diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 598c22f..5de23c0 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -4288,7 +4288,7 @@
 <![CDATA[
   struct snd_rawmidi *rmidi;
   snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags,
-                      irq, irq_flags, &rmidi);
+                      irq, &rmidi);
 ]]>
           </programlisting>
         </informalexample>
@@ -4343,6 +4343,13 @@
 	by itself to start processing the output stream in the irq handler.
 	</para>
 
+	<para>
+	If the MPU-401 interface shares its interrupt with the other logical
+	devices on the card, set <constant>MPU401_INFO_IRQ_HOOK</constant>
+	(see <link linkend="midi-interface-interrupt-handler"><citetitle>
+	below</citetitle></link>).
+	</para>
+
       <para>
         Usually, the port address corresponds to the command port and
         port + 1 corresponds to the data port. If not, you may change
@@ -4375,14 +4382,12 @@
       </para>
 
       <para>
-        The 6th argument specifies the irq number for UART. If the irq
-      is already allocated, pass 0 to the 7th argument
-      (<parameter>irq_flags</parameter>). Otherwise, pass the flags
-      for irq allocation 
-      (<constant>SA_XXX</constant> bits) to it, and the irq will be
-      reserved by the mpu401-uart layer. If the card doesn't generate
-      UART interrupts, pass -1 as the irq number. Then a timer
-      interrupt will be invoked for polling. 
+	The 6th argument specifies the ISA irq number that will be
+	allocated.  If no interrupt is to be allocated (because your
+	code is already allocating a shared interrupt, or because the
+	device does not use interrupts), pass -1 instead.
+	For a MPU-401 device without an interrupt, a polling timer
+	will be used instead.
       </para>
     </section>
 
@@ -4390,12 +4395,13 @@
       <title>Interrupt Handler</title>
       <para>
         When the interrupt is allocated in
-      <function>snd_mpu401_uart_new()</function>, the private
-      interrupt handler is used, hence you don't have anything else to do
-      than creating the mpu401 stuff. Otherwise, you have to call
-      <function>snd_mpu401_uart_interrupt()</function> explicitly when
-      a UART interrupt is invoked and checked in your own interrupt
-      handler.  
+      <function>snd_mpu401_uart_new()</function>, an exclusive ISA
+      interrupt handler is automatically used, hence you don't have
+      anything else to do than creating the mpu401 stuff.  Otherwise, you
+      have to set <constant>MPU401_INFO_IRQ_HOOK</constant>, and call
+      <function>snd_mpu401_uart_interrupt()</function> explicitly from your
+      own interrupt handler when it has determined that a UART interrupt
+      has occurred.
       </para>
 
       <para>
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 8975701..936699e 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -886,6 +886,12 @@
 		disable)
     power_save_controller - Reset HD-audio controller in power-saving mode
 		(default = on)
+    align_buffer_size - Force rounding of buffer/period sizes to multiples
+    		      of 128 bytes. This is more efficient in terms of memory
+		      access but isn't required by the HDA spec and prevents
+		      users from specifying exact period/buffer sizes.
+		      (default = on)
+    snoop	- Enable/disable snooping (default = on)
 
     This module supports multiple cards and autoprobe.
     
diff --git a/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt
index 1482035..e9621e3 100644
--- a/Documentation/sound/alsa/HD-Audio-Controls.txt
+++ b/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -98,3 +98,19 @@
 
 * Auto-Mute Mode
   See Reatek codecs.
+
+
+Analog codecs
+--------------
+
+* Channel Mode
+  This is an enum control to change the surround-channel setup,
+  appears only when the surround channels are available.
+  It gives the number of channels to be used, "2ch", "4ch" and "6ch".
+  According to the configuration, this also controls the
+  jack-retasking of multi-I/O jacks.
+
+* Independent HP
+  When this enum control is enabled, the headphone output is routed
+  from an individual stream (the third PCM such as hw:0,2) instead of
+  the primary stream.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index d70c93b..4f34432 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -29,9 +29,6 @@
 
 ALC260
 ======
-  hp		HP machines
-  hp-3013	HP machines (3013-variant)
-  hp-dc7600	HP DC7600
   fujitsu	Fujitsu S7020
   acer		Acer TravelMate
   will		Will laptops (PB V7900)
@@ -46,15 +43,10 @@
 ALC262
 ======
   fujitsu	Fujitsu Laptop
-  hp-bpc	HP xw4400/6400/8400/9400 laptops
-  hp-bpc-d7000	HP BPC D7000
-  hp-tc-t5735	HP Thin Client T5735
-  hp-rp5700	HP RP5700
   benq		Benq ED8
   benq-t31	Benq T31
   hippo		Hippo (ATI) with jack detection, Sony UX-90s
   hippo_1	Hippo (Benq) with jack detection
-  sony-assamd	Sony ASSAMD
   toshiba-s06	Toshiba S06
   toshiba-rx1	Toshiba RX1
   tyan		Tyan Thunder n6650W (S2915-E)
@@ -66,43 +58,15 @@
 
 ALC267/268
 ==========
-  quanta-il1	Quanta IL1 mini-notebook
-  3stack	3-stack model
-  toshiba	Toshiba A205
-  acer		Acer laptops
-  acer-dmic	Acer laptops with digital-mic
-  acer-aspire	Acer Aspire One
-  dell		Dell OEM laptops (Vostro 1200)
-  zepto		Zepto laptops
-  test		for testing/debugging purpose, almost all controls can
-		adjusted.  Appearing only when compiled with
-		$CONFIG_SND_DEBUG=y
-  auto		auto-config reading BIOS (default)
+  N/A
 
 ALC269
 ======
-  basic		Basic preset
-  quanta	Quanta FL1
   laptop-amic	Laptops with analog-mic input
   laptop-dmic	Laptops with digital-mic input
-  fujitsu	FSC Amilo
-  lifebook	Fujitsu Lifebook S6420
-  auto		auto-config reading BIOS (default)
 
 ALC662/663/272
 ==============
-  3stack-dig	3-stack (2-channel) with SPDIF
-  3stack-6ch	 3-stack (6-channel)
-  3stack-6ch-dig 3-stack (6-channel) with SPDIF
-  5stack-dig	 5-stack with SPDIF
-  lenovo-101e	 Lenovo laptop
-  eeepc-p701	ASUS Eeepc P701
-  eeepc-ep20	ASUS Eeepc EP20
-  ecs		ECS/Foxconn mobo
-  m51va		ASUS M51VA
-  g71v		ASUS G71V
-  h13		ASUS H13
-  g50v		ASUS G50V
   asus-mode1	ASUS
   asus-mode2	ASUS
   asus-mode3	ASUS
@@ -111,15 +75,10 @@
   asus-mode6	ASUS
   asus-mode7	ASUS
   asus-mode8	ASUS
-  dell		Dell with ALC272
-  dell-zm1	Dell ZM1 with ALC272
-  samsung-nc10	Samsung NC10 mini notebook
-  auto		auto-config reading BIOS (default)
 
 ALC680
 ======
-  base		Base model (ASUS NX90)
-  auto		auto-config reading BIOS (default)
+  N/A
 
 ALC882/883/885/888/889
 ======================
@@ -175,28 +134,11 @@
 
 ALC861/660
 ==========
-  3stack	3-jack
-  3stack-dig	3-jack with SPDIF I/O
-  6stack-dig	6-jack with SPDIF I/O
-  3stack-660	3-jack (for ALC660)
-  uniwill-m31	Uniwill M31 laptop
-  toshiba	Toshiba laptop support
-  asus		Asus laptop support
-  asus-laptop	ASUS F2/F3 laptops
-  auto		auto-config reading BIOS (default)
+  N/A
 
 ALC861VD/660VD
 ==============
-  3stack	3-jack
-  3stack-dig	3-jack with SPDIF OUT
-  6stack-dig	6-jack with SPDIF OUT
-  3stack-660	3-jack (for ALC660VD)
-  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
-  lenovo	Lenovo 3000 C200
-  dallas	Dallas laptops
-  hp		HP TX1000
-  asus-v1s	ASUS V1Sn
-  auto		auto-config reading BIOS (default)
+  N/A
 
 CMI9880
 =======
@@ -289,7 +231,6 @@
   hp-dv6736	HP dv6736
   hp-f700	HP Compaq Presario F700
   ideapad	Lenovo IdeaPad laptop
-  lenovo-x200	Lenovo X200 laptop
   toshiba	Toshiba Satellite M300
 
 Conexant 5066
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c82beb0..03e2771 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -447,7 +447,10 @@
 three numbers indicating the codec vendor-id (0x12345678 in the
 example), the codec subsystem-id (0xabcd1234) and the address (2) of
 the codec.  The rest patch entries are applied to this specified codec
-until another codec entry is given.
+until another codec entry is given.  Passing 0 or a negative number to
+the first or the second value will make the check of the corresponding
+field be skipped.  It'll be useful for really broken devices that don't
+initialize SSID properly.
 
 The `[model]` line allows to change the model name of the each codec.
 In the example above, it will be changed to model=auto.
@@ -491,7 +494,7 @@
 The hd-audio driver reads the file via request_firmware().  Thus,
 a patch file has to be located on the appropriate firmware path,
 typically, /lib/firmware.  For example, when you pass the option
-`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be
+`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be
 present.
 
 The patch module option is specific to each card instance, and you
@@ -524,6 +527,54 @@
 check the current value.  If it's non-zero, the feature is turned on.
 
