+	for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) {
 		snd_iprintf(buffer, "%02x:", i);
 		for (j = 0; j < 0x10; ++j)
 			snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j));
@@ -314,6 +314,10 @@ static void oxygen_init(struct oxygen *chip)
 				    OXYGEN_SPDIF_LOCK_MASK |
 				    OXYGEN_SPDIF_RATE_MASK);
 	oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits);
+	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+		       OXYGEN_2WIRE_LENGTH_8 |
+		       OXYGEN_2WIRE_INTERRUPT_MASK |
+		       OXYGEN_2WIRE_SPEED_STANDARD);
 	oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK);
 	oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0);
 	oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0);
@@ -455,7 +459,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	}
 
 	if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) ||
-	    pci_resource_len(pci, 0) < 0x100) {
+	    pci_resource_len(pci, 0) < OXYGEN_IO_SIZE) {
 		snd_printk(KERN_ERR "invalid PCI I/O range\n");
 		err = -ENXIO;
 		goto err_pci_regions;
@@ -534,3 +538,99 @@ void oxygen_pci_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 EXPORT_SYMBOL(oxygen_pci_remove);
+
+#ifdef CONFIG_PM
+int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct oxygen *chip = card->private_data;
+	unsigned int i, saved_interrupt_mask;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+	for (i = 0; i < PCM_COUNT; ++i)
+		if (chip->streams[i])
+			snd_pcm_suspend(chip->streams[i]);
+
+	if (chip->model->suspend)
+		chip->model->suspend(chip);
+
+	spin_lock_irq(&chip->reg_lock);
+	saved_interrupt_mask = chip->interrupt_mask;
+	chip->interrupt_mask = 0;
+	oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+	oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+	spin_unlock_irq(&chip->reg_lock);
+
+	synchronize_irq(chip->irq);
+	flush_scheduled_work();
+	chip->interrupt_mask = saved_interrupt_mask;
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+	pci_set_power_state(pci, pci_choose_state(pci, state));
+	return 0;
+}
+EXPORT_SYMBOL(oxygen_pci_suspend);
+
+static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = {
+	0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff,
+	0x00300000, 0x00000fe4, 0x0ff7001f, 0x00000000
+};
+static const u32 ac97_registers_to_restore[2][0x40 / 32] = {
+	{ 0x18284fa2, 0x03060000 },
+	{ 0x00007fa6, 0x00200000 }
+};
+
+static inline int is_bit_set(const u32 *bitmap, unsigned int bit)
+{
+	return bitmap[bit / 32] & (1 << (bit & 31));
+}
+
+static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec)
+{
+	unsigned int i;
+
+	oxygen_write_ac97(chip, codec, AC97_RESET, 0);
+	msleep(1);
+	for (i = 1; i < 0x40; ++i)
+		if (is_bit_set(ac97_registers_to_restore[codec], i))
+			oxygen_write_ac97(chip, codec, i * 2,
+					  chip->saved_ac97_registers[codec][i]);
+}
+
+int oxygen_pci_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct oxygen *chip = card->private_data;
+	unsigned int i;
+
+	pci_set_power_state(pci, PCI_D0);
+	pci_restore_state(pci);
+	if (pci_enable_device(pci) < 0) {
+		snd_printk(KERN_ERR "cannot reenable device");
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+	pci_set_master(pci);
+
+	oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+	oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+	for (i = 0; i < OXYGEN_IO_SIZE; ++i)
+		if (is_bit_set(registers_to_restore, i))
+			oxygen_write8(chip, i, chip->saved_registers._8[i]);
+	if (chip->has_ac97_0)
+		oxygen_restore_ac97(chip, 0);
+	if (chip->has_ac97_1)
+		oxygen_restore_ac97(chip, 1);
+
+	if (chip->model->resume)
+		chip->model->resume(chip);
+
+	oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+EXPORT_SYMBOL(oxygen_pci_resume);
+#endif /* CONFIG_PM */
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index b17c405e069d..c4ad65a3406f 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -24,6 +24,16 @@
 #include <sound/pcm_params.h>
 #include "oxygen.h"
 
+/* most DMA channels have a 16-bit counter for 32-bit words */
+#define BUFFER_BYTES_MAX		((1 << 16) * 4)
+/* the multichannel DMA channel has a 24-bit counter */
+#define BUFFER_BYTES_MAX_MULTICH	((1 << 24) * 4)
+
+#define PERIOD_BYTES_MIN		64
+
+#define DEFAULT_BUFFER_BYTES		(BUFFER_BYTES_MAX / 2)
+#define DEFAULT_BUFFER_BYTES_MULTICH	(1024 * 1024)
+
 static const struct snd_pcm_hardware oxygen_stereo_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_MMAP_VALID |
@@ -44,11 +54,11 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = {
 	.rate_max = 192000,
 	.channels_min = 2,
 	.channels_max = 2,
-	.buffer_bytes_max = 256 * 1024,
-	.period_bytes_min = 128,
-	.period_bytes_max = 128 * 1024,
+	.buffer_bytes_max = BUFFER_BYTES_MAX,
+	.period_bytes_min = PERIOD_BYTES_MIN,
+	.period_bytes_max = BUFFER_BYTES_MAX / 2,
 	.periods_min = 2,
-	.periods_max = 2048,
+	.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
 };
 static const struct snd_pcm_hardware oxygen_multichannel_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP |
@@ -70,11 +80,11 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = {
 	.rate_max = 192000,
 	.channels_min = 2,
 	.channels_max = 8,
-	.buffer_bytes_max = 2048 * 1024,
-	.period_bytes_min = 128,
-	.period_bytes_max = 256 * 1024,
+	.buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH,
+	.period_bytes_min = PERIOD_BYTES_MIN,
+	.period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2,
 	.periods_min = 2,
-	.periods_max = 16384,
+	.periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN,
 };
 static const struct snd_pcm_hardware oxygen_ac97_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP |
@@ -88,11 +98,11 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = {
 	.rate_max = 48000,
 	.channels_min = 2,
 	.channels_max = 2,
-	.buffer_bytes_max = 256 * 1024,
-	.period_bytes_min = 128,
-	.period_bytes_max = 128 * 1024,
+	.buffer_bytes_max = BUFFER_BYTES_MAX,
+	.period_bytes_min = PERIOD_BYTES_MIN,
+	.period_bytes_max = BUFFER_BYTES_MAX / 2,
 	.periods_min = 2,
-	.periods_max = 2048,
+	.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
 };
 
 static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = {
@@ -155,6 +165,12 @@ static int oxygen_open(struct snd_pcm_substream *substream,
 		if (err < 0)
 			return err;
 	}
+	if (channel == PCM_MULTICH) {
+		err = snd_pcm_hw_constraint_minmax
+			(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000);
+		if (err < 0)
+			return err;
+	}
 	snd_pcm_set_sync(substream);
 	chip->streams[channel] = substream;
 
@@ -517,6 +533,7 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 		pausing = 0;
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -663,12 +680,14 @@ int oxygen_pcm_init(struct oxygen *chip)
 			snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
 						      SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      512 * 1024, 2048 * 1024);
+						      DEFAULT_BUFFER_BYTES_MULTICH,
+						      BUFFER_BYTES_MAX_MULTICH);
 		if (ins)
 			snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
 						      SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      128 * 1024, 256 * 1024);
+						      DEFAULT_BUFFER_BYTES,
+						      BUFFER_BYTES_MAX);
 	}
 
 	outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF);
@@ -688,7 +707,8 @@ int oxygen_pcm_init(struct oxygen *chip)
 		strcpy(pcm->name, "Digital");
 		snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      128 * 1024, 256 * 1024);
+						      DEFAULT_BUFFER_BYTES,
+						      BUFFER_BYTES_MAX);
 	}
 
 	if (chip->has_ac97_1) {
@@ -718,7 +738,8 @@ int oxygen_pcm_init(struct oxygen *chip)
 		strcpy(pcm->name, outs ? "Front Panel" : "Analog 2");
 		snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      128 * 1024, 256 * 1024);
+						      DEFAULT_BUFFER_BYTES,
+						      BUFFER_BYTES_MAX);
 	}
 	return 0;
 }
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 7f84fa5deca2..9a2c16bf94e0 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -79,7 +79,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("Asus AVx00 driver");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
@@ -132,6 +132,9 @@ struct xonar_data {
 	u8 ext_power_int_reg;
 	u8 ext_power_bit;
 	u8 has_power;
+	u8 pcm1796_oversampling;
+	u8 cs4398_fm;
+	u8 cs4362a_fm;
 };
 
 static void pcm1796_write(struct oxygen *chip, unsigned int codec,
@@ -159,6 +162,14 @@ static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value)
 	oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value);
 }
 
+static void xonar_enable_output(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	msleep(data->anti_pop_delay);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+}
+
 static void xonar_common_init(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -170,32 +181,59 @@ static void xonar_common_init(struct oxygen *chip)
 		data->has_power = !!(oxygen_read8(chip, data->ext_power_reg)
 				     & data->ext_power_bit);
 	}
-	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+			  GPIO_CS53x1_M_MASK | data->output_enable_bit);
 	oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
 			      GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK);
 	oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC);
-	msleep(data->anti_pop_delay);
-	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit);
-	oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+	xonar_enable_output(chip);
 }
 
-static void xonar_d2_init(struct oxygen *chip)
+static void update_pcm1796_volume(struct oxygen *chip)
 {
-	struct xonar_data *data = chip->model_data;
 	unsigned int i;
 
-	data->anti_pop_delay = 300;
-	data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
+	for (i = 0; i < 4; ++i) {
+		pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]);
+		pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]);
+	}
+}
+
+static void update_pcm1796_mute(struct oxygen *chip)
+{
+	unsigned int i;
+	u8 value;
+
+	value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD;
+	if (chip->dac_mute)
+		value |= PCM1796_MUTE;
+	for (i = 0; i < 4; ++i)
+		pcm1796_write(chip, i, 18, value);
+}
+
+static void pcm1796_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+	unsigned int i;
 
 	for (i = 0; i < 4; ++i) {
-		pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED |
-			      PCM1796_FMT_24_LJUST | PCM1796_ATLD);
 		pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1);
-		pcm1796_write(chip, i, 20, PCM1796_OS_64);
+		pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
 		pcm1796_write(chip, i, 21, 0);
-		pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */
-		pcm1796_write(chip, i, 17, 0x0f);
 	}
+	update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */
+	update_pcm1796_volume(chip);
+}
+
+static void xonar_d2_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->anti_pop_delay = 300;
+	data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
+	data->pcm1796_oversampling = PCM1796_OS_64;
+
+	pcm1796_init(chip);
 
 	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT);
 	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT);
@@ -217,31 +255,47 @@ static void xonar_d2x_init(struct oxygen *chip)
 	xonar_d2_init(chip);
 }
 
-static void xonar_dx_init(struct oxygen *chip)
+static void update_cs4362a_volumes(struct oxygen *chip)
 {
-	struct xonar_data *data = chip->model_data;
+	u8 mute;
 
-	data->anti_pop_delay = 800;
-	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
-	data->ext_power_reg = OXYGEN_GPI_DATA;
-	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
-	data->ext_power_bit = GPI_DX_EXT_POWER;
+	mute = chip->dac_mute ? CS4362A_MUTE : 0;
+	cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute);
+	cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute);
+	cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute);
+	cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute);
+	cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute);
+	cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute);
+}
 
-	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
-		       OXYGEN_2WIRE_LENGTH_8 |
-		       OXYGEN_2WIRE_INTERRUPT_MASK |
-		       OXYGEN_2WIRE_SPEED_FAST);
+static void update_cs43xx_volume(struct oxygen *chip)
+{
+	cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2);
+	cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2);
+	update_cs4362a_volumes(chip);
+}
+
+static void update_cs43xx_mute(struct oxygen *chip)
+{
+	u8 reg;
+
+	reg = CS4398_MUTEP_LOW | CS4398_PAMUTE;
+	if (chip->dac_mute)
+		reg |= CS4398_MUTE_B | CS4398_MUTE_A;
+	cs4398_write(chip, 4, reg);
+	update_cs4362a_volumes(chip);
+}
+
+static void cs43xx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
 