 
+Tracepoints
+~~~~~~~~~~~
+The hd-audio driver gives a few basic tracepoints.
+`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response`
+traces the response from RIRB (only when read from the codec driver).
+`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc,
+`hda:hda_unsol_event` traces the unsolicited events, and
+`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up
+via power-saving behavior.
+
+Enabling all tracepoints can be done like
+------------------------------------------------------------------------
+  # echo 1 > /sys/kernel/debug/tracing/events/hda/enable
+------------------------------------------------------------------------
+then after some commands, you can traces from
+/sys/kernel/debug/tracing/trace file.  For example, when you want to
+trace what codec command is sent, enable the tracepoint like:
+------------------------------------------------------------------------
+  # cat /sys/kernel/debug/tracing/trace
+  # tracer: nop
+  #
+  #       TASK-PID    CPU#    TIMESTAMP  FUNCTION
+  #          | |       |          |         |
+         <...>-7807  [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019
+         <...>-7807  [002] 105147.774893: hda_send_cmd: [0:0] val=e39019
+         <...>-7807  [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a
+         <...>-7807  [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a
+         <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019
+         <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019
+         <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a
+         <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a
+------------------------------------------------------------------------
+Here `[0:0]` indicates the card number and the codec address, and
+`val` shows the value sent to the codec, respectively.  The value is
+a packed value, and you can decode it via hda-decode-verb program
+included in hda-emu package below.  For example, the value e3a019 is
+to set the left output-amp value to 25.
+------------------------------------------------------------------------
+  % hda-decode-verb 0xe3a019
+  raw value = 0x00e3a019
+  cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19
+  raw value: verb = 0x3a0, parm = 0x19
+  verbname = set_amp_gain_mute
+  amp raw val = 0xa019
+  output, left, idx=0, mute=0, val=25
+------------------------------------------------------------------------
+
+
 Development Tree
 ~~~~~~~~~~~~~~~~
 The latest development codes for HD-audio are found on sound git tree:
diff --git a/MAINTAINERS b/MAINTAINERS
index bbf42cd..1dbbb27 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5991,7 +5991,7 @@
 M:	Takashi Iwai <tiwai@suse.de>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://www.alsa-project.org/
-T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
 T:	git git://git.alsa-project.org/alsa-kernel.git
 S:	Maintained
 F:	Documentation/sound/
diff --git a/include/linux/input.h b/include/linux/input.h
index a637e78..a514fb8 100644
--- a/include/linux/input.h
+++ b/include/linux/input.h
@@ -814,6 +814,7 @@
 #define SW_KEYPAD_SLIDE		0x0a  /* set = keypad slide out */
 #define SW_FRONT_PROXIMITY	0x0b  /* set = front proximity sensor active */
 #define SW_ROTATE_LOCK		0x0c  /* set = rotate locked/disabled */
+#define SW_LINEIN_INSERT	0x0d  /* set = inserted */
 #define SW_MAX			0x0f
 #define SW_CNT			(SW_MAX+1)
 
diff --git a/include/linux/usb/ch9.h b/include/linux/usb/ch9.h
index 0fd3fbd..f302535 100644
--- a/include/linux/usb/ch9.h
+++ b/include/linux/usb/ch9.h
@@ -377,12 +377,6 @@
 #define USB_ENDPOINT_NUMBER_MASK	0x0f	/* in bEndpointAddress */
 #define USB_ENDPOINT_DIR_MASK		0x80
 
-#define USB_ENDPOINT_SYNCTYPE		0x0c
-#define USB_ENDPOINT_SYNC_NONE		(0 << 2)
-#define USB_ENDPOINT_SYNC_ASYNC		(1 << 2)
-#define USB_ENDPOINT_SYNC_ADAPTIVE	(2 << 2)
-#define USB_ENDPOINT_SYNC_SYNC		(3 << 2)
-
 #define USB_ENDPOINT_XFERTYPE_MASK	0x03	/* in bmAttributes */
 #define USB_ENDPOINT_XFER_CONTROL	0
 #define USB_ENDPOINT_XFER_ISOC		1
@@ -390,6 +384,17 @@
 #define USB_ENDPOINT_XFER_INT		3
 #define USB_ENDPOINT_MAX_ADJUSTABLE	0x80
 
+#define USB_ENDPOINT_SYNCTYPE		0x0c
+#define USB_ENDPOINT_SYNC_NONE		(0 << 2)
+#define USB_ENDPOINT_SYNC_ASYNC		(1 << 2)
+#define USB_ENDPOINT_SYNC_ADAPTIVE	(2 << 2)
+#define USB_ENDPOINT_SYNC_SYNC		(3 << 2)
+
+#define USB_ENDPOINT_USAGE_MASK		0x30
+#define USB_ENDPOINT_USAGE_DATA		0x00
+#define USB_ENDPOINT_USAGE_FEEDBACK	0x10
+#define USB_ENDPOINT_USAGE_IMPLICIT_FB	0x20	/* Implicit feedback Data endpoint */
+
 /*-------------------------------------------------------------------------*/
 
 /**
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 5d6074f..a2e4ff5 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -706,7 +706,7 @@
  *                                                                          *
  ****************************************************************************/
 
-#define SNDRV_CTL_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 6)
+#define SNDRV_CTL_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 7)
 
 struct snd_ctl_card_info {
 	int card;			/* card number */
@@ -803,6 +803,8 @@
 			unsigned int items;	/* R: number of items */
 			unsigned int item;	/* W: item number */
 			char name[64];		/* R: value name */
+			__u64 names_ptr;	/* W: names list (ELEM_ADD only) */
+			unsigned int names_length;
 		} enumerated;
 		unsigned char reserved[128];
 	} value;
diff --git a/include/sound/initval.h b/include/sound/initval.h
index 1daa6df..f99a0d2 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -62,7 +62,7 @@
 {
 	while (*irq_table != -1) {
 		if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
-				 IRQF_DISABLED | IRQF_PROBE_SHARED, "ALSA Test IRQ",
+				 IRQF_PROBE_SHARED, "ALSA Test IRQ",
 				 (void *) irq_table)) {
 			free_irq(*irq_table, (void *) irq_table);
 			return *irq_table;
diff --git a/include/sound/jack.h b/include/sound/jack.h
index c140fc7..63c7907 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -42,6 +42,7 @@
 	SND_JACK_MECHANICAL	= 0x0008, /* If detected separately */
 	SND_JACK_VIDEOOUT	= 0x0010,
 	SND_JACK_AVOUT		= SND_JACK_LINEOUT | SND_JACK_VIDEOOUT,
+	SND_JACK_LINEIN		= 0x0020,
 
 	/* Kept separate from switches to facilitate implementation */
 	SND_JACK_BTN_0		= 0x4000,
diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h
index 1f1d53f..20230db 100644
--- a/include/sound/mpu401.h
+++ b/include/sound/mpu401.h
@@ -50,7 +50,10 @@
 #define MPU401_INFO_INTEGRATED	(1 << 2)	/* integrated h/w port */
 #define MPU401_INFO_MMIO	(1 << 3)	/* MMIO access */
 #define MPU401_INFO_TX_IRQ	(1 << 4)	/* independent TX irq */
+#define MPU401_INFO_IRQ_HOOK	(1 << 5)	/* mpu401 irq handler is called
+						   from driver irq handler */
 #define MPU401_INFO_NO_ACK	(1 << 6)	/* No ACK cmd needed */
+#define MPU401_INFO_USE_TIMER	(1 << 15)	/* internal */
 
 #define MPU401_MODE_BIT_INPUT		0
 #define MPU401_MODE_BIT_OUTPUT		1
@@ -73,8 +76,7 @@
 	unsigned long port;		/* base port of MPU-401 chip */
 	unsigned long cport;		/* port + 1 (usually) */
 	struct resource *res;		/* port resource */
-	int irq;			/* IRQ number of MPU-401 chip (-1 = poll) */
-	int irq_flags;
+	int irq;			/* IRQ number of MPU-401 chip */
 
 	unsigned long mode;		/* MPU401_MODE_XXXX */
 	int timer_invoked;
@@ -131,7 +133,6 @@
 			unsigned long port,
 			unsigned int info_flags,
 			int irq,
-			int irq_flags,
 			struct snd_rawmidi ** rrawmidi);
 
 #endif /* __SOUND_MPU401_H */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 57e71fa..3e7fda6 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -825,6 +825,8 @@
 int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
 			       unsigned int cond,
 			       snd_pcm_hw_param_t var);
+int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
+			       unsigned int base_rate);
 int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime,
 			unsigned int cond,
 			int var,
@@ -1035,6 +1037,8 @@
 	atomic_dec(&substream->mmap_count);
 }
 
+int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
+			     struct vm_area_struct *area);
 /* mmap for io-memory area */
 #if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA)
 #define SNDRV_PCM_INFO_MMAP_IOMEM	SNDRV_PCM_INFO_MMAP
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index 3687a6c..762af68 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -1067,7 +1067,6 @@
 	printk(KERN_DEBUG PFX "created and attached onyx instance\n");
 	return 0;
  fail:
-	i2c_set_clientdata(client, NULL);
 	kfree(onyx);
 	return -ENODEV;
 }
@@ -1112,8 +1111,7 @@
 
 	aoa_codec_unregister(&onyx->codec);
 	of_node_put(onyx->codec.node);
-	if (onyx->codec_info)
-		kfree(onyx->codec_info);
+	kfree(onyx->codec_info);
 	kfree(onyx);
 	return 0;
 }
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 3fd1a7e..552b97a 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@
 	sdev->pcmid = -1;
 	list_del(&ldev->list);
 	layouts_list_items--;
+	kfree(ldev);
  outnodev:
  	of_node_put(sound);
  	layout_device = NULL;
- 	kfree(ldev);
 	return -ENODEV;
 }
 
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index d0cead3..e518d38 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -443,7 +443,7 @@
 	mutex_lock(&aaci->irq_lock);
 	if (!aaci->users++) {
 		ret = request_irq(aaci->dev->irq[0], aaci_irq,
-			   IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci);
+			   IRQF_SHARED, DRIVER_NAME, aaci);
 		if (ret != 0)
 			aaci->users--;
 	}
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 88eec38..8ad6535 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -359,7 +359,7 @@
 	if (ret)
 		goto err_clk2;
 
-	ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
+	ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL);
 	if (ret < 0)
 		goto err_irq;
 
diff --git a/sound/core/control.c b/sound/core/control.c
index f8c5be4..978fe1a 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -989,7 +989,6 @@
 	void *tlv_data;			/* TLV data */
 	unsigned long tlv_data_size;	/* TLV data size */
 	void *priv_data;		/* private data (like strings for enumerated type) */
-	unsigned long priv_data_size;	/* size of private data in bytes */
 };
 
 static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol,
@@ -1001,6 +1000,28 @@
 	return 0;
 }
 
+static int snd_ctl_elem_user_enum_info(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_info *uinfo)
+{
+	struct user_element *ue = kcontrol->private_data;
+	const char *names;
+	unsigned int item;
+
+	item = uinfo->value.enumerated.item;
+
+	*uinfo = ue->info;
+
+	item = min(item, uinfo->value.enumerated.items - 1);
+	uinfo->value.enumerated.item = item;
+
+	names = ue->priv_data;
+	for (; item > 0; --item)
+		names += strlen(names) + 1;
+	strcpy(uinfo->value.enumerated.name, names);
+
+	return 0;
+}
+
 static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
 {
@@ -1055,11 +1076,46 @@
 	return change;
 }
 
+static int snd_ctl_elem_init_enum_names(struct user_element *ue)
+{
+	char *names, *p;
+	size_t buf_len, name_len;
+	unsigned int i;
+
+	if (ue->info.value.enumerated.names_length > 64 * 1024)
+		return -EINVAL;
+
+	names = memdup_user(
+		(const void __user *)ue->info.value.enumerated.names_ptr,
+		ue->info.value.enumerated.names_length);
+	if (IS_ERR(names))
+		return PTR_ERR(names);
+
+	/* check that there are enough valid names */
+	buf_len = ue->info.value.enumerated.names_length;
+	p = names;
+	for (i = 0; i < ue->info.value.enumerated.items; ++i) {
+		name_len = strnlen(p, buf_len);
+		if (name_len == 0 || name_len >= 64 || name_len == buf_len) {
+			kfree(names);
+			return -EINVAL;
+		}
+		p += name_len + 1;
+		buf_len -= name_len + 1;
+	}
+
+	ue->priv_data = names;
+	ue->info.value.enumerated.names_ptr = 0;
+
+	return 0;
+}
+
 static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol)
 {
 	struct user_element *ue = kcontrol->private_data;
-	if (ue->tlv_data)
-		kfree(ue->tlv_data);
+
+	kfree(ue->tlv_data);
+	kfree(ue->priv_data);
 	kfree(ue);
 }
 
@@ -1072,8 +1128,8 @@
 	long private_size;
 	struct user_element *ue;
 	int idx, err;
-	
-	if (card->user_ctl_count >= MAX_USER_CONTROLS)
+
+	if (!replace && card->user_ctl_count >= MAX_USER_CONTROLS)
 		return -ENOMEM;
 	if (info->count < 1)
 		return -EINVAL;
@@ -1101,7 +1157,10 @@
 	memcpy(&kctl.id, &info->id, sizeof(info->id));
 	kctl.count = info->owner ? info->owner : 1;
 	access |= SNDRV_CTL_ELEM_ACCESS_USER;
-	kctl.info = snd_ctl_elem_user_info;
+	if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED)
+		kctl.info = snd_ctl_elem_user_enum_info;
+	else
+		kctl.info = snd_ctl_elem_user_info;
 	if (access & SNDRV_CTL_ELEM_ACCESS_READ)
 		kctl.get = snd_ctl_elem_user_get;
 	if (access & SNDRV_CTL_ELEM_ACCESS_WRITE)
@@ -1122,6 +1181,11 @@
 		if (info->count > 64)
 			return -EINVAL;
 		break;
+	case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
+		private_size = sizeof(unsigned int);
+		if (info->count > 128 || info->value.enumerated.items == 0)
+			return -EINVAL;
+		break;
 	case SNDRV_CTL_ELEM_TYPE_BYTES:
 		private_size = sizeof(unsigned char);
 		if (info->count > 512)
@@ -1143,9 +1207,17 @@
 	ue->info.access = 0;
 	ue->elem_data = (char *)ue + sizeof(*ue);
 	ue->elem_data_size = private_size;
+	if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) {
+		err = snd_ctl_elem_init_enum_names(ue);
+		if (err < 0) {
+			kfree(ue);
+			return err;
+		}
+	}
 	kctl.private_free = snd_ctl_elem_user_free;
 	_kctl = snd_ctl_new(&kctl, access);
 	if (_kctl == NULL) {
+		kfree(ue->priv_data);
 		kfree(ue);
 		return -ENOMEM;
 	}
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index 4268744..2bb95a7 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -83,6 +83,8 @@
 			u32 items;
 			u32 item;
 			char name[64];
+			u64 names_ptr;
+			u32 names_length;
 		} enumerated;
 		unsigned char reserved[128];
 	} value;
@@ -372,6 +374,8 @@
 				   &data32->value.enumerated,
 				   sizeof(data->value.enumerated)))
 			goto error;
+		data->value.enumerated.names_ptr =
+			(uintptr_t)compat_ptr(data->value.enumerated.names_ptr);
 		break;
 	default:
 		break;
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index a70ee7f..031e215 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -272,7 +272,14 @@
 			if (get_user(device, (int __user *)arg))
 				return -EFAULT;
 			mutex_lock(&register_mutex);
-			device = device < 0 ? 0 : device + 1;
+
+			if (device < 0)
+				device = 0;
+			else if (device < SNDRV_MINOR_HWDEPS)
+				device++;
+			else
+				device = SNDRV_MINOR_HWDEPS;
+
 			while (device < SNDRV_MINOR_HWDEPS) {
 				if (snd_hwdep_search(card, device))
 					break;
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 53b53e9..240a3e1 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -30,6 +30,7 @@
 	SW_LINEOUT_INSERT,
 	SW_JACK_PHYSICAL_INSERT,
 	SW_VIDEOOUT_INSERT,
+	SW_LINEIN_INSERT,
 };
 
 static int snd_jack_dev_free(struct snd_device *device)
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index d8359cf..1b5e0c4 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -499,7 +499,7 @@
 	
 	memset(&id, 0, sizeof(id));
 	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
-	strcpy(id.name, name);
+	strlcpy(id.name, name, sizeof(id.name));
 	id.index = index;
 	return snd_ctl_find_id(card, &id);
 }
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 86d0caf..95d1e78 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1399,6 +1399,32 @@
 
 EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2);
 
+static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params,
+					   struct snd_pcm_hw_rule *rule)
+{
+	unsigned int base_rate = (unsigned int)(uintptr_t)rule->private;
+	struct snd_interval *rate;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	return snd_interval_list(rate, 1, &base_rate, 0);
+}
+
+/**
+ * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling
+ * @runtime: PCM runtime instance
+ * @base_rate: the rate at which the hardware does not resample
+ */
+int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
+			       unsigned int base_rate)
+{
+	return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE,
+				   SNDRV_PCM_HW_PARAM_RATE,
+				   snd_pcm_hw_rule_noresample_func,
+				   (void *)(uintptr_t)base_rate,
+				   SNDRV_PCM_HW_PARAM_RATE, -1);
+}
+EXPORT_SYMBOL(snd_pcm_hw_rule_noresample);
+
 static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params,
 				  snd_pcm_hw_param_t var)
 {
@@ -1761,6 +1787,10 @@
 	snd_pcm_uframes_t avail = 0;
 	long wait_time, tout;
 
+	init_waitqueue_entry(&wait, current);
+	set_current_state(TASK_INTERRUPTIBLE);
+	add_wait_queue(&runtime->tsleep, &wait);
+
 	if (runtime->no_period_wakeup)
 		wait_time = MAX_SCHEDULE_TIMEOUT;
 	else {
@@ -1771,16 +1801,32 @@
 		}
 		wait_time = msecs_to_jiffies(wait_time * 1000);
 	}
-	init_waitqueue_entry(&wait, current);
-	add_wait_queue(&runtime->tsleep, &wait);
+
 	for (;;) {
 		if (signal_pending(current)) {
 			err = -ERESTARTSYS;
 			break;
 		}
+
+		/*
+		 * We need to check if space became available already
+		 * (and thus the wakeup happened already) first to close
+		 * the race of space already having become available.
+		 * This check must happen after been added to the waitqueue
+		 * and having current state be INTERRUPTIBLE.
+		 */
+		if (is_playback)
+			avail = snd_pcm_playback_avail(runtime);
+		else
+			avail = snd_pcm_capture_avail(runtime);
+		if (avail >= runtime->twake)
+			break;
 		snd_pcm_stream_unlock_irq(substream);
-		tout = schedule_timeout_interruptible(wait_time);
+
+		tout = schedule_timeout(wait_time);
+
 		snd_pcm_stream_lock_irq(substream);
+		set_current_state(TASK_INTERRUPTIBLE);
 		switch (runtime->status->state) {
 		case SNDRV_PCM_STATE_SUSPENDED:
 			err = -ESTRPIPE;
@@ -1806,14 +1852,9 @@
 			err = -EIO;
 			break;
 		}
-		if (is_playback)
-			avail = snd_pcm_playback_avail(runtime);
-		else
-			avail = snd_pcm_capture_avail(runtime);
-		if (avail >= runtime->twake)
-			break;
 	}
  _endloop:
+	set_current_state(TASK_RUNNING);
 	remove_wait_queue(&runtime->tsleep, &wait);
 	*availp = avail;
 	return err;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1c6be91..77d7df2 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2058,16 +2058,12 @@
 
 static int snd_pcm_open_file(struct file *file,
 			     struct snd_pcm *pcm,
-			     int stream,
-			     struct snd_pcm_file **rpcm_file)
+			     int stream)
 {
 	struct snd_pcm_file *pcm_file;
 	struct snd_pcm_substream *substream;
 	int err;
 
-	if (rpcm_file)
-		*rpcm_file = NULL;
-
 	err = snd_pcm_open_substream(pcm, stream, file, &substream);
 	if (err < 0)
 		return err;
@@ -2083,8 +2079,7 @@
 		substream->pcm_release = pcm_release_private;
 	}
 	file->private_data = pcm_file;
-	if (rpcm_file)
-		*rpcm_file = pcm_file;
+
 	return 0;
 }
 
@@ -2113,7 +2108,6 @@
 static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream)
 {
 	int err;
-	struct snd_pcm_file *pcm_file;
 	wait_queue_t wait;
 
 	if (pcm == NULL) {
@@ -2131,7 +2125,7 @@
 	add_wait_queue(&pcm->open_wait, &wait);
 	mutex_lock(&pcm->open_mutex);
 	while (1) {
-		err = snd_pcm_open_file(file, pcm, stream, &pcm_file);
+		err = snd_pcm_open_file(file, pcm, stream);
 		if (err >= 0)
 			break;
 		if (err == -EAGAIN) {
@@ -3156,8 +3150,8 @@
 /*
  * mmap the DMA buffer on RAM
  */
-static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
-				struct vm_area_struct *area)
+int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
+			     struct vm_area_struct *area)
 {
 	area->vm_flags |= VM_RESERVED;
 #ifdef ARCH_HAS_DMA_MMAP_COHERENT
@@ -3177,6 +3171,7 @@
 	area->vm_ops = &snd_pcm_vm_ops_data_fault;
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap);
 
 /*
  * mmap the DMA buffer on I/O memory area
@@ -3242,7 +3237,7 @@
 	if (substream->ops->mmap)
 		err = substream->ops->mmap(substream, area);
 	else
-		err = snd_pcm_default_mmap(substream, area);
+		err = snd_pcm_lib_default_mmap(substream, area);
 	if (!err)
 		atomic_inc(&substream->mmap_count);
 	return err;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7c1cbf0..67ebf1c 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -328,6 +328,8 @@
 		mutex_unlock(&register_mutex);
 	} else {
 		timer = timeri->timer;
+		if (snd_BUG_ON(!timer))
+			goto out;
 		/* wait, until the active callback is finished */
 		spin_lock_irq(&timer->lock);
 		while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
@@ -353,6 +355,7 @@
 		}
 		mutex_unlock(&register_mutex);
 	}
+ out:
 	if (timeri->private_free)
 		timeri->private_free(timeri);
 	kfree(timeri->owner);
@@ -531,6 +534,8 @@
 	if (err < 0)
 		return err;
 	timer = timeri->timer;
+	if (!timer)
+		return -EINVAL;
 	spin_lock_irqsave(&timer->lock, flags);
 	timeri->cticks = timeri->ticks;
 	timeri->pticks = 0;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index a0da775..4067f15 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -575,7 +575,8 @@
 static int loopback_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
 {
-	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	return snd_pcm_lib_alloc_vmalloc_buffer(substream,
+						params_buffer_bytes(params));
 }
 
 static int loopback_hw_free(struct snd_pcm_substream *substream)
@@ -587,7 +588,7 @@
 	mutex_lock(&dpcm->loopback->cable_lock);
 	cable->valid &= ~(1 << substream->stream);
 	mutex_unlock(&dpcm->loopback->cable_lock);
-	return snd_pcm_lib_free_pages(substream);
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
 }
 
 static unsigned int get_cable_index(struct snd_pcm_substream *substream)
@@ -740,6 +741,8 @@
 	.prepare =	loopback_prepare,
 	.trigger =	loopback_trigger,
 	.pointer =	loopback_pointer,
+	.page =		snd_pcm_lib_get_vmalloc_page,
+	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 static struct snd_pcm_ops loopback_capture_ops = {
@@ -751,6 +754,8 @@
 	.prepare =	loopback_prepare,
 	.trigger =	loopback_trigger,
 	.pointer =	loopback_pointer,
+	.page =		snd_pcm_lib_get_vmalloc_page,
+	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 static int __devinit loopback_pcm_new(struct loopback *loopback,
@@ -771,10 +776,6 @@
 	strcpy(pcm->name, "Loopback PCM");
 
 	loopback->pcm[device] = pcm;
-
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
-			snd_dma_continuous_data(GFP_KERNEL),
-			0, 2 * 1024 * 1024);
 	return 0;
 }
 
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index 5cfcb90..2c7a763 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@