 	/* set CPEN (control port mode) and power down */
 	cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN);
 	cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
 	/* configure */
-	cs4398_write(chip, 2, CS4398_FM_SINGLE |
-		     CS4398_DEM_NONE | CS4398_DIF_LJUST);
+	cs4398_write(chip, 2, data->cs4398_fm);
 	cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L);
-	cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE);
-	cs4398_write(chip, 5, 0xfe);
-	cs4398_write(chip, 6, 0xfe);
 	cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP |
 		     CS4398_ZERO_CROSS | CS4398_SOFT_RAMP);
 	cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST);
@@ -249,21 +303,35 @@ static void xonar_dx_init(struct oxygen *chip)
 		      CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP);
 	cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE);
 	cs4362a_write(chip, 0x05, 0);
-	cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE |
-		      CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
-	cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE |
-		      CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
-	cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE |
-		      CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
-	cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE);
+	cs4362a_write(chip, 0x06, data->cs4362a_fm);
+	cs4362a_write(chip, 0x09, data->cs4362a_fm);
+	cs4362a_write(chip, 0x0c, data->cs4362a_fm);
+	update_cs43xx_volume(chip);
+	update_cs43xx_mute(chip);
 	/* clear power down */
 	cs4398_write(chip, 8, CS4398_CPEN);
 	cs4362a_write(chip, 0x01, CS4362A_CPEN);
+}
+
+static void xonar_dx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->anti_pop_delay = 800;
+	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+	data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
+	data->cs4362a_fm = CS4362A_FM_SINGLE |
+		CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+
+	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+		       OXYGEN_2WIRE_LENGTH_8 |
+		       OXYGEN_2WIRE_INTERRUPT_MASK |
+		       OXYGEN_2WIRE_SPEED_FAST);
+
+	cs43xx_init(chip);
 
 	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
 			  GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
@@ -291,37 +359,28 @@ static void xonar_dx_cleanup(struct oxygen *chip)
 	oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
 }
 
-static void set_pcm1796_params(struct oxygen *chip,
-			       struct snd_pcm_hw_params *params)
+static void xonar_d2_resume(struct oxygen *chip)
 {
-	unsigned int i;
-	u8 value;
-
-	value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
-	for (i = 0; i < 4; ++i)
-		pcm1796_write(chip, i, 20, value);
+	pcm1796_init(chip);
+	xonar_enable_output(chip);
 }
 
-static void update_pcm1796_volume(struct oxygen *chip)
+static void xonar_dx_resume(struct oxygen *chip)
 {
-	unsigned int i;
-
-	for (i = 0; i < 4; ++i) {
-		pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]);
-		pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]);
-	}
+	cs43xx_init(chip);
+	xonar_enable_output(chip);
 }
 
-static void update_pcm1796_mute(struct oxygen *chip)
+static void set_pcm1796_params(struct oxygen *chip,
+			       struct snd_pcm_hw_params *params)
 {
+	struct xonar_data *data = chip->model_data;
 	unsigned int i;
-	u8 value;
 
-	value = PCM1796_FMT_24_LJUST | PCM1796_ATLD;
-	if (chip->dac_mute)
-		value |= PCM1796_MUTE;
+	data->pcm1796_oversampling =
+		params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
 	for (i = 0; i < 4; ++i)
-		pcm1796_write(chip, i, 18, value);
+		pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
 }
 
 static void set_cs53x1_params(struct oxygen *chip,
@@ -342,55 +401,24 @@ static void set_cs53x1_params(struct oxygen *chip,
 static void set_cs43xx_params(struct oxygen *chip,
 			      struct snd_pcm_hw_params *params)
 {
-	u8 fm_cs4398, fm_cs4362a;
+	struct xonar_data *data = chip->model_data;
 
-	fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST;
-	fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+	data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST;
+	data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
 	if (params_rate(params) <= 50000) {
-		fm_cs4398 |= CS4398_FM_SINGLE;
-		fm_cs4362a |= CS4362A_FM_SINGLE;
+		data->cs4398_fm |= CS4398_FM_SINGLE;
+		data->cs4362a_fm |= CS4362A_FM_SINGLE;
 	} else if (params_rate(params) <= 100000) {
-		fm_cs4398 |= CS4398_FM_DOUBLE;
-		fm_cs4362a |= CS4362A_FM_DOUBLE;
+		data->cs4398_fm |= CS4398_FM_DOUBLE;
+		data->cs4362a_fm |= CS4362A_FM_DOUBLE;
 	} else {
-		fm_cs4398 |= CS4398_FM_QUAD;
-		fm_cs4362a |= CS4362A_FM_QUAD;
+		data->cs4398_fm |= CS4398_FM_QUAD;
+		data->cs4362a_fm |= CS4362A_FM_QUAD;
 	}
-	cs4398_write(chip, 2, fm_cs4398);
-	cs4362a_write(chip, 0x06, fm_cs4362a);
-	cs4362a_write(chip, 0x09, fm_cs4362a);
-	cs4362a_write(chip, 0x0c, fm_cs4362a);
-}
-
-static void update_cs4362a_volumes(struct oxygen *chip)
-{
-	u8 mute;
-
-	mute = chip->dac_mute ? CS4362A_MUTE : 0;
-	cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute);
-	cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute);
-	cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute);
-	cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute);
-	cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute);
-	cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute);
-}
-
-static void update_cs43xx_volume(struct oxygen *chip)
-{
-	cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2);
-	cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2);
-	update_cs4362a_volumes(chip);
-}
-
-static void update_cs43xx_mute(struct oxygen *chip)
-{
-	u8 reg;
-
-	reg = CS4398_MUTEP_LOW | CS4398_PAMUTE;
-	if (chip->dac_mute)
-		reg |= CS4398_MUTE_B | CS4398_MUTE_A;
-	cs4398_write(chip, 4, reg);
-	update_cs4362a_volumes(chip);
+	cs4398_write(chip, 2, data->cs4398_fm);
+	cs4362a_write(chip, 0x06, data->cs4362a_fm);
+	cs4362a_write(chip, 0x09, data->cs4362a_fm);
+	cs4362a_write(chip, 0x0c, data->cs4362a_fm);
 }
 
 static void xonar_gpio_changed(struct oxygen *chip)
@@ -535,6 +563,8 @@ static const struct oxygen_model xonar_models[] = {
 		.control_filter = xonar_d2_control_filter,
 		.mixer_init = xonar_mixer_init,
 		.cleanup = xonar_cleanup,
+		.suspend = xonar_cleanup,
+		.resume = xonar_d2_resume,
 		.set_dac_params = set_pcm1796_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_pcm1796_volume,
@@ -563,6 +593,8 @@ static const struct oxygen_model xonar_models[] = {
 		.control_filter = xonar_d2_control_filter,
 		.mixer_init = xonar_mixer_init,
 		.cleanup = xonar_cleanup,
+		.suspend = xonar_cleanup,
+		.resume = xonar_d2_resume,
 		.set_dac_params = set_pcm1796_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_pcm1796_volume,
@@ -592,6 +624,8 @@ static const struct oxygen_model xonar_models[] = {
 		.control_filter = xonar_dx_control_filter,
 		.mixer_init = xonar_dx_mixer_init,
 		.cleanup = xonar_dx_cleanup,
+		.suspend = xonar_dx_cleanup,
+		.resume = xonar_dx_resume,
 		.set_dac_params = set_cs43xx_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_cs43xx_volume,
@@ -636,6 +670,10 @@ static struct pci_driver xonar_driver = {
 	.id_table = xonar_ids,
 	.probe = xonar_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+	.suspend = oxygen_pci_suspend,
+	.resume = oxygen_pci_resume,
+#endif
 };
 
 static int __init alsa_card_xonar_init(void)
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 7fdcdc8c6b64..2c7e25336795 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -516,7 +516,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 	int capture_mask = 0;
 	int playback_mask = 0;
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	struct timeval my_tv1, my_tv2;
 	do_gettimeofday(&my_tv1);
 #endif
@@ -623,7 +623,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 
 	mutex_unlock(&mgr->setup_mutex);
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	do_gettimeofday(&my_tv2);
 	snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n",
 		    (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 78aa81feaa4a..abe5c59b72df 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -473,7 +473,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = {
 [CMD_AUDIO_LEVEL_ADJUST] =		{ 0xc22000, 0, RMH_SSIZE_FIXED },
 };
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 static char* cmd_names[] = {
 [CMD_VERSION] =				"CMD_VERSION",
 [CMD_SUPPORTED] =			"CMD_SUPPORTED",
@@ -549,7 +549,7 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 				}
 			}
 		}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (rmh->cmd_idx < CMD_LAST_INDEX)
 			snd_printdd("    stat[%d]=%x\n", i, data);
 #endif
@@ -597,7 +597,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		data |= 0x008000;	/* MASK_MORE_THAN_1_WORD_COMMAND */
 	else
 		data &= 0xff7fff;	/* MASK_1_WORD_COMMAND */
-#ifdef CONFIG_SND_DEBUG_DETECT
+#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]);
 #endif
@@ -624,7 +624,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		for (i=1; i < rmh->cmd_len; i++) {
 			/* send other words */
 			data = rmh->cmd[i];
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 			if (rmh->cmd_idx < CMD_LAST_INDEX)
 				snd_printdd("    cmd[%d]=%x\n", i, data);
 #endif
@@ -847,7 +847,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
 	int state, i, err;
 	int audio_mask;
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	struct timeval my_tv1, my_tv2;
 	do_gettimeofday(&my_tv1);
 #endif
@@ -894,7 +894,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
 		if (err)
 			return err;
 	}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	do_gettimeofday(&my_tv2);
 	snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n",
 		    (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
@@ -951,7 +951,7 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err,
 				  enum pcxhr_async_err_src err_src, int pipe,
 				  int is_capture)
 {
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	static char* err_src_name[] = {
 		[PCXHR_ERR_PIPE]	= "Pipe",
 		[PCXHR_ERR_STREAM]	= "Stream",
@@ -1169,7 +1169,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 				    mgr->dsp_time_last, dsp_time_new);
 			mgr->dsp_time_err++;
 		}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#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))
@@ -1208,7 +1208,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 		mgr->src_it_dsp = reg;
 		tasklet_hi_schedule(&mgr->msg_taskq);
 	}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	if (reg & PCXHR_FATAL_DSP_ERR)
 		snd_printdd("FATAL DSP ERROR : %x\n", reg);
 #endif
diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig
index c9fa1a2bc58b..7fbb190adf6d 100644
--- a/sound/pcmcia/Kconfig
+++ b/sound/pcmcia/Kconfig
@@ -1,11 +1,16 @@
 # ALSA PCMCIA drivers
 
-menu "PCMCIA devices"
-	depends on SND!=n && PCMCIA
+menuconfig SND_PCMCIA
+	bool "PCMCIA sound devices"
+	depends on PCMCIA
+	default y
+	help
+	  Support for sound devices connected via the PCMCIA bus.
+
+if SND_PCMCIA && PCMCIA
 
 config SND_VXPOCKET
 	tristate "Digigram VXpocket"
-	depends on SND && PCMCIA
 	select SND_VX_LIB
 	help
 	  Say Y here to include support for Digigram VXpocket and
@@ -16,7 +21,6 @@ config SND_VXPOCKET
 
 config SND_PDAUDIOCF
 	tristate "Sound Core PDAudioCF"
-	depends on SND && PCMCIA
 	select SND_PCM
 	help
 	  Say Y here to include support for Sound Core PDAudioCF
@@ -25,4 +29,5 @@ config SND_PDAUDIOCF
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-pdaudiocf.
 