 		   "0x%x done\n", (unsigned int)ml403_ac97cr->port);
 	/* get irq */
 	irq = platform_get_irq(pfdev, 0);
-	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+	if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
 			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
@@ -1166,7 +1166,7 @@
 		   "request (playback) irq %d done\n",
 		   ml403_ac97cr->irq);
 	irq = platform_get_irq(pfdev, 1);
-	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+	if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
 			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 149d05a..1c02852 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -86,8 +86,7 @@
 	}
 
 	err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0,
-				  irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0,
-				  NULL);
+				  irq[dev], NULL);
 	if (err < 0) {
 		printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]);
 		goto _err;
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 2af0999..e91698a 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -3,7 +3,7 @@
  *  Routines for control of MPU-401 in UART mode
  *
  *  MPU-401 supports UART mode which is not capable generate transmit
- *  interrupts thus output is done via polling. Also, if irq < 0, then
+ *  interrupts thus output is done via polling. Without interrupt,
  *  input is done also via polling. Do not expect good performance.
  *
  *
@@ -374,7 +374,7 @@
 			/* first time - flush FIFO */
 			while (max-- > 0)
 				mpu->read(mpu, MPU401D(mpu));
-			if (mpu->irq < 0)
+			if (mpu->info_flags & MPU401_INFO_USE_TIMER)
 				snd_mpu401_uart_add_timer(mpu, 1);
 		}
 		
@@ -383,7 +383,7 @@
 		snd_mpu401_uart_input_read(mpu);
 		spin_unlock_irqrestore(&mpu->input_lock, flags);
 	} else {
-		if (mpu->irq < 0)
+		if (mpu->info_flags & MPU401_INFO_USE_TIMER)
 			snd_mpu401_uart_remove_timer(mpu, 1);
 		clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode);
 	}
@@ -496,7 +496,7 @@
 static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
 {
 	struct snd_mpu401 *mpu = rmidi->private_data;
-	if (mpu->irq_flags && mpu->irq >= 0)
+	if (mpu->irq >= 0)
 		free_irq(mpu->irq, (void *) mpu);
 	release_and_free_resource(mpu->res);
 	kfree(mpu);
@@ -509,8 +509,7 @@
  * @hardware: the hardware type, MPU401_HW_XXXX
  * @port: the base address of MPU401 port
  * @info_flags: bitflags MPU401_INFO_XXX
- * @irq: the irq number, -1 if no interrupt for mpu
- * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved.
+ * @irq: the ISA irq number, -1 if not to be allocated
  * @rrawmidi: the pointer to store the new rawmidi instance
  *
  * Creates a new MPU-401 instance.
@@ -525,7 +524,7 @@
 			unsigned short hardware,
 			unsigned long port,
 			unsigned int info_flags,
-			int irq, int irq_flags,
+			int irq,
 			struct snd_rawmidi ** rrawmidi)
 {
 	struct snd_mpu401 *mpu;
@@ -577,8 +576,8 @@
 		mpu->cport = port + 2;
 	else
 		mpu->cport = port + 1;
-	if (irq >= 0 && irq_flags) {
-		if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags,
+	if (irq >= 0) {
+		if (request_irq(irq, snd_mpu401_uart_interrupt, 0,
 				"MPU401 UART", (void *) mpu)) {
 			snd_printk(KERN_ERR "mpu401_uart: "
 				   "unable to grab IRQ %d\n", irq);
@@ -586,9 +585,10 @@
 			return -EBUSY;
 		}
 	}
+	if (irq < 0 && !(info_flags & MPU401_INFO_IRQ_HOOK))
+		info_flags |= MPU401_INFO_USE_TIMER;
 	mpu->info_flags = info_flags;
 	mpu->irq = irq;
-	mpu->irq_flags = irq_flags;
 	if (card->shortname[0])
 		snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI",
 			 card->shortname);
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5c426df..1eef4cc 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -589,7 +589,7 @@
 		return -EBUSY;
 	}
 	mcard->port = port;
-	if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) {
+	if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) {
 		snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq);
 		return -EBUSY;
 	}
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index a25fb7b..fc1d822 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -816,7 +816,7 @@
 
 	if (irq >= 0 && irq != SNDRV_AUTO_IRQ) {
 		if (request_irq(irq, snd_uart16550_interrupt,
-				IRQF_DISABLED, "Serial MIDI", uart)) {
+				0, "Serial MIDI", uart)) {
 			snd_printk(KERN_WARNING
 				   "irq %d busy. Using Polling.\n", irq);
 		} else {
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 4400308..cd094ec 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -51,7 +51,6 @@
 	struct fw_unit *unit;
 	struct fw_device *device;
 	u64 audio_base;
-	struct fw_address_handler iris_handler;
 	struct snd_pcm_substream *pcm;
 	struct mutex mutex;
 	struct iso_packets_buffer buffer;
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 3fc257d..cbe6bb9 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -778,9 +778,10 @@
 {
 	struct fwspk *fwspk = dev_get_drvdata(dev);
 
-	mutex_lock(&fwspk->mutex);
 	amdtp_out_stream_pcm_abort(&fwspk->stream);
 	snd_card_disconnect(fwspk->card);
+
+	mutex_lock(&fwspk->mutex);
 	fwspk_stop_stream(fwspk);
 	mutex_unlock(&fwspk->mutex);
 
@@ -796,8 +797,8 @@
 	fcp_bus_reset(fwspk->unit);
 
 	if (cmp_connection_update(&fwspk->connection) < 0) {
-		mutex_lock(&fwspk->mutex);
 		amdtp_out_stream_pcm_abort(&fwspk->stream);
+		mutex_lock(&fwspk->mutex);
 		fwspk_stop_stream(fwspk);
 		mutex_unlock(&fwspk->mutex);
 		return;
diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c
index 3cb75bc..a87a2b5 100644
--- a/sound/isa/ad1816a/ad1816a.c
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -204,7 +204,7 @@
 
 	if (mpu_port[dev] > 0) {
 		if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-					mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED,
+					mpu_port[dev], 0, mpu_irq[dev],
 					NULL) < 0)
 			printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]);
 	}
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
index 05aef8b..177eed3 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -595,7 +595,7 @@
 		snd_ad1816a_free(chip);
 		return -EBUSY;
 	}
-	if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) {
+	if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) {
 		snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq);
 		snd_ad1816a_free(chip);
 		return -EBUSY;
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 20becc8..706effd 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -256,7 +256,6 @@
 					mpu_type,
 					mpu_port[dev], 0, 
 					mpu_irq[dev],
-					mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
 					NULL) < 0)
 			snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
 	}
diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c
index aac8dc1..b7bdbf3 100644
--- a/sound/isa/azt2320.c
+++ b/sound/isa/azt2320.c
@@ -234,8 +234,7 @@
 	if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
 		if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320,
 				mpu_port[dev], 0,
-				mpu_irq[dev], IRQF_DISABLED,
-				NULL) < 0)
+				mpu_irq[dev], NULL) < 0)
 			snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
 	}
 
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index fe79a16..dca69f8 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -597,7 +597,7 @@
 	if (mpuport[dev] != SNDRV_AUTO_PORT) {
 		if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
 					mpuport[dev], 0, mpuirq[dev],
-					IRQF_DISABLED, NULL) < 0)
+					NULL) < 0)
 			printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n",
 				mpuport[dev]);
 	}
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index cb9153e..409fa0a 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -131,7 +131,6 @@
 			mpu_irq[n] = -1;
 		if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
 					mpu_port[n], 0, mpu_irq[n],
-					mpu_irq[n] >= 0 ? IRQF_DISABLED : 0,
 					NULL) < 0)
 			dev_warn(dev, "MPU401 not detected\n");
 	}
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 999dc1e..0dbde46 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -449,8 +449,7 @@
 			mpu_irq[dev] = -1;
 		if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
 					mpu_port[dev], 0,
-					mpu_irq[dev],
-					mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0)
+					mpu_irq[dev], NULL) < 0)
 			printk(KERN_WARNING IDENT ": MPU401 not detected\n");
 	}
 
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index 0cde813..5493e9e 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -174,7 +174,7 @@
 			chip->mpu_port > 0) {
 		error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
 				chip->mpu_port, 0,
-				mpu_irq[n], IRQF_DISABLED, NULL);
+				mpu_irq[n], NULL);
 		if (error < 0)
 			return error;
 	}
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 0767620..d3eab6f 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -661,7 +661,7 @@
 		snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
 		return -EBUSY;
 	}
-	if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) {
+	if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) {
 		snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
 		return -EBUSY;
 	}
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index fb4d6b3..bf6ad0b 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1805,7 +1805,7 @@
 		return -EBUSY;
 	}
 
-	if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx",
+	if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx",
 			(void *) card)) {
 		snd_es18xx_free(card);
 		snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq);
@@ -2160,8 +2160,8 @@
 
 	if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX,
-					  mpu_port[dev], 0,
-					  irq[dev], 0, &chip->rmidi);
+					  mpu_port[dev], MPU401_INFO_IRQ_HOOK,
+					  -1, &chip->rmidi);
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c
index ee54df0..e51d324 100644
--- a/sound/isa/galaxy/galaxy.c
+++ b/sound/isa/galaxy/galaxy.c
@@ -585,8 +585,7 @@
 
 	if (mpu_port[n] >= 0) {
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-					  mpu_port[n], 0, mpu_irq[n],
-					  IRQF_DISABLED, NULL);
+					  mpu_port[n], 0, mpu_irq[n], NULL);
 		if (err < 0)
 			goto error;
 	}
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index 12eb98f..3167e5a 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -180,7 +180,7 @@
 		snd_gus_free(gus);
 		return -EBUSY;
 	}
-	if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) {
+	if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) {
 		snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq);
 		snd_gus_free(gus);
 		return -EBUSY;
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index 008e8e5..c4733c0 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -317,8 +317,7 @@
 
 	if (es1688->mpu_port >= 0x300) {
 		error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
-				es1688->mpu_port, 0,
-				mpu_irq[n], IRQF_DISABLED, NULL);
+				es1688->mpu_port, 0, mpu_irq[n], NULL);
 		if (error < 0)
 			goto out;
 	}
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
index 3e4a58b..c43faa0 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -291,7 +291,7 @@
 		goto _err;
 	}
 
-	if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) {
+	if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) {
 		snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
 		err = -EBUSY;
 		goto _err;
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index c7b80e4..5f869a3 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -684,7 +684,7 @@
 	if ((err = snd_gus_initialize(gus)) < 0)
 		return err;
 
-	if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED,
+	if (request_irq(xirq, snd_interwave_interrupt, 0,
 			"InterWave", iwcard)) {
 		snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
 		return -EBUSY;
diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c
index 91d6023..0961e2c 100644
--- a/sound/isa/msnd/msnd_pinnacle.c
+++ b/sound/isa/msnd/msnd_pinnacle.c
@@ -600,7 +600,7 @@
 					  mpu_io[0],
 					  MPU401_MODE_INPUT |
 					  MPU401_MODE_OUTPUT,
-					  mpu_irq[0], IRQF_DISABLED,
+					  mpu_irq[0],
 					  &chip->rmidi);
 		if (err < 0) {
 			printk(KERN_ERR LOGNAME
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 9b915e2..bbafb0b 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -667,7 +667,7 @@
 	err = snd_opl3sa2_detect(card);
 	if (err < 0)
 		return err;
-	err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED,
+	err = request_irq(xirq, snd_opl3sa2_interrupt, 0,
 			  "OPL3-SA2", card);
 	if (err) {
 		snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq);
@@ -707,8 +707,9 @@
 	}
 	if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2,
-					       midi_port[dev], 0,
-					       xirq, 0, &chip->rmidi)) < 0)
+					       midi_port[dev],
+					       MPU401_INFO_IRQ_HOOK, -1,
+					       &chip->rmidi)) < 0)
 			return err;
 	}
 	sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 8c24102..d94d0f3 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -1377,8 +1377,7 @@
 		rmidi = NULL;
 	else {
 		error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-				mpu_port, 0, miro->mpu_irq, IRQF_DISABLED,
-				&rmidi);
+				mpu_port, 0, miro->mpu_irq, &rmidi);
 		if (error < 0)
 			snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
 				   mpu_port);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index c35dc68..6dbbfa7 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -892,7 +892,7 @@
 #endif
 #ifdef OPTi93X
 	error = request_irq(irq, snd_opti93x_interrupt,
-			    IRQF_DISABLED, DEV_NAME" - WSS", chip);
+			    0, DEV_NAME" - WSS", chip);
 	if (error < 0) {
 		snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq);
 		return error;
@@ -914,7 +914,7 @@
 		rmidi = NULL;
 	else {
 		error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-				mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi);
+				mpu_port, 0, mpu_irq, &rmidi);
 		if (error)
 			snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
 				   mpu_port);
diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c
index 8ccbcdd..54e3c2c 100644
--- a/sound/isa/sb/jazz16.c
+++ b/sound/isa/sb/jazz16.c
@@ -322,7 +322,6 @@
 					MPU401_HW_MPU401,
 					mpu_port[dev], 0,
 					mpu_irq[dev],
-					mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
 					NULL) < 0)
 			snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n",
 					mpu_port[dev]);
diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c
index 4d1c5a3..237f8bd 100644
--- a/sound/isa/sb/sb16.c
+++ b/sound/isa/sb/sb16.c
@@ -394,8 +394,9 @@
 
 	if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SB,
-					       chip->mpu_port, 0,
-					       xirq, 0, &chip->rmidi)) < 0)
+					       chip->mpu_port,
+					       MPU401_INFO_IRQ_HOOK, -1,
+					       &chip->rmidi)) < 0)
 			return err;
 		chip->rmidi_callback = snd_mpu401_uart_interrupt;
 	}
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index eae6c1c..d2e1921 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -240,7 +240,7 @@
 	if (request_irq(irq, irq_handler,
 			(hardware == SB_HW_ALS4000 ||
 			 hardware == SB_HW_CS5530) ?
-			IRQF_SHARED : IRQF_DISABLED,
+			IRQF_SHARED : 0,
 			"SoundBlaster", (void *) chip)) {
 		snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq);
 		snd_sbdsp_free(chip);
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index 9a8bbf6..207c161 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -658,8 +658,7 @@
 		if (snd_mpu401_uart_new(card, 0,
 					MPU401_HW_MPU401,
 					mpu_port[dev], 0,
-					mpu_irq[dev], IRQF_DISABLED,
-					NULL) < 0)
+					mpu_irq[dev], NULL) < 0)
 			snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n",
 					mpu_port[dev]);
 	}
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index e2d5d2d..f2379e1 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -825,8 +825,7 @@
 	int err;
 
 	err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port,
-				  MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED,
-				  &rawmidi);
+				  MPU401_INFO_INTEGRATED, irq, &rawmidi);
 	if (err == 0) {
 		struct snd_mpu401 *mpu = rawmidi->private_data;
 		mpu->open_input = mpu401_open;
diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c
index 711670e..8714297 100644
--- a/sound/isa/wavefront/wavefront.c
+++ b/sound/isa/wavefront/wavefront.c
@@ -418,7 +418,7 @@
 		return -EBUSY;
 	}
 	if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt,
-			IRQF_DISABLED, "ICS2115", acard)) {
+			0, "ICS2115", acard)) {
 		snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]);
 		return -EBUSY;
 	}
@@ -449,8 +449,7 @@
 	if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) {
 		err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232,
 					  cs4232_mpu_port[dev], 0,
-					  cs4232_mpu_irq[dev], IRQF_DISABLED,
-					  NULL);
+					  cs4232_mpu_irq[dev], NULL);
 		if (err < 0) {
 			snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n");
 			return err;
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 2a42cc3..7277c5b 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1833,7 +1833,7 @@
 	}
 	chip->cport = cport;
 	if (!(hwshare & WSS_HWSHARE_IRQ))
-		if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED,
+		if (request_irq(irq, snd_wss_interrupt, 0,
 				"WSS", (void *) chip)) {
 			snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq);
 			snd_wss_free(chip);
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index 446cf97..7567ebd 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -465,13 +465,13 @@
 
 	flags = claim_dma_lock();
 	if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX,
-			"AC97 TX", au1000_dma_interrupt, IRQF_DISABLED,
+			"AC97 TX", au1000_dma_interrupt, 0,
 			au1000->stream[PLAYBACK])) < 0) {
 		release_dma_lock(flags);
 		return -EBUSY;
 	}
 	if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX,
-			"AC97 RX", au1000_dma_interrupt, IRQF_DISABLED,
+			"AC97 RX", au1000_dma_interrupt, 0,
 			au1000->stream[CAPTURE])) < 0){
 		release_dma_lock(flags);
 		return -EBUSY;
diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
index 8f7d175..6f13ab4 100644
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@
 
 	if (pcm_channels & 2)
 	{
-		foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
-		arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
+		foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
+		arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
 	}
 	else
 	{
-		foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
-		arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
+		foo = (PIT_TICK_RATE + (arg / 2)) / arg;
+		arg = (PIT_TICK_RATE + (foo / 2)) / foo;
 	}
 
 	pcm_speed = arg;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 9b800ce..2fc0624 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -673,7 +673,8 @@
 
 	if (pss_cdrom_port == -1) {	/* If cdrom port enablation wasn't requested */
 		printk(KERN_INFO "PSS: CDROM port not enabled.\n");
-	} else if (check_region(pss_cdrom_port, 2)) {
+	} else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
+		pss_cdrom_port = -1;
 		printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
 	} else {
 		set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1232,7 +1233,8 @@
 		if(pssmpu)
 			unload_pss_mpu(&cfg_mpu);
 		unload_pss(&cfg);
-	}
+	} else if (pss_cdrom_port != -1)
+		release_region(pss_cdrom_port, 2);
 
 	if(!pss_keep_settings)	/* Keep hardware settings if asked */
 	{
diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c
index 48cda6c..8021c85 100644
--- a/sound/oss/sound_timer.c
+++ b/sound/oss/sound_timer.c
@@ -320,7 +320,7 @@
 	n = sound_alloc_timerdev();
 	if (n == -1)
 		n = 0;		/* Overwrite the system timer */
-	strcpy(sound_timer.info.name, name);
+	strlcpy(sound_timer.info.name, name, sizeof(sound_timer.info.name));
 	sound_timer_devs[n] = &sound_timer;
 }
 EXPORT_SYMBOL(sound_timer_init);
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5b..8816804 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
 # ALSA PCI drivers
 
+config SND_TEA575X
+	tristate
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
 menuconfig SND_PCI
 	bool "PCI sound devices"
 	depends on PCI
@@ -563,11 +568,6 @@
 	  FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
 	  SF64-PCR) into the snd-fm801 driver.
 
-config SND_TEA575X
-	tristate
-	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
 source "sound/pci/hda/Kconfig"
 
 config SND_HDSP
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1..a872d0a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@
 	0x103c0944, /* HP nc6220 */
 	0x103c0934, /* HP nc8220 */
 	0x103c006d, /* HP nx9105 */
+	0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
 	0x17340088, /* FSC Scenic-W */
 	0 /* end */
 };
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index a9c1af3..0462869 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -931,8 +931,9 @@
 
 	if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000,
 					iobase + ALS4K_IOB_30_MIDI_DATA,
-					MPU401_INFO_INTEGRATED,
-					pci->irq, 0, &chip->rmidi)) < 0) {
+					MPU401_INFO_INTEGRATED |
+					MPU401_INFO_IRQ_HOOK,
+					-1, &chip->rmidi)) < 0) {
 		printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n",
 				iobase + ALS4K_IOB_30_MIDI_DATA);
 		goto out_err;
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca1..bd47521 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@
 	if (!p_cache)
 		return NULL;
 
-	p_cache->p_info =
-		kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+	p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+				  GFP_KERNEL);
 	if (!p_cache->p_info) {
 		kfree(p_cache);
 		return NULL;
 	}
-	memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = control_count;
 	p_cache->p_cache = p_dsp_control_buffer;
diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
index 0dc8d25..e6c6a0f 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -84,7 +84,7 @@
 #ifdef VORTEX_MPU401_LEGACY
 	if ((temp =
 	     snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330,
-				 0, 0, 0, &rmidi)) != 0) {
+				 MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) {
 		hwwrite(vortex->mmio, VORTEX_CTRL,
 			(hwread(vortex->mmio, VORTEX_CTRL) &
 			 ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
@@ -94,8 +94,8 @@
 	port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA);
 	if ((temp =
 	     snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port,