-endmenu
+endif	# SND_PCMCIA
+
diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig
index cacb0b136883..777de2b17178 100644
--- a/sound/ppc/Kconfig
+++ b/sound/ppc/Kconfig
@@ -1,17 +1,17 @@
 # ALSA PowerMac drivers
 
-menu "ALSA PowerMac devices"
-	depends on SND!=n && PPC
-
-comment "ALSA PowerMac requires I2C"
-	depends on SND && I2C=n
+menuconfig SND_PPC
+	bool "PowerPC sound devices"
+	depends on PPC64 || PPC32
+	default y
+	help
+	  Support for sound devices specific to PowerPC architectures.
 
-comment "ALSA PowerMac requires INPUT"
-	depends on SND && INPUT=n
+if SND_PPC
 
 config SND_POWERMAC
 	tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)"
-	depends on SND && I2C && INPUT && PPC_PMAC
+	depends on I2C && INPUT && PPC_PMAC
 	select SND_PCM
 	help
 	  Say Y here to include support for the integrated sound device.
@@ -32,14 +32,9 @@ config SND_POWERMAC_AUTO_DRC
 	  Note that you can turn on/off DRC manually even without this
 	  option.
 
-endmenu
-
-menu "ALSA PowerPC devices"
-	depends on SND!=n && ( PPC64 || PPC32 )
-
 config SND_PS3
 	tristate "PS3 Audio support"
-	depends on SND && PS3_PS3AV
+	depends on PS3_PS3AV
 	select SND_PCM
 	default m
 	help
@@ -52,4 +47,5 @@ config SND_PS3_DEFAULT_START_DELAY
 	int "Startup delay time in ms"
 	depends on SND_PS3
 	default "2000"
-endmenu
+
+endif	# SND_PPC
diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig
index b7e08ef22a94..cfc143985802 100644
--- a/sound/sh/Kconfig
+++ b/sound/sh/Kconfig
@@ -1,14 +1,22 @@
 # ALSA SH drivers
 
-menu "SUPERH devices"
-	depends on SND!=n && SUPERH
+menuconfig SND_SUPERH
+	bool "SUPERH sound devices"
+	depends on SUPERH
+	default y
+	help
+	  Support for sound devices specific to SUPERH architectures.
+	  Drivers that are implemented on ASoC can be found in
+	  "ALSA for SoC audio support" section.
+
+if SND_SUPERH
 
 config SND_AICA
 	tristate "Dreamcast Yamaha AICA sound"
-	depends on SH_DREAMCAST && SND
+	depends on SH_DREAMCAST
 	select SND_PCM
 	help
 	  ALSA Sound driver for the SEGA Dreamcast console.
 
-endmenu
+endif	# SND_SUPERH
 
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac4bfe8..fd7bc4f89072 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -2,15 +2,8 @@
 # SoC audio configuration
 #
 
-menu "System on Chip audio support"
-	depends on SND!=n
-
-config SND_SOC_AC97_BUS
-	bool
-
-config SND_SOC
+menuconfig SND_SOC
 	tristate "ALSA for SoC audio support"
-	depends on SND
 	select SND_PCM
 	---help---
 
@@ -23,6 +16,11 @@ config SND_SOC
 	  This ASoC audio support can also be built as a module.  If so, the module
 	  will be called snd-soc-core.
 
+if SND_SOC
+
+config SND_SOC_AC97_BUS
+	bool
+
 # All the supported Soc's
 source "sound/soc/at91/Kconfig"
 source "sound/soc/pxa/Kconfig"
@@ -35,4 +33,5 @@ source "sound/soc/omap/Kconfig"
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
 
-endmenu
+endif	# SND_SOC
+
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
index 5cb93fd3a407..905186502e00 100644
--- a/sound/soc/at91/Kconfig
+++ b/sound/soc/at91/Kconfig
@@ -1,6 +1,6 @@
 config SND_AT91_SOC
 	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
-	depends on ARCH_AT91 && SND_SOC
+	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
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index 1347dcf3f80b..4a383a4a0ff1 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -191,7 +191,7 @@ static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 
 	/* speaker connected to LHPOUT */
 	{"Ext Spk", NULL, "LHPOUT"},
@@ -199,9 +199,6 @@ static const char *intercon[][3] = {
 	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
 	{"MICIN", NULL, "Mic Bias"},
 	{"Mic Bias", NULL, "Int Mic"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 /*
@@ -209,20 +206,14 @@ static const char *intercon[][3] = {
  */
 static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i;
-
 	DBG("eti_b1_wm8731_init() called\n");
 
 	/* Add specific widgets */
-	for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) {
-		snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]);
-	}
+	snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
+				  ARRAY_SIZE(eti_b1_dapm_widgets));
 
 	/* Set up specific audio path interconnects */
-	for(i = 0; intercon[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-			intercon[i][1], intercon[i][2]);
-	}
+	snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
 
 	/* not connected */
 	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3903ab7dfa4a..d4a5fe42f6e0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,25 @@
 config SND_SOC_AC97_CODEC
 	tristate
-	depends on SND_SOC
+	select SND_AC97_CODEC
 
 config SND_SOC_WM8731
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM8750
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM8753
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM9712
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM9713
 	tristate
-	depends on SND_SOC
 
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate
-	depends on SND_SOC
 
 # Cirrus Logic CS4270 Codec Hardware Mute Support
 # Select if you have external muting circuitry attached to your CS4270.
@@ -43,4 +37,4 @@ config SND_SOC_CS4270_VD33_ERRATA
 
 config SND_SOC_TLV320AIC3X
 	tristate
-	depends on SND_SOC && I2C
+	depends on I2C
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2a1ffe396908..7bf2081b46fb 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -10,9 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    17th Oct 2005   Initial version.
- *
  * Generic AC97 support.
  */
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 09b1661b8a3a..dc8a38d9e53a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -49,7 +49,7 @@
 #include "tlv320aic3x.h"
 
 #define AUDIO_NAME "aic3x"
-#define AIC3X_VERSION "0.1"
+#define AIC3X_VERSION "0.2"
 
 /* codec private data */
 struct aic3x_priv {
@@ -138,6 +138,20 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg,
 		return -EIO;
 }
 
+/*
+ * read from the aic3x register space
+ */
+static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
+		      u8 *value)
+{
+	*value = reg & 0xff;
+	if (codec->hw_read(codec->control_data, value, 1) != 1)
+		return -EIO;
+
+	aic3x_write_reg_cache(codec, reg, *value);
+	return 0;
+}
+
 #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
@@ -483,7 +497,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("LINE2R"),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* Left Output */
 	{"Left DAC Mux", "DAC_L1", "Left DAC"},
 	{"Left DAC Mux", "DAC_L2", "Left DAC"},
@@ -627,102 +641,20 @@ static const char *intercon[][3] = {
 	{"Right Line Out", NULL, "Right Line2 Bypass Mixer"},
 	{"Mono Out", NULL, "Right Line2 Bypass Mixer"},
 	{"Right HP Out", NULL, "Right Line2 Bypass Mixer"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int aic3x_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* set up audio path interconnects */
-	for (i = 0; intercon[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-					   intercon[i][1], intercon[i][2]);
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
-struct aic3x_rate_divs {
-	u32 mclk;
-	u32 rate;
-	u32 fsref_reg;
-	u8 sr_reg:4;
-	u8 pllj_reg;
-	u16 plld_reg;
-};
-
-/* AIC3X codec mclk clock divider coefficients */
-static const struct aic3x_rate_divs aic3x_divs[] = {
-	/* 8k */
-	{12000000, 8000, 48000, 0xa, 16, 3840},
-	{19200000, 8000, 48000, 0xa, 10, 2400},
-	{22579200, 8000, 48000, 0xa, 8, 7075},
-	{33868800, 8000, 48000, 0xa, 5, 8049},
-	/* 11.025k */
-	{12000000, 11025, 44100, 0x6, 15, 528},
-	{19200000, 11025, 44100, 0x6, 9, 4080},
-	{22579200, 11025, 44100, 0x6, 8, 0},
-	{33868800, 11025, 44100, 0x6, 5, 3333},
-	/* 16k */
-	{12000000, 16000, 48000, 0x4, 16, 3840},
-	{19200000, 16000, 48000, 0x4, 10, 2400},
-	{22579200, 16000, 48000, 0x4, 8, 7075},
-	{33868800, 16000, 48000, 0x4, 5, 8049},
-	/* 22.05k */
-	{12000000, 22050, 44100, 0x2, 15, 528},
-	{19200000, 22050, 44100, 0x2, 9, 4080},
-	{22579200, 22050, 44100, 0x2, 8, 0},
-	{33868800, 22050, 44100, 0x2, 5, 3333},
-	/* 32k */
-	{12000000, 32000, 48000, 0x1, 16, 3840},
-	{19200000, 32000, 48000, 0x1, 10, 2400},
-	{22579200, 32000, 48000, 0x1, 8, 7075},
-	{33868800, 32000, 48000, 0x1, 5, 8049},
-	/* 44.1k */
-	{12000000, 44100, 44100, 0x0, 15, 528},
-	{19200000, 44100, 44100, 0x0, 9, 4080},
-	{22579200, 44100, 44100, 0x0, 8, 0},
-	{33868800, 44100, 44100, 0x0, 5, 3333},
-	/* 48k */
-	{12000000, 48000, 48000, 0x0, 16, 3840},
-	{19200000, 48000, 48000, 0x0, 10, 2400},
-	{22579200, 48000, 48000, 0x0, 8, 7075},
-	{33868800, 48000, 48000, 0x0, 5, 8049},
-	/* 64k */
-	{12000000, 64000, 96000, 0x1, 16, 3840},
-	{19200000, 64000, 96000, 0x1, 10, 2400},
-	{22579200, 64000, 96000, 0x1, 8, 7075},
-	{33868800, 64000, 96000, 0x1, 5, 8049},
-	/* 88.2k */
-	{12000000, 88200, 88200, 0x0, 15, 528},
-	{19200000, 88200, 88200, 0x0, 9, 4080},
-	{22579200, 88200, 88200, 0x0, 8, 0},
-	{33868800, 88200, 88200, 0x0, 5, 3333},
-	/* 96k */
-	{12000000, 96000, 96000, 0x0, 16, 3840},
-	{19200000, 96000, 96000, 0x0, 10, 2400},
-	{22579200, 96000, 96000, 0x0, 8, 7075},
-	{33868800, 96000, 96000, 0x0, 5, 8049},
-};
-
-static inline int aic3x_get_divs(int mclk, int rate)
-{
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) {
-		if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk)
-			return i;
-	}
-
-	return 0;
-}
-
 static int aic3x_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params)
 {
@@ -730,49 +662,107 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
-	int i;
-	u8 data, pll_p, pll_r, pll_j;
-	u16 pll_d;
-
-	i = aic3x_get_divs(aic3x->sysclk, params_rate(params));
+	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
+	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
+	u16 pll_d = 1;
 
-	/* Route Left DAC to left channel input and
-	 * right DAC to right channel input */
-	data = (LDAC2LCH | RDAC2RCH);
-	switch (aic3x_divs[i].fsref_reg) {
-	case 44100:
-		data |= FSREF_44100;
+	/* select data word length */
+	data =
+	    aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
-	case 48000:
-		data |= FSREF_48000;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		data |= (0x01 << 4);
 		break;
-	case 88200:
-		data |= FSREF_44100 | DUAL_RATE_MODE;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		data |= (0x02 << 4);
 		break;
-	case 96000:
-		data |= FSREF_48000 | DUAL_RATE_MODE;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		data |= (0x03 << 4);
 		break;
 	}
+	aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
+
+	/* Fsref can be 44100 or 48000 */
+	fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000;
+
+	/* Try to find a value for Q which allows us to bypass the PLL and
+	 * generate CODEC_CLK directly. */
+	for (pll_q = 2; pll_q < 18; pll_q++)
+		if (aic3x->sysclk / (128 * pll_q) == fsref) {
+			bypass_pll = 1;
+			break;
+		}
+
+	if (bypass_pll) {
+		pll_q &= 0xf;
+		aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
+		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
+	} else
+		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+
+	/* Route Left DAC to left channel input and
+	 * right DAC to right channel input */
+	data = (LDAC2LCH | RDAC2RCH);
+	data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
+	if (params_rate(params) >= 64000)
+		data |= DUAL_RATE_MODE;
 	aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
 