-				 MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO,
-				 0, 0, &rmidi)) != 0) {
+				 MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO |
+				 MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) {
 		hwwrite(vortex->mmio, VORTEX_CTRL,
 			(hwread(vortex->mmio, VORTEX_CTRL) &
 			 ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a2..d24fe42 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@
 	int err;
 
 	snd_azf3328_dbgcallenter();
-	if (dev >= SNDRV_CARDS)
-		return -ENODEV;
+	if (dev >= SNDRV_CARDS) {
+		err = -ENODEV;
+		goto out;
+	}
 	if (!enable[dev]) {
 		dev++;
-		return -ENOENT;
+		err = -ENOENT;
+		goto out;
 	}
 
 	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 	if (err < 0)
-		return err;
+		goto out;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
@@ -2649,8 +2652,9 @@
 	   since our hardware ought to be similar, thus use same ID. */
 	err = snd_mpu401_uart_new(
 		card, 0,
-		MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED,
-		pci->irq, 0, &chip->rmidi
+		MPU401_HW_AZT2320, chip->mpu_io,
+		MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+		-1, &chip->rmidi
 	);
 	if (err < 0) {
 		snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n",
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 9cf99fb..da9c732 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3228,8 +3228,9 @@
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
 					       iomidi,
 					       (integrated_midi ?
-						MPU401_INFO_INTEGRATED : 0),
-					       cm->irq, 0, &cm->rmidi)) < 0) {
+						MPU401_INFO_INTEGRATED : 0) |
+					       MPU401_INFO_IRQ_HOOK,
+					       -1, &cm->rmidi)) < 0) {
 			printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi);
 		}
 	}
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 457d211..2c86226 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -404,7 +404,7 @@
 	int err;
 	int playback_count, capture_count;
 
-	playback_count = (IEC958 == device) ? 1 : 8;
+	playback_count = (IEC958 == device) ? 1 : 256;
 	capture_count = (FRONT == device) ? 1 : 0;
 	err = snd_pcm_new(atc->card, "ctxfi", device,
 			  playback_count, capture_count, &pcm);
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index c749fa7..e134b3a 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -20,7 +20,7 @@
 #include "cthardware.h"
 #include <linux/slab.h>
 
-#define SRC_RESOURCE_NUM	64
+#define SRC_RESOURCE_NUM	256
 #define SRCIMP_RESOURCE_NUM	256
 
 static unsigned int conj_mask;
diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h
index b23adfc..e6da60e 100644
--- a/sound/pci/ctxfi/ctvmem.h
+++ b/sound/pci/ctxfi/ctvmem.h
@@ -18,7 +18,7 @@
 #ifndef CTVMEM_H
 #define CTVMEM_H
 
-#define CT_PTP_NUM	1	/* num of device page table pages */
+#define CT_PTP_NUM	4	/* num of device page table pages */
 
 #include <linux/mutex.h>
 #include <linux/list.h>
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 622bace..e22b8e2 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1146,6 +1146,11 @@
 		kfree(epcm);
 		return err;
 	}
+	err = snd_pcm_hw_rule_noresample(runtime, 48000);
+	if (err < 0) {
+		kfree(epcm);
+		return err;
+	}
 	mix = &emu->pcm_mixer[substream->number];
 	for (i = 0; i < 4; i++)
 		mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i;
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 26a5a2f..718a264 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1854,8 +1854,9 @@
 		}
 	}
 	if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-				chip->mpu_port, MPU401_INFO_INTEGRATED,
-				chip->irq, 0, &chip->rmidi) < 0) {
+				chip->mpu_port,
+				MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+				-1, &chip->rmidi) < 0) {
 		printk(KERN_ERR "es1938: unable to initialize MPU-401\n");
 	} else {
 		// this line is vital for MIDI interrupt handling on ess-solo1
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 99ea932..407e4ab 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2843,8 +2843,9 @@
 	if (enable_mpu[dev]) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
 					       chip->io_port + ESM_MPU401_PORT,
-					       MPU401_INFO_INTEGRATED,
-					       chip->irq, 0, &chip->rmidi)) < 0) {
+					       MPU401_INFO_INTEGRATED |
+					       MPU401_INFO_IRQ_HOOK,
+					       -1, &chip->rmidi)) < 0) {
 			printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n");
 		}
 	}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f9123f0..136f723 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@
 module_param_array(tea575x_tuner, int, NULL, 0444);
 MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
 
+#define TUNER_DISABLED		(1<<3)
 #define TUNER_ONLY		(1<<4)
 #define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
 
@@ -728,11 +729,14 @@
 	{ .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" },
 };
 
+#define get_tea575x_gpio(chip) \
+	(&snd_fm801_tea575x_gpios[((chip)->tea575x_tuner & TUNER_TYPE_MASK) - 1])
+
 static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins)
 {
 	struct fm801 *chip = tea->private_data;
 	unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
-	struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
+	struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
 
 	reg &= ~(FM801_GPIO_GP(gpio.data) |
 		 FM801_GPIO_GP(gpio.clk) |
@@ -750,7 +754,7 @@
 {
 	struct fm801 *chip = tea->private_data;
 	unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
-	struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
+	struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
 
 	return  (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 |
 		(reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0;
@@ -760,7 +764,7 @@
 {
 	struct fm801 *chip = tea->private_data;
 	unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
-	struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
+	struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
 
 	/* use GPIO lines and set write enable bit */
 	reg |= FM801_GPIO_GS(gpio.data) |
@@ -1150,7 +1154,8 @@
 
       __end_hw:
 #ifdef CONFIG_SND_FM801_TEA575X_BOOL
-	snd_tea575x_exit(&chip->tea);
+	if (!(chip->tea575x_tuner & TUNER_DISABLED))
+		snd_tea575x_exit(&chip->tea);
 #endif
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1236,7 +1241,6 @@
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
 		if (snd_tea575x_init(&chip->tea)) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
 			return -ENODEV;
 		}
 	} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1245,17 +1249,19 @@
 			chip->tea575x_tuner = tea575x_tuner;
 			if (!snd_tea575x_init(&chip->tea)) {
 				snd_printk(KERN_INFO "detected TEA575x radio type %s\n",
-					snd_fm801_tea575x_gpios[tea575x_tuner - 1].name);
+					   get_tea575x_gpio(chip)->name);
 				break;
 			}
 		}
 		if (tea575x_tuner == 4) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
-			return -ENODEV;
+			chip->tea575x_tuner = TUNER_DISABLED;
 		}
 	}
-	strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+	if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+		strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name,
+			sizeof(chip->tea.card));
+	}
 #endif
 
 	*rchip = chip;
@@ -1306,8 +1312,9 @@
 	}
 	if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801,
 				       FM801_REG(chip, MPU401_DATA),
-				       MPU401_INFO_INTEGRATED,
-				       chip->irq, 0, &chip->rmidi)) < 0) {
+				       MPU401_INFO_INTEGRATED |
+				       MPU401_INFO_IRQ_HOOK,
+				       -1, &chip->rmidi)) < 0) {
 		snd_card_free(card);
 		return err;
 	}
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 87365d5..f928d66 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -6,6 +6,9 @@
 snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
 snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
 
+# for trace-points
+CFLAGS_hda_codec.o := -I$(src)
+
 snd-hda-codec-realtek-objs :=	patch_realtek.o
 snd-hda-codec-cmedia-objs :=	patch_cmedia.o
 snd-hda-codec-analog-objs :=	patch_analog.o
diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c
index 21ec2cb..3b5170b 100644
--- a/sound/pci/hda/alc260_quirks.c
+++ b/sound/pci/hda/alc260_quirks.c
@@ -7,9 +7,6 @@
 enum {
 	ALC260_AUTO,
 	ALC260_BASIC,
-	ALC260_HP,
-	ALC260_HP_DC7600,
-	ALC260_HP_3013,
 	ALC260_FUJITSU_S702X,
 	ALC260_ACER,
 	ALC260_WILL,
@@ -142,8 +139,6 @@
 /* Mixer combinations
  *
  * basic: base_output + input + pc_beep + capture
- * HP: base_output + input + capture_alt
- * HP_3013: hp_3013 + input + capture
  * fujitsu: fujitsu + capture
  * acer: acer + capture
  */
@@ -170,145 +165,6 @@
 	{ } /* end */
 };
 
-/* update HP, line and mono out pins according to the master switch */
-static void alc260_hp_master_update(struct hda_codec *codec)
-{
-	update_speakers(codec);
-}
-
-static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct alc_spec *spec = codec->spec;
-	*ucontrol->value.integer.value = !spec->master_mute;
-	return 0;
-}
-
-static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct alc_spec *spec = codec->spec;
-	int val = !*ucontrol->value.integer.value;
-
-	if (val == spec->master_mute)
-		return 0;
-	spec->master_mute = val;
-	alc260_hp_master_update(codec);
-	return 1;
-}
-
-static const struct snd_kcontrol_new alc260_hp_output_mixer[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
-		.info = snd_ctl_boolean_mono_info,
-		.get = alc260_hp_master_sw_get,
-		.put = alc260_hp_master_sw_put,
-	},
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
-			      HDA_OUTPUT),
-	HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb alc260_hp_unsol_verbs[] = {
-	{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{},
-};
-
-static void alc260_hp_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x0f;
-	spec->autocfg.speaker_pins[0] = 0x10;
-	spec->autocfg.speaker_pins[1] = 0x11;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
-		.info = snd_ctl_boolean_mono_info,
-		.get = alc260_hp_master_sw_get,
-		.put = alc260_hp_master_sw_put,
-	},
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static void alc260_hp_3013_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x10;
-	spec->autocfg.speaker_pins[1] = 0x11;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls alc260_dc7600_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
-	HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb alc260_hp_3013_unsol_verbs[] = {
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{},
-};
-
-static void alc260_hp_3012_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x10;
-	spec->autocfg.speaker_pins[0] = 0x0f;
-	spec->autocfg.speaker_pins[1] = 0x11;
-	spec->autocfg.speaker_pins[2] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
 /* Fujitsu S702x series laptops.  ALC260 pin usage: Mic/Line jack = 0x12,
  * HP jack = 0x14, CD audio =  0x16, internal speaker = 0x10.
  */
@@ -480,106 +336,6 @@
 	{ }
 };
 
-#if 0 /* should be identical with alc260_init_verbs? */
-static const struct hda_verb alc260_hp_init_verbs[] = {
-	/* Headphone and output */
-	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
-	/* mono output */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-	/* Mic1 (rear panel) pin widget for input and vref at 80% */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	/* Mic2 (front panel) pin widget for input and vref at 80% */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	/* Line In pin widget for input */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	/* Line-2 pin widget for output */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-	/* CD pin widget for input */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	/* unmute amp left and right */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
-	/* set connection select to line in (default select for this ADC) */
-	{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
-	/* unmute Line-Out mixer amp left and right (volume = 0) */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* mute pin widget amp left and right (no gain on this amp) */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-	/* unmute HP mixer amp left and right (volume = 0) */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* mute pin widget amp left and right (no gain on this amp) */
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
-	 * Line In 2 = 0x03
-	 */
-	/* mute analog inputs */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
-	/* Unmute Front out path */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	/* Unmute Headphone out path */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	/* Unmute Mono out path */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	{ }
-};
-#endif
-
-static const struct hda_verb alc260_hp_3013_init_verbs[] = {
-	/* Line out and output */
-	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-	/* mono output */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-	/* Mic1 (rear panel) pin widget for input and vref at 80% */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	/* Mic2 (front panel) pin widget for input and vref at 80% */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	/* Line In pin widget for input */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	/* Headphone pin widget for output */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
-	/* CD pin widget for input */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	/* unmute amp left and right */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
-	/* set connection select to line in (default select for this ADC) */
-	{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
-	/* unmute Line-Out mixer amp left and right (volume = 0) */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* mute pin widget amp left and right (no gain on this amp) */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-	/* unmute HP mixer amp left and right (volume = 0) */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* mute pin widget amp left and right (no gain on this amp) */
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
-	 * Line In 2 = 0x03
-	 */
-	/* mute analog inputs */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
-	/* Unmute Front out path */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	/* Unmute Headphone out path */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	/* Unmute Mono out path */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	{ }
-};
-
 /* Initialisation sequence for ALC260 as configured in Fujitsu S702x
  * laptops.  ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
  * audio = 0x16, internal speaker = 0x10.
@@ -1093,9 +849,6 @@
  */
 static const char * const alc260_models[ALC260_MODEL_LAST] = {
 	[ALC260_BASIC]		= "basic",
-	[ALC260_HP]		= "hp",
-	[ALC260_HP_3013]	= "hp-3013",
-	[ALC260_HP_DC7600]	= "hp-dc7600",
 	[ALC260_FUJITSU_S702X]	= "fujitsu",
 	[ALC260_ACER]		= "acer",
 	[ALC260_WILL]		= "will",
@@ -1112,15 +865,6 @@
 	SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
-	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
-	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
-	SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
-	SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
-	SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
 	SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
 	SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
 	SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
@@ -1144,54 +888,6 @@
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
 	},
-	[ALC260_HP] = {
-		.mixers = { alc260_hp_output_mixer,
-			    alc260_input_mixer },
-		.init_verbs = { alc260_init_verbs,
-				alc260_hp_unsol_verbs },
-		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
-		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
-		.adc_nids = alc260_adc_nids_alt,
-		.num_channel_mode = ARRAY_SIZE(alc260_modes),
-		.channel_mode = alc260_modes,
-		.input_mux = &alc260_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc260_hp_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC260_HP_DC7600] = {
-		.mixers = { alc260_hp_dc7600_mixer,
-			    alc260_input_mixer },
-		.init_verbs = { alc260_init_verbs,
-				alc260_hp_dc7600_verbs },
-		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
-		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
-		.adc_nids = alc260_adc_nids_alt,
-		.num_channel_mode = ARRAY_SIZE(alc260_modes),
-		.channel_mode = alc260_modes,
-		.input_mux = &alc260_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc260_hp_3012_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC260_HP_3013] = {
-		.mixers = { alc260_hp_3013_mixer,
-			    alc260_input_mixer },
-		.init_verbs = { alc260_hp_3013_init_verbs,
-				alc260_hp_3013_unsol_verbs },
-		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
-		.dac_nids = alc260_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
-		.adc_nids = alc260_adc_nids_alt,
-		.num_channel_mode = ARRAY_SIZE(alc260_modes),
-		.channel_mode = alc260_modes,
-		.input_mux = &alc260_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc260_hp_3013_setup,
-		.init_hook = alc_inithook,
-	},
 	[ALC260_FUJITSU_S702X] = {
 		.mixers = { alc260_fujitsu_mixer },
 		.init_verbs = { alc260_fujitsu_init_verbs },
diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c
index 8d2097d..7894b2b 100644
--- a/sound/pci/hda/alc262_quirks.c
+++ b/sound/pci/hda/alc262_quirks.c
@@ -10,13 +10,7 @@
 	ALC262_HIPPO,
 	ALC262_HIPPO_1,
 	ALC262_FUJITSU,
-	ALC262_HP_BPC,
-	ALC262_HP_BPC_D7000_WL,
-	ALC262_HP_BPC_D7000_WF,
-	ALC262_HP_TC_T5735,
-	ALC262_HP_RP5700,
 	ALC262_BENQ_ED8,
-	ALC262_SONY_ASSAMD,
 	ALC262_BENQ_T31,
 	ALC262_ULTRA,
 	ALC262_LENOVO_3000,
@@ -66,163 +60,30 @@
 	{ } /* end */
 };
 
-/* update HP, line and mono-out pins according to the master switch */
-#define alc262_hp_master_update		alc260_hp_master_update
-
-static void alc262_hp_bpc_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x1b;
-	spec->autocfg.speaker_pins[0] = 0x16;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static void alc262_hp_wildwest_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x16;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-#define alc262_hp_master_sw_get		alc260_hp_master_sw_get
-#define alc262_hp_master_sw_put		alc260_hp_master_sw_put
-
-#define ALC262_HP_MASTER_SWITCH					\
-	{							\
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,		\
-		.name = "Master Playback Switch",		\
-		.info = snd_ctl_boolean_mono_info,		\
-		.get = alc262_hp_master_sw_get,			\
-		.put = alc262_hp_master_sw_put,			\
-	}, \
-	{							\
-		.iface = NID_MAPPING,				\
-		.name = "Master Playback Switch",		\
-		.private_value = 0x15 | (0x16 << 8) | (0x1b << 16),	\
-	}
-
-
-static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
-	ALC262_HP_MASTER_SWITCH,
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
-			      HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
-			    HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
-	ALC262_HP_MASTER_SWITCH,
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
-			      HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
-			    HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
-	HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hp_t5735_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb alc262_hp_t5735_verbs[] = {
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{ }
-};
-
-static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb alc262_hp_rp5700_verbs[] = {
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
-	{}
-};
-
-static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
-	.num_items = 1,
-	.items = {
-		{ "Line", 0x1 },
-	},
-};
-
 /* bind hp and internal speaker mute (with plug check) as master switch */
-#define alc262_hippo_master_update	alc262_hp_master_update
-#define alc262_hippo_master_sw_get	alc262_hp_master_sw_get
-#define alc262_hippo_master_sw_put	alc262_hp_master_sw_put
+
+static int alc262_hippo_master_sw_get(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	*ucontrol->value.integer.value = !spec->master_mute;
+	return 0;
+}
+
+static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	int val = !*ucontrol->value.integer.value;
+
+	if (val == spec->master_mute)
+		return 0;
+	spec->master_mute = val;
+	update_outputs(codec);
+	return 1;
+}
 
 #define ALC262_HIPPO_MASTER_SWITCH				\
 	{							\
@@ -239,6 +100,9 @@
 			     (SUBDEV_SPEAKER(0) << 16), \
 	}
 
+#define alc262_hp_master_sw_get		alc262_hippo_master_sw_get
+#define alc262_hp_master_sw_put		alc262_hippo_master_sw_put
+
 static const struct snd_kcontrol_new alc262_hippo_mixer[] = {
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -279,8 +143,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc262_hippo1_setup(struct hda_codec *codec)
@@ -289,8 +152,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 
@@ -353,8 +215,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 
@@ -496,8 +357,7 @@
 	spec->ext_mic_pin = 0x18;
 	spec->int_mic_pin = 0x12;
 	spec->auto_mic = 1;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN);
 }
 
 /*
@@ -571,27 +431,6 @@
 	},
 };
 
-static const struct hda_input_mux alc262_HP_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x1 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
-		{ "AUX IN", 0x6 },
-	},
-};
-
-static const struct hda_input_mux alc262_HP_D7000_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x2 },
-		{ "Line", 0x1 },
-		{ "CD", 0x4 },
-	},
-};
-
 static void alc262_fujitsu_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -599,8 +438,7 @@
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.hp_pins[1] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 /* bind volumes of both NID 0x0c and 0x0d */
@@ -646,8 +484,7 @@
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
@@ -752,8 +589,8 @@
 	mute = 0;
 	/* auto-mute only when HP is used as HP */
 	if (!spec->cur_mux[0]) {
-		spec->jack_present = snd_hda_jack_detect(codec, 0x15);
-		if (spec->jack_present)
+		spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15);
+		if (spec->hp_jack_present)
 			mute = HDA_AMP_MUTE;
 	}
 	/* mute/unmute internal speaker */
@@ -817,206 +654,6 @@
 	{ } /* end */
 };
 
-static const struct hda_verb alc262_HP_BPC_init_verbs[] = {
-	/*
-	 * Unmute ADC0-2 and set the default input to mic-in
-	 */
-	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
-	 * mixer widget