 	/* codec sample rate select */
-	data = aic3x_divs[i].sr_reg;
+	data = (fsref * 20) / params_rate(params);
+	if (params_rate(params) < 64000)
+		data /= 2;
+	data /= 5;
+	data -= 2;
 	data |= (data << 4);
 	aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data);
 
-	/* Use PLL for generation Fsref by equation:
-	 * Fsref = (MCLK * K * R)/(2048 * P);
-	 * Fix P = 2 and R = 1 and calculate K, if
-	 * K = J.D, i.e. J - an interger portion of K and D is the fractional
-	 * one with 4 digits of precision;
-	 * Example:
-	 * For MCLK = 22.5792 MHz and Fsref = 48kHz:
-	 * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074
+	if (bypass_pll)
+		return 0;
+
+	/* Use PLL
+	 * find an apropriate setup for j, d, r and p by iterating over
+	 * p and r - j and d are calculated for each fraction.
+	 * Up to 128 values are probed, the closest one wins the game.
+	 * The sysclk is divided by 1000 to prevent integer overflows.
 	 */
-	pll_p = 2;
-	pll_r = 1;
-	pll_j = aic3x_divs[i].pllj_reg;
-	pll_d = aic3x_divs[i].plld_reg;
+	codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000);
+
+	for (r = 1; r <= 16; r++)
+		for (p = 1; p <= 8; p++) {
+			int clk, tmp = (codec_clk * pll_r * 10) / pll_p;
+			u8 j = tmp / 10000;
+			u16 d = tmp % 10000;
+
+			if (j > 63)
+				continue;
+
+			if (d != 0 && aic3x->sysclk < 10000000)
+				continue;
+
+			/* This is actually 1000 * ((j + (d/10000)) * r) / p
+			 * The term had to be converted to get rid of the
+			 * division by 10000 */
+			clk = ((10000 * j * r) + (d * r)) / (10 * p);
+
+			/* check whether this values get closer than the best
+			 * ones we had before */
+			if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
+				pll_j = j; pll_d = d; pll_r = r; pll_p = p;
+				last_clk = clk;
+			}
+
+			/* Early exit for exact matches */
+			if (clk == codec_clk)
+				break;
+		}
+
+	if (last_clk == 0) {
+		printk(KERN_ERR "%s(): unable to setup PLL\n", __func__);
+		return -EINVAL;
+	}
 
 	data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
 	aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT));
@@ -782,24 +772,6 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	aic3x_write(codec, AIC3X_PLL_PROGD_REG,
 		    (pll_d & 0x3F) << PLLD_LSB_SHIFT);
 
-	/* select data word length */
-	data =
-	    aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-		data |= (0x01 << 4);
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		data |= (0x02 << 4);
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		data |= (0x03 << 4);
-		break;
-	}
-	aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
-
 	return 0;
 }
 
@@ -826,16 +798,8 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 
-	switch (freq) {
-	case 12000000:
-	case 19200000:
-	case 22579200:
-	case 33868800:
-		aic3x->sysclk = freq;
-		return 0;
-	}
-
-	return -EINVAL;
+	aic3x->sysclk = freq;
+	return 0;
 }
 
 static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
@@ -883,13 +847,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
+static int aic3x_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
 {
 	struct aic3x_priv *aic3x = codec->private_data;
 	u8 reg;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0:
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* all power is driven by DAPM system */
 		if (aic3x->master) {
 			/* enable pll */
@@ -898,10 +863,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
 				    reg | PLL_ENABLE);
 		}
 		break;
-	case SNDRV_CTL_POWER_D1:
-	case SNDRV_CTL_POWER_D2:
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot:
+	case SND_SOC_BIAS_STANDBY:
 		/*
 		 * all power is driven by DAPM system,
 		 * so output power is safe if bypass was set
@@ -913,7 +877,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
 				    reg & ~PLL_ENABLE);
 		}
 		break;
-	case SNDRV_CTL_POWER_D3cold:
+	case SND_SOC_BIAS_OFF:
 		/* force all power off */
 		reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL);
 		aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON);
@@ -949,11 +913,38 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
 		}
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 
 	return 0;
 }
 
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state)
+{
+	u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+	u8 bit = gpio ? 3: 0;
+	u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit);
+	aic3x_write(codec, reg, val | (!!state << bit));
+}
+EXPORT_SYMBOL_GPL(aic3x_set_gpio);
+
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
+{
+	u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+	u8 val, bit = gpio ? 2: 1;
+
+	aic3x_read(codec, reg, &val);
+	return (val >> bit) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_get_gpio);
+
+int aic3x_headset_detected(struct snd_soc_codec *codec)
+{
+	u8 val;
+	aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
+	return (val >> 2) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_headset_detected);
+
 #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)
@@ -988,7 +979,7 @@ static int aic3x_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;
 
-	aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -1008,7 +999,7 @@ static int aic3x_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	aic3x_dapm_event(codec, codec->suspend_dapm_state);
+	aic3x_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
 }
@@ -1020,13 +1011,14 @@ static int aic3x_resume(struct platform_device *pdev)
 static int aic3x_init(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
 	codec->name = "aic3x";
 	codec->owner = THIS_MODULE;
 	codec->read = aic3x_read_reg_cache;
 	codec->write = aic3x_write;
-	codec->dapm_event = aic3x_dapm_event;
+	codec->set_bias_level = aic3x_set_bias_level;
 	codec->dai = &aic3x_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = sizeof(aic3x_reg);
@@ -1108,7 +1100,11 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
 
 	/* off, with power on */
-	aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* setup GPIO functions */
+	aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
+	aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
 
 	aic3x_add_controls(codec);
 	aic3x_add_widgets(codec);
@@ -1217,6 +1213,12 @@ static struct i2c_client client_template = {
 	.name = "AIC3X",
 	.driver = &aic3x_i2c_driver,
 };
+
+static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len)
+{
+	value[0] = i2c_smbus_read_byte_data(client, value[0]);
+	return (len == 1);
+}
 #endif
 
 static int aic3x_probe(struct platform_device *pdev)
@@ -1251,6 +1253,7 @@ static int aic3x_probe(struct platform_device *pdev)
 	if (setup->i2c_address) {
 		normal_i2c[0] = setup->i2c_address;
 		codec->hw_write = (hw_write_t) i2c_master_send;
+		codec->hw_read = (hw_read_t) aic3x_i2c_read;
 		ret = i2c_add_driver(&aic3x_i2c_driver);
 		if (ret != 0)
 			printk(KERN_ERR "can't add i2c driver");
@@ -1268,7 +1271,7 @@ static int aic3x_remove(struct platform_device *pdev)
 
 	/* power down chip */
 	if (codec->control_data)
-		aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3);
+		aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index d0cdeeb629de..c1dd1ac0ceac 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -108,6 +108,13 @@
 #define DACR1_2_RLOPM_VOL		92
 #define LLOPM_CTRL			86
 #define RLOPM_CTRL			93
+/* GPIO/IRQ registers */
+#define AIC3X_STICKY_IRQ_FLAGS_REG	96
+#define AIC3X_RT_IRQ_FLAGS_REG		97
+#define AIC3X_GPIO1_REG			98
+#define AIC3X_GPIO2_REG			99
+#define AIC3X_GPIOA_REG			100
+#define AIC3X_GPIOB_REG			101
 /* Clock generation control register */
 #define AIC3X_CLKGEN_CTRL_REG		102
 
@@ -128,12 +135,15 @@
 
 /* PLL registers bitfields */
 #define PLLP_SHIFT		0
+#define PLLQ_SHIFT		3
 #define PLLR_SHIFT		0
 #define PLLJ_SHIFT		2
 #define PLLD_MSB_SHIFT		0
 #define PLLD_LSB_SHIFT		2
 
 /* Clock generation register bits */
+#define CODEC_CLKIN_PLLDIV	0
+#define CODEC_CLKIN_CLKDIV	1
 #define PLL_CLKIN_SHIFT		4
 #define MCLK_SOURCE		0x0
 #define PLL_CLKDIV_SHIFT	0
@@ -171,8 +181,49 @@
 /* Default input volume */
 #define DEFAULT_GAIN    0x20
 
+/* GPIO API */
+enum {
+	AIC3X_GPIO1_FUNC_DISABLED		= 0,
+	AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC	= 1,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX		= 2,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2		= 3,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4		= 4,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8		= 5,
+	AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ	= 6,
+	AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ		= 7,
+	AIC3X_GPIO1_FUNC_INPUT			= 8,
+	AIC3X_GPIO1_FUNC_OUTPUT			= 9,
+	AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK	= 10,
+	AIC3X_GPIO1_FUNC_AUDIO_WORDCLK		= 11,
+	AIC3X_GPIO1_FUNC_BUTTON_IRQ		= 12,
+	AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ	= 13,
+	AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ	= 14,
+	AIC3X_GPIO1_FUNC_ALL_IRQ		= 16
+};
+
+enum {
+	AIC3X_GPIO2_FUNC_DISABLED		= 0,
+	AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ	= 2,
+	AIC3X_GPIO2_FUNC_INPUT			= 3,
+	AIC3X_GPIO2_FUNC_OUTPUT			= 4,
+	AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT	= 5,
+	AIC3X_GPIO2_FUNC_AUDIO_BITCLK		= 8,
+	AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9,
+	AIC3X_GPIO2_FUNC_ALL_IRQ		= 10,
+	AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11,
+	AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12,
+	AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ	= 13,
+	AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ		= 14,
+	AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ	= 15
+};
+
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+int aic3x_headset_detected(struct snd_soc_codec *codec);
+
 struct aic3x_setup_data {
 	unsigned short i2c_address;
+	unsigned int gpio_func[2];
 };
 
 extern struct snd_soc_codec_dai aic3x_dai;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0cf9265fca8f..5acf43ab104e 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -193,7 +193,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"),
 SND_SOC_DAPM_INPUT("LLINEIN"),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
 	{"Output Mixer", "HiFi Playback Switch", "DAC"},
@@ -214,22 +214,14 @@ static const char *intercon[][3] = {
 	{"Line Input", NULL, "LLINEIN"},
 	{"Line Input", NULL, "RLINEIN"},
 	{"Mic Bias", NULL, "MICIN"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8731_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
-	/* set up audio path interconnects */
-	for (i = 0; intercon[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-			intercon[i][1], intercon[i][2]);
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -435,29 +427,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8731_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* vref/mid, osc on, dac unmute */
 		wm8731_write(codec, WM8731_PWR, reg);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* everything off except vref/vmid, */
 		wm8731_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		/* everything off, dac mute, inactive */
 		wm8731_write(codec, WM8731_ACTIVE, 0x0);
 		wm8731_write(codec, WM8731_PWR, 0xffff);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -503,7 +495,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	wm8731_write(codec, WM8731_ACTIVE, 0x0);
-	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -521,8 +513,8 @@ static int wm8731_resume(struct platform_device *pdev)
 		data[1] = cache[i] & 0x00ff;
 		codec->hw_write(codec->control_data, data, 2);
 	}
-	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-	wm8731_dapm_event(codec, codec->suspend_dapm_state);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	wm8731_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
@@ -539,7 +531,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8731_read_reg_cache;
 	codec->write = wm8731_write;
-	codec->dapm_event = wm8731_dapm_event;
+	codec->set_bias_level = wm8731_set_bias_level;
 	codec->dai = &wm8731_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = sizeof(wm8731_reg);
@@ -557,7 +549,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
 	}
 
 	/* power on device */
-	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* set the update bits */
 	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
@@ -730,7 +722,7 @@ static int wm8731_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 16cd5d4d5ad9..1f11ad24551a 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -378,7 +378,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("RINPUT3"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left mixer */
 	{"Left Mixer", "Playback Switch", "Left DAC"},
 	{"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
@@ -470,22 +470,14 @@ static const char *audio_map[][3] = {
 	/* ADC */
 	{"Left ADC", NULL, "Left ADC Mux"},
 	{"Right ADC", NULL, "Right ADC Mux"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8750_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+				  ARRAY_SIZE(wm8750_dapm_widgets));
 
-	/* set up audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -686,29 +678,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8750_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* set vmid to 50k and unmute dac */
 		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		/* set vmid to 5k for quick power up */
 		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* mute dac and set vmid to 500k, enable VREF */
 		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		wm8750_write(codec, WM8750_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -748,7 +740,7 @@ static void wm8750_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec =
 		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8750_dapm_event(codec, codec->dapm_state);
+	wm8750_set_bias_level(codec, codec->bias_level);
 }
 
 static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
@@ -756,7 +748,7 @@ static int wm8750_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;
 
-	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -777,12 +769,12 @@ static int wm8750_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8750 caps */
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-		wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
-		codec->dapm_state = SNDRV_CTL_POWER_D0;
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+		wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+		codec->bias_level = SND_SOC_BIAS_ON;
 		schedule_delayed_work(&codec->delayed_work,
 					msecs_to_jiffies(1000));
 	}
@@ -803,7 +795,7 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8750_read_reg_cache;
 	codec->write = wm8750_write;
-	codec->dapm_event = wm8750_dapm_event;
+	codec->set_bias_level = wm8750_set_bias_level;
 	codec->dai = &wm8750_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = sizeof(wm8750_reg);
@@ -821,8 +813,8 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	}
 
 	/* charge output caps */
-	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
-	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
 
 	/* set the update bits */
@@ -1021,7 +1013,7 @@ static int wm8750_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index fb41826c4c4c..c32e6326be6c 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -523,7 +523,7 @@ SND_SOC_DAPM_INPUT("MIC2"),
 SND_SOC_DAPM_VMID("VREF"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left mixer */
 	{"Left Mixer", "Left Playback Switch", "Left DAC"},
 	{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
@@ -674,23 +674,14 @@ static const char *audio_map[][3] = {
 
 	/* ACOP */
 	{"ACOP", NULL, "ALC Mixer"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8753_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
 
-	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-
-	/* set up the WM8753 audio map */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -1274,29 +1265,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8753_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* set vmid to 50k and unmute dac */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		/* set vmid to 5k for quick power up */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* mute dac and set vmid to 500k, enable VREF */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		wm8753_write(codec, WM8753_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -1500,7 +1491,7 @@ static void wm8753_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec =
 		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8753_dapm_event(codec, codec->dapm_state);
+	wm8753_set_bias_level(codec, codec->bias_level);
 }
 
 static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
@@ -1512,7 +1503,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 	if (!codec->card)
 		return 0;
 
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -1537,12 +1528,12 @@ static int wm8753_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8753 caps */
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-		codec->dapm_state = SNDRV_CTL_POWER_D0;
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+		wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+		codec->bias_level = SND_SOC_BIAS_ON;
 		schedule_delayed_work(&codec->delayed_work,
 			msecs_to_jiffies(caps_charge));
 	}
@@ -1563,7 +1554,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8753_read_reg_cache;
 	codec->write = wm8753_write;
-	codec->dapm_event = wm8753_dapm_event;
+	codec->set_bias_level = wm8753_set_bias_level;
 	codec->dai = wm8753_dai;
 	codec->num_dai = 2;
 	codec->reg_cache_size = sizeof(wm8753_reg);
@@ -1584,8 +1575,8 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	}
 
 	/* charge output caps */
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work,
 		msecs_to_jiffies(caps_charge));
 
@@ -1792,7 +1783,7 @@ static int wm8753_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 76c1e2d33e7d..d9789f1c8903 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -9,9 +9,6 @@
  *  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
- *    4th Feb 2006   Initial version.
  */
 
 #include <linux/init.h>
@@ -351,7 +348,7 @@ SND_SOC_DAPM_INPUT("MIC1"),
 SND_SOC_DAPM_INPUT("MIC2"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* virtual mixer - mixes left & right channels for spk and mono */
 	{"AC97 Mixer", NULL, "Left DAC"},
 	{"AC97 Mixer", NULL, "Right DAC"},
@@ -446,21 +443,14 @@ static const char *audio_map[][3] = {
 	{"Speaker PGA", NULL, "Speaker Mux"},
 	{"LOUT2", NULL, "Speaker PGA"},
 	{"ROUT2", NULL, "Speaker PGA"},
-
-	{NULL, NULL, NULL},
 };
 
 static int wm9712_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
+				  ARRAY_SIZE(wm9712_dapm_widgets));
 
-	/* set up audio path connects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-					   audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -574,23 +564,23 @@ struct snd_soc_codec_dai wm9712_dai[] = {
 };
 EXPORT_SYMBOL_GPL(wm9712_dai);
 
-static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9712_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		ac97_write(codec, AC97_POWERDOWN, 0x0000);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		/* disable everything including AC link */
 		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -618,7 +608,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
 
-	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -635,7 +625,7 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 		return ret;
 	}
 
-	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	if (ret == 0) {
 		/* Sync reg_cache with the hardware after cold reset */
@@ -647,8 +637,8 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 		}
 	}
 
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
-		wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm9712_set_bias_level(codec, SND_SOC_BIAS_ON);
 
 	return ret;
 }
@@ -682,7 +672,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	codec->num_dai = ARRAY_SIZE(wm9712_dai);
 	codec->write = ac97_write;
 	codec->read = ac97_read;
-	codec->dapm_event = wm9712_dapm_event;
+	codec->set_bias_level = wm9712_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -706,7 +696,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	/* set alc mux to none */
 	ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
 
-	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm9712_add_controls(codec);
 	wm9712_add_widgets(codec);
 	ret = snd_soc_register_card(socdev);
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 1f241161445c..4f516a5a5616 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -10,9 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    4th Feb 2006   Initial version.
- *
  *  Features:-
  *
  *   o Support for AC97 Codec, Voice DAC and Aux DAC
@@ -456,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"),
 SND_SOC_DAPM_VMID("VMID"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left HP mixer */
 	{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
 	{"Left HP Mixer", "Voice Playback Switch",   "Voice DAC"},
@@ -607,21 +604,14 @@ static const char *audio_map[][3] = {
 	{"Capture Mono Mux", "Stereo", "Capture Mixer"},
 	{"Capture Mono Mux", "Left", "Left Capture Source"},
 	{"Capture Mono Mux", "Right", "Right Capture Source"},
-
-	{NULL, NULL, NULL},
 };
 
 static int wm9713_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+				  ARRAY_SIZE(wm9713_dapm_widgets));
 
-	/* set up audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -1097,33 +1087,33 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
 }
 EXPORT_SYMBOL_GPL(wm9713_reset);
 
-static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9713_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 reg;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* enable thermal shutdown */
 		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
 		ac97_write(codec, AC97_EXTENDED_MID, reg);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* enable master bias and vmid */
 		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
 		ac97_write(codec, AC97_EXTENDED_MID, reg);
 		ac97_write(codec, AC97_POWERDOWN, 0x0000);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		/* disable everything including AC link */
 		ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
 		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -1160,7 +1150,7 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 		return ret;
 	}
 
-	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* do we need to re-start the PLL ? */
 	if (wm9713->pll_out)
@@ -1176,8 +1166,8 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 		}
 	}
 
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
-		wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
 
 	return ret;
 }
@@ -1216,7 +1206,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	codec->num_dai = ARRAY_SIZE(wm9713_dai);
 	codec->write = ac97_write;
 	codec->read = ac97_read;
-	codec->dapm_event = wm9713_dapm_event;
+	codec->set_bias_level = wm9713_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -1238,7 +1228,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 		goto reset_err;
 	}
 
-	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* unmute the adc - move to kcontrol */
 	reg = ac97_read(codec, AC97_CD) & 0x7fff;
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 20680c551aab..8f7e33834902 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,6 +1,6 @@
 config SND_DAVINCI_SOC
 	tristate "SoC Audio for the TI DAVINCI chip"
-	depends on ARCH_DAVINCI && SND_SOC
+	depends on ARCH_DAVINCI
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the DAVINCI AC97 or I2S interface. You will also need
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index fcd165240333..4c70a0ed3397 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -71,7 +71,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 };
 
 /* davinci-evm machine audio_mapnections to the codec pins */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* Headphone connected to HPLOUT, HPROUT */
 	{"Headphone Jack", NULL, "HPLOUT"},
 	{"Headphone Jack", NULL, "HPROUT"},
@@ -90,23 +90,17 @@ static const char *audio_map[][3] = {
 	{"LINE2L", NULL, "Line In"},
 	{"LINE1R", NULL, "Line In"},
 	{"LINE2R", NULL, "Line In"},
-
-	{NULL, NULL, NULL},
 };
 
 /* Logic for a aic3x as connected on a davinci-evm */
 static int evm_aic3x_init(struct snd_soc_codec *codec)
 {
-	int i;
-
 	/* Add davinci-evm specific widgets */
-	for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* Set up davinci-evm specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-					   audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	/* not connected */
 	snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 257101f44e9e..19802e27df4b 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -2,7 +2,7 @@ menu "ALSA SoC audio for Freescale SOCs"
 
 config SND_SOC_MPC8610
 	bool "ALSA SoC support for the MPC8610 SOC"
-	depends on SND_SOC && MPC8610_HPCD
+	depends on MPC8610_HPCD
 	default y if MPC8610
 	help
 	  Say Y if you want to add support for codecs attached to the SSI
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6533563a6011..c32487b6c7b0 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -30,15 +30,15 @@
 
 #include <asm/mach-types.h>
 #include <asm/arch/hardware.h>
-#include <asm/arch/gpio.h>
+#include <linux/gpio.h>
 #include <asm/arch/mcbsp.h>
 
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 #include "../codecs/tlv320aic3x.h"
 
-#define RX44_HEADSET_AMP_GPIO	10
-#define RX44_SPEAKER_AMP_GPIO	101
+#define N810_HEADSET_AMP_GPIO	10
+#define N810_SPEAKER_AMP_GPIO	101
 
 static struct clk *sys_clkout2;
 static struct clk *sys_clkout2_src;
@@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w,
 			  struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
-		omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1);
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
 	else
-		omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0);
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
 
 	return 0;
 }
@@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w,
 			   struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
-		omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1);
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
 	else
-		omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0);
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
 
 	return 0;
 }
@@ -177,7 +177,7 @@ static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	{"Headphone Jack", NULL, "HPLOUT"},
 	{"Headphone Jack", NULL, "HPROUT"},
 
@@ -217,13 +217,11 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add N810 specific widgets */
-	for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
+				  ARRAY_SIZE(aic33_dapm_widgets));
 
 	/* Set up N810 specific audio path audio_map */
-	for (i = 0; i < ARRAY_SIZE(audio_map); i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_sync_endpoints(codec);
 
@@ -305,12 +303,12 @@ static int __init n810_soc_init(void)
 	clk_set_parent(sys_clkout2_src, func96m_clk);
 	clk_set_rate(sys_clkout2, 12000000);
 
-	if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0)
+	if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
 		BUG();
-	if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0)
+	if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
 		BUG();
-	omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0);
-	omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0);
+	gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
+	gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
 
 	return 0;
 err2:
@@ -325,6 +323,9 @@ err1:
 
 static void __exit n810_soc_exit(void)
 {
+	gpio_free(N810_SPEAKER_AMP_GPIO);
+	gpio_free(N810_HEADSET_AMP_GPIO);
+
 	platform_device_unregister(n810_snd_device);
 }
 
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 484f883459e0..12f6ac99b04c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,6 +1,6 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
-	depends on ARCH_PXA && SND_SOC
+	depends on ARCH_PXA
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the PXA2xx AC97, I2S or SSP interface. You will also need
@@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800
 	help
 	  Say Y if you want to add support for SoC audio on the
 	  Toshiba e800 PDA
+
+config SND_PXA2XX_SOC_EM_X270
+	tristate "SoC Audio support for CompuLab EM-x270"
+	depends on SND_PXA2XX_SOC && MACH_EM_X270
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  CompuLab EM-x270.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 04e5646f75ba..5bc8edf9dca9 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o
 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
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
 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
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 7f32a1167572..edeea63e80e8 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -11,10 +11,6 @@
  *  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
- *    30th Nov 2005   Initial version.
- *
  */
 
 #include <linux/module.h>
@@ -247,7 +243,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL),
 };
 
 /* Corgi machine audio map (connections to the codec pins) */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headset Jack  - in = micin, out = LHPOUT*/
 	{"Headset Jack", NULL, "LHPOUT"},
@@ -265,8 +261,6 @@ static const char *audio_map[][3] = {
 
 	/* Same as the above but no mic bias for line signals */
 	{"MICIN", NULL, "Line Jack"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -303,13 +297,11 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add corgi specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
 	/* Set up corgi specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_sync_endpoints(codec);
 	return 0;
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
new file mode 100644
index 000000000000..02dcac39cdf6
--- /dev/null
+++ b/sound/soc/pxa/em-x270.c
@@ -0,0 +1,102 @@
+/*
+ * em-x270.c  --  SoC audio for EM-X270
+ *
+ * Copyright 2007 CompuLab, Ltd.
+ *
+ * Author: Mike Rapoport <mike@compulab.co.il>
+ *
+ * Copied from tosa.c:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.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 <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_dai_link em_x270_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	},
+};
+
+static struct snd_soc_machine em_x270 = {
+	.name = "EM-X270",
+	.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,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *em_x270_snd_device;
+
+static int __init em_x270_init(void)
+{
+	int ret;
+
+	if (!machine_is_em_x270())
+		return -ENODEV;
+
+	em_x270_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!em_x270_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata);
+	em_x270_snd_devdata.dev = &em_x270_snd_device->dev;
+	ret = platform_device_add(em_x270_snd_device);
+
+	if (ret)
+		platform_device_put(em_x270_snd_device);
+
+	return ret;
+}
+
+static void __exit em_x270_exit(void)
+{
+	platform_device_unregister(em_x270_snd_device);
+}
+
+module_init(em_x270_init);
+module_exit(em_x270_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mike Rapoport");
+MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 7e830b218943..810f1fe158ab 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -215,8 +215,8 @@ SND_SOC_DAPM_HP("Headphone Jack", NULL),
 SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
 };
 
-/* Corgi machine audio_mapnections to the codec pins */
-static const char *audio_map[][3] = {
+/* Corgi machine connections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headphone connected to LHPOUT1, RHPOUT1 */
 	{"Headphone Jack", NULL, "LHPOUT"},
@@ -225,8 +225,6 @@ static const char *audio_map[][3] = {
 	/* speaker connected to LOUT, ROUT */
 	{"Ext Spk", NULL, "ROUT"},
 	{"Ext Spk", NULL, "LOUT"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Off", "Headphone"};
@@ -263,13 +261,11 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add poodle specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
 	/* Set up poodle specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_sync_endpoints(codec);
 	return 0;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 425071030970..35090c2870ff 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -9,9 +9,6 @@
  *  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
- *    12th Aug 2005   Initial version.
  */
 
 #include <linux/init.h>
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d8b8372db00e..092b5c776b40 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -12,9 +12,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    30th Nov 2005   Initial version.
- *
  */
 
 #include <linux/module.h>
@@ -250,7 +247,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 };
 
 /* Spitz machine audio_map */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headphone connected to LOUT1, ROUT1 */
 	{"Headphone Jack", NULL, "LOUT1"},
@@ -269,8 +266,6 @@ static const char *audio_map[][3] = {
 
 	/* line is connected to input 1 - no bias */
 	{"LINPUT1", NULL, "Line Jack"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -313,13 +308,11 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add spitz specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+				  ARRAY_SIZE(wm8750_dapm_widgets));
 
-	/* Set up spitz specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	/* Set up spitz specific audio paths */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_sync_endpoints(codec);
 	return 0;
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 7346d7e5d066..465ff0f458ef 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -12,9 +12,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    30th Nov 2005   Initial version.
- *
  * GPIO's
  *  1 - Jack Insertion
  *  5 - Hookswitch (headset answer/hang up switch)
@@ -154,7 +151,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL),
 };
 
 /* tosa audio map */
-static const char *audio_map[][3] = {
+static const snd_soc_dapm_route audio_map[] = {
 
 	/* headphone connected to HPOUTL, HPOUTR */
 	{"Headphone Jack", NULL, "HPOUTL"},
@@ -173,8 +170,6 @@ static const char *audio_map[][3] = {
 	{"Headset Jack", NULL, "HPOUTR"},
 	{"LINEINR", NULL, "Mic Bias"},
 	{"Mic Bias", NULL, "Headset Jack"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -208,15 +203,11 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
 	}
 
 	/* add tosa specific widgets */
-	for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
-		snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
-	}
+	snd_soc_dapm_new_controls(codec, &tosa_dapm_widgets,
+				  ARRAY_SIZE(tosa_dapm_widgets));
 
 	/* set up tosa specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_sync_endpoints(codec);
 	return 0;
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 1f6dbfc4caa8..b9f2353effeb 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,7 +1,6 @@
 config SND_S3C24XX_SOC
 	tristate "SoC Audio for the Samsung S3C24XX chips"
-	depends on ARCH_S3C2410 && SND_SOC
-	select SND_PCM
+	depends on ARCH_S3C2410
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the S3C24XX AC97, I2S or SSP interface. You will also need
@@ -16,7 +15,6 @@ config SND_S3C2412_SOC_I2S
 config SND_S3C2443_SOC_AC97
 	tristate
 	select AC97_BUS
-	select SND_AC97_CODEC
 	select SND_SOC_AC97_BUS
 	
 config SND_S3C24XX_SOC_NEO1973_WM8753
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 0e9d1c5f2484..c1a0161bc72e 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -10,10 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    20th Jan 2007   Initial version.
- *    05th Feb 2007   Rename all to Neo1973
- *
  */
 
 #include <linux/module.h>
@@ -43,6 +39,14 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
 
+/* Debugging stuff */
+#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0
+#if S3C24XX_SOC_NEO1973_WM8753_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x)
+#else
+#define DBG(x...)
+#endif
+
 /* define the scenarios */
 #define NEO_AUDIO_OFF			0
 #define NEO_GSM_CALL_AUDIO_HANDSET	1
@@ -67,6 +71,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
+	DBG("Entered %s\n", __func__);
+
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	switch (params_rate(params)) {
@@ -151,6 +157,8 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
 
+	DBG("Entered %s\n", __func__);
+
 	/* disable the PLL */
 	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
 }
@@ -172,6 +180,8 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
+	DBG("Entered %s\n", __func__);
+
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	if (params_rate(params) != 8000)
@@ -213,6 +223,8 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
 
+	DBG("Entered %s\n", __func__);
+
 	/* disable the PLL */
 	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
 }
@@ -233,6 +245,8 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
 
 static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
 {
+	DBG("Entered %s\n", __func__);
+
 	switch (neo1973_scenario) {
 	case NEO_AUDIO_OFF:
 		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
@@ -315,6 +329,8 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 
+	DBG("Entered %s\n", __func__);
+
 	if (neo1973_scenario == ucontrol->value.integer.value[0])
 		return 0;
 
@@ -327,6 +343,8 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
 
 static void lm4857_write_regs(void)
 {
+	DBG("Entered %s\n", __func__);
+
 	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
 		printk(KERN_ERR "lm4857: i2c write failed\n");
 }
@@ -338,6 +356,8 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
 	int shift = (kcontrol->private_value >> 8) & 0x0F;
 	int mask = (kcontrol->private_value >> 16) & 0xFF;
 
+	DBG("Entered %s\n", __func__);
+
 	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
 	return 0;
 }
@@ -364,6 +384,8 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
 
+	DBG("Entered %s\n", __func__);
+
 	if (value)
 		value -= 5;
 
@@ -376,6 +398,8 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = ucontrol->value.integer.value[0];
 
+	DBG("Entered %s\n", __func__);
+
 	if (value)
 		value += 5;
 
@@ -397,8 +421,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
 };
 
 
-/* example machine audio_mapnections */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route dapm_routes[] = {
 
 	/* Connections to the lm4857 amp */
 	{"Audio Out", NULL, "LOUT1"},
@@ -421,8 +444,6 @@ static const char *audio_map[][3] = {
 
 	/* Connect the ALC pins */
 	{"ACIN", NULL, "ACOP"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *lm4857_mode[] = {
@@ -483,6 +504,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
+	DBG("Entered %s\n", __func__);
+
 	/* set up NC codec pins */
 	snd_soc_dapm_set_endpoint(codec, "LOUT2", 0);
 	snd_soc_dapm_set_endpoint(codec, "ROUT2", 0);
@@ -496,8 +519,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
 
 	/* Add neo1973 specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
 
 	/* add neo1973 specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
@@ -508,11 +531,9 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 			return err;
 	}
 
-	/* set up neo1973 specific audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	/* set up neo1973 specific audio routes */
+	err = snd_soc_dapm_add_routes(codec, dapm_routes,
+				      ARRAY_SIZE(dapm_routes));
 
 	snd_soc_dapm_sync_endpoints(codec);
 	return 0;
@@ -583,6 +604,8 @@ static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind)
 {
 	int ret;
 
+	DBG("Entered %s\n", __func__);
+
 	client_template.adapter = adap;
 	client_template.addr = addr;
 
@@ -606,6 +629,8 @@ exit_err:
 
 static int lm4857_i2c_detach(struct i2c_client *client)
 {
+	DBG("Entered %s\n", __func__);
+
 	i2c_detach_client(client);
 	kfree(client);
 	return 0;
@@ -613,6 +638,8 @@ static int lm4857_i2c_detach(struct i2c_client *client)
 
 static int lm4857_i2c_attach(struct i2c_adapter *adap)
 {
+	DBG("Entered %s\n", __func__);
+
 	return i2c_probe(adap, &addr_data, lm4857_amp_probe);
 }
 
@@ -620,6 +647,8 @@ static u8 lm4857_state;
 
 static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 {
+	DBG("Entered %s\n", __func__);
+
 	dev_dbg(&dev->dev, "lm4857_suspend\n");
 	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
 	if (lm4857_state) {
@@ -631,6 +660,8 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 
 static int lm4857_resume(struct i2c_client *dev)
 {
+	DBG("Entered %s\n", __func__);
+
 	if (lm4857_state) {
 		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
 		lm4857_write_regs();
@@ -640,6 +671,8 @@ static int lm4857_resume(struct i2c_client *dev)
 
 static void lm4857_shutdown(struct i2c_client *dev)
 {
+	DBG("Entered %s\n", __func__);
+
 	dev_dbg(&dev->dev, "lm4857_shutdown\n");
 	lm4857_regs[LM4857_CTRL] &= 0xf0;
 	lm4857_write_regs();
@@ -671,6 +704,8 @@ static int __init neo1973_init(void)
 {
 	int ret;
 
+	DBG("Entered %s\n", __func__);
+
 	neo1973_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!neo1973_snd_device)
 		return -ENOMEM;
@@ -691,6 +726,8 @@ static int __init neo1973_init(void)
 
 static void __exit neo1973_exit(void)
 {
+	DBG("Entered %s\n", __func__);
+
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
 }
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index e81d9a6c83da..0eed140dcd9b 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -10,9 +10,6 @@
  *  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.
- *
- *  Revision history
- *	21st Mar 2007   Initial Version
  */
 
 #include <linux/init.h>
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 1ed6afd45459..4c52f7946d9e 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -12,11 +12,6 @@
  *  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
- *    11th Dec 2006   Merged with Simtec driver
- *    10th Nov 2006   Initial version.
  */
 
 #include <linux/init.h>
@@ -180,7 +175,7 @@ static void s3c24xx_snd_rxctrl(int on)
 static int s3c24xx_snd_lrsync(void)
 {
 	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+	int timeout = 50; /* 5ms */
 
 	DBG("Entered %s\n", __func__);
 
@@ -189,8 +184,9 @@ static int s3c24xx_snd_lrsync(void)
 		if (iiscon & S3C2410_IISCON_LRINDEX)
 			break;
 
-		if (time_after(jiffies, timeout))
+		if (!timeout--)
 			return -ETIMEDOUT;
+		udelay(100);
 	}
 
 	return 0;
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 7806ae614617..ef599745159c 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -12,10 +12,6 @@
  *  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
- *    11th Dec 2006   Merged with Simtec driver
- *    10th Nov 2006   Initial version.
  */
 
 #include <linux/module.h>
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index b4a56302b9ab..8515d6ff03f2 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -10,9 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    8th Mar 2007   Initial version.
- *
  */
 
 #include <linux/module.h>
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 4c1e013381c9..54bd604012af 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -3,7 +3,7 @@ menu "SoC Audio support for SuperH"
 
 config SND_SOC_PCM_SH7760
 	tristate "SoC Audio support for Renesas SH7760"
-	depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG
+	depends on CPU_SUBTYPE_SH7760 && SH_DMABRG
 	help
 	  Enable this option for SH7760 AC97/I2S audio support.
 
@@ -13,10 +13,9 @@ config SND_SOC_PCM_SH7760
 ##
 
 config SND_SOC_SH4_HAC
+	tristate
 	select AC97_BUS
 	select SND_SOC_AC97_BUS
-	select SND_AC97_CODEC
-	tristate
 
 config SND_SOC_SH4_SSI
 	tristate
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e148db940cfc..a3f091e0843a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -14,10 +14,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    12th Aug 2005   Initial version.
- *    25th Oct 2005   Working Codec, Interface and Platform registration.
- *
  *  TODO:
  *   o Add hw rules to enforce rates, etc.
  *   o More testing with other codecs/machines.
@@ -112,9 +108,9 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
-static inline const char* get_dai_name(int type)
+static inline const char *get_dai_name(int type)
 {
-	switch(type) {
+	switch (type) {
 	case SND_SOC_DAI_AC97_BUS:
 	case SND_SOC_DAI_AC97:
 		return "AC97";
@@ -182,9 +178,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	/* Check that the codec and cpu DAI's are compatible */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		runtime->hw.rate_min =
-			max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min);
+			max(codec_dai->playback.rate_min,
+			    cpu_dai->playback.rate_min);
 		runtime->hw.rate_max =
-			min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max);
+			min(codec_dai->playback.rate_max,
+			    cpu_dai->playback.rate_max);
 		runtime->hw.channels_min =
 			max(codec_dai->playback.channels_min,
 				cpu_dai->playback.channels_min);
@@ -197,9 +195,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			codec_dai->playback.rates & cpu_dai->playback.rates;
 	} else {
 		runtime->hw.rate_min =
-			max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min);
+			max(codec_dai->capture.rate_min,
+			    cpu_dai->capture.rate_min);
 		runtime->hw.rate_max =
-			min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max);
+			min(codec_dai->capture.rate_max,
+			    cpu_dai->capture.rate_max);
 		runtime->hw.channels_min =
 			max(codec_dai->capture.channels_min,
 				cpu_dai->capture.channels_min);
@@ -229,7 +229,7 @@ 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: %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);
@@ -276,7 +276,7 @@ static void close_delayed_work(struct work_struct *work)
 	int i;
 
 	mutex_lock(&pcm_mutex);
-	for(i = 0; i < codec->num_dai; i++) {
+	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
 
 		dbg("pop wq checking: %s status: %s waiting: %s\n",
@@ -287,12 +287,12 @@ static void close_delayed_work(struct work_struct *work)
 		/* are we waiting on this codec DAI stream */
 		if (codec_dai->pop_wait == 1) {
 
-			/* power down the codec to D1 if no longer active */
+			/* Reduce power if no longer active */
 			if (codec->active == 0) {
 				dbg("pop wq D1 %s %s\n", codec->name,
 					codec_dai->playback.stream_name);
-				snd_soc_dapm_device_event(socdev,
-					SNDRV_CTL_POWER_D1);
+				snd_soc_dapm_set_bias_level(socdev,
+					SND_SOC_BIAS_PREPARE);
 			}
 
 			codec_dai->pop_wait = 0;
@@ -300,12 +300,12 @@ static void close_delayed_work(struct work_struct *work)
 				codec_dai->playback.stream_name,
 				SND_SOC_DAPM_STREAM_STOP);
 
-			/* power down the codec power domain if no longer active */
+			/* 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);
-				snd_soc_dapm_device_event(socdev,
-					SNDRV_CTL_POWER_D3hot);
+				snd_soc_dapm_set_bias_level(socdev,
+					SND_SOC_BIAS_STANDBY);
 			}
 		}
 	}
@@ -365,8 +365,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 			SND_SOC_DAPM_STREAM_STOP);
 
 		if (codec->active == 0 && codec_dai->pop_wait == 0)
-			snd_soc_dapm_device_event(socdev,
-						SNDRV_CTL_POWER_D3hot);
+			snd_soc_dapm_set_bias_level(socdev,
+						SND_SOC_BIAS_STANDBY);
 	}
 
 	mutex_unlock(&pcm_mutex);
@@ -439,9 +439,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	} else {
 		/* no delayed work - do we need to power up codec */
-		if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
+		if (codec->bias_level != SND_SOC_BIAS_ON) {
 
-			snd_soc_dapm_device_event(socdev,  SNDRV_CTL_POWER_D1);
+			snd_soc_dapm_set_bias_level(socdev,
+						    SND_SOC_BIAS_PREPARE);
 
 			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 				snd_soc_dapm_stream_event(codec,
@@ -452,7 +453,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
 
-			snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0);
+			snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
 			if (codec_dai->dai_ops.digital_mute)
 				codec_dai->dai_ops.digital_mute(codec_dai, 0);
 
@@ -514,7 +515,7 @@ 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);
 		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't set interface %s hw params\n",
+			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
 			goto interface_err;
 		}
@@ -523,7 +524,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (platform->pcm_ops->hw_params) {
 		ret = platform->pcm_ops->hw_params(substream, params);
 		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't set platform %s hw params\n",
+			printk(KERN_ERR "asoc: platform %s hw params failed\n",
 				platform->name);
 			goto platform_err;
 		}
@@ -542,7 +543,7 @@ interface_err:
 		codec_dai->ops.hw_free(substream);
 
 codec_err:
-	if(machine->ops && machine->ops->hw_free)
+	if (machine->ops && machine->ops->hw_free)
 		machine->ops->hw_free(substream);
 
 	mutex_unlock(&pcm_mutex);
@@ -631,15 +632,15 @@ static struct snd_pcm_ops soc_pcm_ops = {
 /* powers down audio subsystem for suspend */
 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_codec_device *codec_dev = socdev->codec_dev;
+	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 snd_soc_codec *codec = socdev->codec;
 	int i;
 
 	/* mute any active DAC's */
-	for(i = 0; i < machine->num_links; i++) {
+	for (i = 0; i < machine->num_links; i++) {
 		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
 		if (dai->dai_ops.digital_mute && dai->playback.active)
 			dai->dai_ops.digital_mute(dai, 1);
@@ -652,7 +653,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	if (machine->suspend_pre)
 		machine->suspend_pre(pdev, state);
 
-	for(i = 0; i < machine->num_links; i++) {
+	for (i = 0; i < machine->num_links; i++) {
 		struct snd_soc_cpu_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);
@@ -662,9 +663,9 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 
 	/* close any waiting streams and save state */
 	run_delayed_work(&socdev->delayed_work);
-	codec->suspend_dapm_state = codec->dapm_state;
+	codec->suspend_bias_level = codec->bias_level;
 
-	for(i = 0; i < codec->num_dai; i++) {
+	for (i = 0; i < codec->num_dai; i++) {
 		char *stream = codec->dai[i].playback.stream_name;
 		if (stream != NULL)
 			snd_soc_dapm_stream_event(codec, stream,
@@ -678,7 +679,7 @@ 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++) {
+	for (i = 0; i < machine->num_links; i++) {
 		struct snd_soc_cpu_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);
@@ -693,17 +694,17 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 /* powers up audio subsystem after a suspend */
 static int soc_resume(struct platform_device *pdev)
 {
- 	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 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 snd_soc_codec *codec = socdev->codec;
 	int i;
 
 	if (machine->resume_pre)
 		machine->resume_pre(pdev);
 
-	for(i = 0; i < machine->num_links; i++) {
+	for (i = 0; i < machine->num_links; i++) {
 		struct snd_soc_cpu_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);
@@ -712,8 +713,8 @@ static int soc_resume(struct platform_device *pdev)
 	if (codec_dev->resume)
 		codec_dev->resume(pdev);
 
-	for(i = 0; i < codec->num_dai; i++) {
-		char* stream = codec->dai[i].playback.stream_name;
+	for (i = 0; i < codec->num_dai; i++) {
+		char *stream = codec->dai[i].playback.stream_name;
 		if (stream != NULL)
 			snd_soc_dapm_stream_event(codec, stream,
 				SND_SOC_DAPM_STREAM_RESUME);
@@ -723,14 +724,14 @@ static int soc_resume(struct platform_device *pdev)
 				SND_SOC_DAPM_STREAM_RESUME);
 	}
 
-	/* unmute any active DAC's */
-	for(i = 0; i < machine->num_links; i++) {
+	/* unmute any active DACs */
+	for (i = 0; i < machine->num_links; i++) {
 		struct snd_soc_codec_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 < machine->num_links; i++) {
+	for (i = 0; i < machine->num_links; i++) {
 		struct snd_soc_cpu_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);
@@ -760,7 +761,7 @@ static int soc_probe(struct platform_device *pdev)
 
 	if (machine->probe) {
 		ret = machine->probe(pdev);
-		if(ret < 0)
+		if (ret < 0)
 			return ret;
 	}
 
@@ -768,20 +769,20 @@ static int soc_probe(struct platform_device *pdev)
 		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
 		if (cpu_dai->probe) {
 			ret = cpu_dai->probe(pdev);
-			if(ret < 0)
+			if (ret < 0)
 				goto cpu_dai_err;
 		}
 	}
 
 	if (codec_dev->probe) {
 		ret = codec_dev->probe(pdev);
-		if(ret < 0)
+		if (ret < 0)
 			goto cpu_dai_err;
 	}
 
 	if (platform->probe) {
 		ret = platform->probe(pdev);
-		if(ret < 0)
+		if (ret < 0)
 			goto platform_err;
 	}
 
@@ -868,7 +869,7 @@ 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,
+	sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
 		get_dai_name(cpu_dai->type), num);
 
 	if (codec_dai->playback.channels_min)
@@ -879,7 +880,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
 		capture, &pcm);
 	if (ret < 0) {
-		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
+			codec->name);
 		kfree(rtd);
 		return ret;
 	}
@@ -928,8 +930,9 @@ static ssize_t codec_reg_show(struct device *dev,
 		step = codec->reg_cache_step;
 
 	count += sprintf(buf, "%s registers\n", codec->name);
-	for(i = 0; i < codec->reg_cache_size; i += step)
-		count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i));
+	for (i = 0; i < codec->reg_cache_size; i += step)
+		count += sprintf(buf + count, "%2x: %4x\n", i,
+			codec->read(codec, i));
 
 	return count;
 }
@@ -1072,7 +1075,7 @@ 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++) {
+	for (i = 0; i < machine->num_links; i++) {
 		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't create pcm %s\n",
@@ -1102,7 +1105,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 	struct snd_soc_machine *machine = socdev->machine;
 	int ret = 0, i, ac97 = 0, err = 0;
 
-	for(i = 0; i < machine->num_links; i++) {
+	for (i = 0; i < machine->num_links; i++) {
 		if (socdev->machine->dai_link[i].init) {
 			err = socdev->machine->dai_link[i].init(codec);
 			if (err < 0) {
@@ -1111,7 +1114,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 				continue;
 			}
 		}
-		if (socdev->machine->dai_link[i].codec_dai->type == 
+		if (socdev->machine->dai_link[i].codec_dai->type ==
 			SND_SOC_DAI_AC97_BUS)
 			ac97 = 1;
 	}
@@ -1122,7 +1125,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 
 	ret = snd_card_register(codec->card);
 	if (ret < 0) {
-		printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
+		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
 				codec->name);
 		goto out;
 	}
@@ -1146,7 +1149,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 
 	err = device_create_file(socdev->dev, &dev_attr_codec_reg);
 	if (err < 0)
-		printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n");
+		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
 	mutex_unlock(&codec->mutex);
 
@@ -1172,7 +1175,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
 
 	mutex_lock(&codec->mutex);
 #ifdef CONFIG_SND_SOC_AC97_BUS
-	for(i = 0; i < codec->num_dai; i++) {
+	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
 		if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
 			soc_ac97_dev_unregister(codec);
@@ -1282,7 +1285,8 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
 	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
 		;
 	val = snd_soc_read(codec, e->reg);
-	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+	ucontrol->value.enumerated.item[0]
+		= (val >> e->shift_l) & (bitmask - 1);
 	if (e->shift_l != e->shift_r)
 		ucontrol->value.enumerated.item[1] =
 			(val >> e->shift_r) & (bitmask - 1);
@@ -1576,7 +1580,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
 	val = val << shift;
 	val2 = val2 << shift;
 
-	if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
+	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);
@@ -1592,7 +1597,7 @@ static int __devinit snd_soc_init(void)
 
 static void snd_soc_exit(void)
 {
- 	platform_driver_unregister(&soc_driver);
+	platform_driver_unregister(&soc_driver);
 }
 
 module_init(snd_soc_init);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index af3326c63504..728f3ac2f304 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -10,11 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    12th Aug 2005   Initial version.
- *    25th Oct 2005   Implemented path power domain.
- *    18th Dec 2005   Implemented machine and stream level power domain.
- *
  *  Features:
  *    o Changes power status of internal codec blocks depending on the
  *      dynamic configuration of codec internal audio paths and active
@@ -768,21 +763,18 @@ static ssize_t dapm_widget_show(struct device *dev,
 		}
 	}
 
-	switch(codec->dapm_state){
-	case SNDRV_CTL_POWER_D0:
-		state = "D0";
-		break;
-	case SNDRV_CTL_POWER_D1:
-		state = "D1";
+	switch (codec->bias_level) {
+	case SND_SOC_BIAS_ON:
+		state = "On";
 		break;
-	case SNDRV_CTL_POWER_D2:
-		state = "D2";
+	case SND_SOC_BIAS_PREPARE:
+		state = "Prepare";
 		break;
-	case SNDRV_CTL_POWER_D3hot:
-		state = "D3hot";
+	case SND_SOC_BIAS_STANDBY:
+		state = "Standby";
 		break;
-	case SNDRV_CTL_POWER_D3cold:
-		state = "D3cold";
+	case SND_SOC_BIAS_OFF:
+		state = "Off";
 		break;
 	}
 	count += sprintf(buf + count, "PM State: %s\n", state);
@@ -841,21 +833,8 @@ int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
 
-/**
- * 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.
- *
- * 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)
+static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
+	const char *sink, const char *control, const char *source)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
@@ -957,9 +936,64 @@ err:
 	kfree(path);
 	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
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * 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.
+ *
+ * Returns 0 for success else error. On error all resources can be freed
+ * with a call to snd_soc_card_free().
+ */
+int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+			    const struct snd_soc_dapm_route *route, int num)
+{
+	int i, ret;
+
+	for (i = 0; i < num; i++) {
+		ret = snd_soc_dapm_add_route(codec, route->sink,
+					     route->control, route->source);
+		if (ret < 0) {
+			printk(KERN_ERR "Failed to add route %s->%s\n",
+			       route->source,
+			       route->sink);
+			return ret;
+		}
+		route++;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
+
+/**
  * snd_soc_dapm_new_widgets - add new dapm widgets
  * @codec: audio codec
  *
@@ -1234,6 +1268,33 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
 
 /**
+ * snd_soc_dapm_new_controls - create new dapm controls
+ * @codec: audio codec
+ * @widget: widget array
+ * @num: number of widgets
+ *
+ * Creates new DAPM controls based upon the templates.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+	const struct snd_soc_dapm_widget *widget,
+	int num)
+{
+	int i, ret;
+
+	for (i = 0; i < num; i++) {
+		ret = snd_soc_dapm_new_control(codec, widget);
+		if (ret < 0)
+			return ret;
+		widget++;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
+
+
+/**
  * snd_soc_dapm_stream_event - send a stream event to the dapm core
  * @codec: audio codec
  * @stream: stream name
@@ -1294,27 +1355,28 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 
 /**
- * snd_soc_dapm_device_event - send a device event to the dapm core
+ * snd_soc_dapm_set_bias_level - set the bias level for the system
  * @socdev: audio device
- * @event: device event
+ * @level: level to configure
  *
- * Sends a device event to the dapm core. The core then makes any
- * necessary machine or codec power changes..
+ * Configure the bias (power) levels for the SoC audio device.
  *
  * Returns 0 for success else error.
  */
-int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event)
+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;
+	int ret = 0;
 
-	if (machine->dapm_event)
-		machine->dapm_event(machine, event);
-	if (codec->dapm_event)
-		codec->dapm_event(codec, event);
-	return 0;
+	if (machine->set_bias_level)
+		ret = machine->set_bias_level(machine, level);
+	if (ret == 0 && codec->set_bias_level)
+		ret = codec->set_bias_level(codec, level);
+
+	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event);
 
 /**
  * snd_soc_dapm_set_endpoint - set audio endpoint status
@@ -1343,6 +1405,29 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
 
 /**
+ * snd_soc_dapm_get_endpoint_status - get audio endpoint status
+ * @codec: audio codec
+ * @endpoint: audio signal endpoint (or start point)
+ *
+ * Get audio endpoint status - connected or disconnected.
+ *
+ * Returns status
+ */
+int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec,
+	char *endpoint)
+{
+	struct snd_soc_dapm_widget *w;
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (!strcmp(w->name, endpoint))
+			return w->connected;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status);
+
+/**
  * snd_soc_dapm_free - free dapm resources
  * @socdev: SoC device
  *
diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig
index 079e22af074c..d75deba5617d 100644
--- a/sound/sparc/Kconfig
+++ b/sound/sparc/Kconfig
@@ -1,11 +1,17 @@
 # ALSA Sparc drivers
 
-menu "ALSA Sparc devices"
-	depends on SND!=n && SPARC
+menuconfig SND_SPARC
+	bool "Sparc sound devices"
+	depends on SPARC
+	default y
+	help
+	  Support for sound devices specific to Sun SPARC architectures.
+
+if SND_SPARC
 
 config SND_SUN_AMD7930
 	tristate "Sun AMD7930"
-	depends on SBUS && SND
+	depends on SBUS
 	select SND_PCM
 	help
 	  Say Y here to include support for AMD7930 sound device on Sun.
@@ -15,7 +21,6 @@ config SND_SUN_AMD7930
 
 config SND_SUN_CS4231
 	tristate "Sun CS4231"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include support for CS4231 sound device on Sun.
@@ -25,7 +30,7 @@ config SND_SUN_CS4231
 
 config SND_SUN_DBRI
 	tristate "Sun DBRI"
-	depends on SND && SBUS
+	depends on SBUS
 	select SND_PCM
 	help
 	  Say Y here to include support for DBRI sound device on Sun.
@@ -33,4 +38,4 @@ config SND_SUN_DBRI
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-sun-dbri.
 
-endmenu
+endif	# SND_SPARC
diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig
index 0d08c29213c8..e6485be2e6f7 100644
--- a/sound/spi/Kconfig
+++ b/sound/spi/Kconfig
@@ -1,7 +1,13 @@
 #SPI drivers
 
-menu "SPI devices"
-	depends on SND != n
+menuconfig SND_SPI
+	bool "SPI sound devices"
+	depends on SPI
+	default y
+	help
+	  Support for sound devices connected via the SPI bus.
+
+if SND_SPI
 
 config SND_AT73C213
 	tristate "Atmel AT73C213 DAC driver"
@@ -28,4 +34,5 @@ config SND_AT73C213_TARGET_BITRATE
 
 	  Set to 48000 Hz by default.
 
-endmenu
+endif	# SND_SPI
+
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 9351b8a765b9..ffcdc8f4ef66 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -1,11 +1,16 @@
 # ALSA USB drivers
 
-menu "USB devices"
-	depends on SND!=n && USB!=n
+menuconfig SND_USB
+	bool "USB sound devices"
+	depends on USB
+	default y
+	help
+	  Support for sound devices connected via the USB bus.
+
+if SND_USB && USB
 
 config SND_USB_AUDIO
 	tristate "USB Audio/MIDI driver"
-	depends on SND && USB
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -18,7 +23,7 @@ config SND_USB_AUDIO
 
 config SND_USB_USX2Y
 	tristate "Tascam US-122, US-224 and US-428 USB driver"
-	depends on SND && USB && (X86 || PPC || ALPHA)
+	depends on X86 || PPC || ALPHA
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -31,7 +36,6 @@ config SND_USB_USX2Y
 
 config SND_USB_CAIAQ
 	tristate "Native Instruments USB audio devices"
-	depends on SND && USB
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -63,5 +67,5 @@ config SND_USB_CAIAQ_INPUT
 	   * Native Instruments Kore Controller 2
 	   * Native Instruments Audio Kontrol 1
 
-endmenu
+endif	# SND_USB
 
diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c
index 24970a5c888f..b3a603325835 100644
--- a/sound/usb/caiaq/caiaq-audio.c
+++ b/sound/usb/caiaq/caiaq-audio.c
@@ -637,6 +637,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	switch (dev->chip.usb_id) {
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO):
 		dev->samplerates |= SNDRV_PCM_RATE_88200;
 		dev->samplerates |= SNDRV_PCM_RATE_192000;
 		break;
diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index a972f77bd785..83175083e50f 100644
--- a/sound/usb/caiaq/caiaq-device.c
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -42,14 +42,15 @@
 #endif
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.6");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
 			 "{Native Instruments, Kore Controller},"
 			 "{Native Instruments, Kore Controller 2},"
-			 "{Native Instruments, Audio Kontrol 1}"
-			 "{Native Instruments, Audio 8 DJ}}");
+			 "{Native Instruments, Audio Kontrol 1},"
+			 "{Native Instruments, Audio 8 DJ},"
+			 "{Native Instruments, Session I/O}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
@@ -110,6 +111,11 @@ static struct usb_device_id snd_usb_id_table[] = {
 		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
 		.idProduct =    USB_PID_AUDIO8DJ
 	},
+	{
+		.match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
+		.idProduct =    USB_PID_SESSIONIO
+	},
 	{ /* terminator */ }
 };
 
diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h
index 96a491379c60..f9fbdbae269d 100644
--- a/sound/usb/caiaq/caiaq-device.h
+++ b/sound/usb/caiaq/caiaq-device.h
@@ -11,6 +11,7 @@
 #define USB_PID_KORECONTROLLER2	0x4712
 #define USB_PID_AK1		0x0815
 #define USB_PID_AUDIO8DJ	0x1978
+#define USB_PID_SESSIONIO	0x1915
 
 #define EP1_BUFSIZE 64
 #define CAIAQ_USB_STR_LEN 0xff
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 82a8d14c26af..b7ab3ee7647e 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -1379,6 +1379,39 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
+{
+	/* Roland SonicCell */
+	USB_DEVICE(0x0582, 0x00c2),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Roland",
+		.product_name = "SonicCell",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables  = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+
+
 /* Guillemot devices */
 {
 	/*