-	 * Note: PASD motherboards uses the Line In 2 as the input for
-	 * front panel mic (mic 2)
-	 */
-	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-
-	/*
-	 * Set up output mixers (0x0c - 0x0e)
-	 */
-	/* set vol=0 to output mixers */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	/* set up input amps for analog loopback */
-	/* Amp Indices: DAC = 0, mixer = 1 */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-        {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-        {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-
-
-	/* FIXME: use matrix-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
-	/* Input mixer1: only unmute Mic */
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
-	/* Input mixer2 */
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
-	/* Input mixer3 */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
-
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-
-	{ }
-};
-
-static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
-	/*
-	 * Unmute ADC0-2 and set the default input to mic-in
-	 */
-	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
-	 * mixer widget
-	 * Note: PASD motherboards uses the Line In 2 as the input for front
-	 * panel mic (mic 2)
-	 */
-	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/*
-	 * Set up output mixers (0x0c - 0x0e)
-	 */
-	/* set vol=0 to output mixers */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	/* set up input amps for analog loopback */
-	/* Amp Indices: DAC = 0, mixer = 1 */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },	/* HP */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },	/* Mono */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },	/* rear MIC */
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },	/* Line in */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },	/* Front MIC */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },	/* Line out */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },	/* CD in */
-
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-
-	/* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-
-	/* FIXME: use matrix-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
-	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
-        /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))},  */
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
-	/* Input mixer2 */
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
-        /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
-	/* Input mixer3 */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
-        /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
-
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-
-	{ }
-};
-
 static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
 
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },	/* Front Speaker */
@@ -1042,13 +679,8 @@
 	[ALC262_HIPPO]		= "hippo",
 	[ALC262_HIPPO_1]	= "hippo_1",
 	[ALC262_FUJITSU]	= "fujitsu",
-	[ALC262_HP_BPC]		= "hp-bpc",
-	[ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
-	[ALC262_HP_TC_T5735]	= "hp-tc-t5735",
-	[ALC262_HP_RP5700]	= "hp-rp5700",
 	[ALC262_BENQ_ED8]	= "benq",
 	[ALC262_BENQ_T31]	= "benq-t31",
-	[ALC262_SONY_ASSAMD]	= "sony-assamd",
 	[ALC262_TOSHIBA_S06]	= "toshiba-s06",
 	[ALC262_TOSHIBA_RX1]	= "toshiba-rx1",
 	[ALC262_ULTRA]		= "ultra",
@@ -1061,41 +693,6 @@
 static const struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
-			   ALC262_HP_BPC),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
-			   ALC262_HP_BPC),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
-			   ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
-			   ALC262_AUTO),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
-			   ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
-	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
-	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
-	SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
-	SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
-	SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
-	SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
-	SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
-	SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
-		      ALC262_HP_TC_T5735),
-	SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
-	SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
-	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
-	SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
-	SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
-	SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
-#if 0 /* disable the quirk since model=auto works better in recent versions */
-	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
-			   ALC262_SONY_ASSAMD),
-#endif
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
 		      ALC262_TOSHIBA_RX1),
 	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
@@ -1166,68 +763,6 @@
 		.setup = alc262_fujitsu_setup,
 		.init_hook = alc_inithook,
 	},
-	[ALC262_HP_BPC] = {
-		.mixers = { alc262_HP_BPC_mixer },
-		.init_verbs = { alc262_HP_BPC_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
-		.dac_nids = alc262_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc262_modes),
-		.channel_mode = alc262_modes,
-		.input_mux = &alc262_HP_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc262_hp_bpc_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC262_HP_BPC_D7000_WF] = {
-		.mixers = { alc262_HP_BPC_WildWest_mixer },
-		.init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
-		.dac_nids = alc262_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc262_modes),
-		.channel_mode = alc262_modes,
-		.input_mux = &alc262_HP_D7000_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc262_hp_wildwest_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC262_HP_BPC_D7000_WL] = {
-		.mixers = { alc262_HP_BPC_WildWest_mixer,
-			    alc262_HP_BPC_WildWest_option_mixer },
-		.init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
-		.dac_nids = alc262_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc262_modes),
-		.channel_mode = alc262_modes,
-		.input_mux = &alc262_HP_D7000_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc262_hp_wildwest_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC262_HP_TC_T5735] = {
-		.mixers = { alc262_hp_t5735_mixer },
-		.init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs },
-		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
-		.dac_nids = alc262_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc262_modes),
-		.channel_mode = alc262_modes,
-		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc262_hp_t5735_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC262_HP_RP5700] = {
-		.mixers = { alc262_hp_rp5700_mixer },
-		.init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
-		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
-		.dac_nids = alc262_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc262_modes),
-		.channel_mode = alc262_modes,
-		.input_mux = &alc262_hp_rp5700_capture_source,
-        },
 	[ALC262_BENQ_ED8] = {
 		.mixers = { alc262_base_mixer },
 		.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
@@ -1238,19 +773,6 @@
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 	},
-	[ALC262_SONY_ASSAMD] = {
-		.mixers = { alc262_sony_mixer },
-		.init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
-		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
-		.dac_nids = alc262_dac_nids,
-		.hp_nid = 0x02,
-		.num_channel_mode = ARRAY_SIZE(alc262_modes),
-		.channel_mode = alc262_modes,
-		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc262_hippo_setup,
-		.init_hook = alc_inithook,
-	},
 	[ALC262_BENQ_T31] = {
 		.mixers = { alc262_benq_t31_mixer },
 		.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
deleted file mode 100644
index be58bf2..0000000
--- a/sound/pci/hda/alc268_quirks.c
+++ /dev/null
@@ -1,636 +0,0 @@
-/*
- * ALC267/ALC268 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC268 models */
-enum {
-	ALC268_AUTO,
-	ALC267_QUANTA_IL1,
-	ALC268_3ST,
-	ALC268_TOSHIBA,
-	ALC268_ACER,
-	ALC268_ACER_DMIC,
-	ALC268_ACER_ASPIRE_ONE,
-	ALC268_DELL,
-	ALC268_ZEPTO,
-#ifdef CONFIG_SND_DEBUG
-	ALC268_TEST,
-#endif
-	ALC268_MODEL_LAST /* last tag */
-};
-
-/*
- *  ALC268 channel source setting (2 channel)
- */
-#define ALC268_DIGOUT_NID	ALC880_DIGOUT_NID
-#define alc268_modes		alc260_modes
-
-static const hda_nid_t alc268_dac_nids[2] = {
-	/* front, hp */
-	0x02, 0x03
-};
-
-static const hda_nid_t alc268_adc_nids[2] = {
-	/* ADC0-1 */
-	0x08, 0x07
-};
-
-static const hda_nid_t alc268_adc_nids_alt[1] = {
-	/* ADC0 */
-	0x08
-};
-
-static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
-
-static const struct snd_kcontrol_new alc268_base_mixer[] = {
-	/* output mixer control */
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct snd_kcontrol_new alc268_toshiba_mixer[] = {
-	/* output mixer control */
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
-	ALC262_HIPPO_MASTER_SWITCH,
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct hda_verb alc268_eapd_verbs[] = {
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
-/* Toshiba specific */
-static const struct hda_verb alc268_toshiba_verbs[] = {
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{ } /* end */
-};
-
-/* Acer specific */
-/* bind volumes of both NID 0x02 and 0x03 */
-static const struct hda_bind_ctls alc268_acer_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static void alc268_acer_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-#define alc268_acer_master_sw_get	alc262_hp_master_sw_get
-#define alc268_acer_master_sw_put	alc262_hp_master_sw_put
-
-static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
-	/* output mixer control */
-	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x15,
-		.info = snd_ctl_boolean_mono_info,
-		.get = alc268_acer_master_sw_get,
-		.put = alc268_acer_master_sw_put,
-	},
-	HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct snd_kcontrol_new alc268_acer_mixer[] = {
-	/* output mixer control */
-	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
-		.info = snd_ctl_boolean_mono_info,
-		.get = alc268_acer_master_sw_get,
-		.put = alc268_acer_master_sw_put,
-	},
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
-	/* output mixer control */
-	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
-		.info = snd_ctl_boolean_mono_info,
-		.get = alc268_acer_master_sw_get,
-		.put = alc268_acer_master_sw_put,
-	},
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct hda_verb alc268_acer_aspire_one_verbs[] = {
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
-	{ }
-};
-
-static const struct hda_verb alc268_acer_verbs[] = {
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{ }
-};
-
-/* unsolicited event for HP jack sensing */
-#define alc268_toshiba_setup		alc262_hippo_setup
-
-static void alc268_acer_lc_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x12;
-	spec->auto_mic = 1;
-}
-
-static const struct snd_kcontrol_new alc268_dell_mixer[] = {
-	/* output mixer control */
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct hda_verb alc268_dell_verbs[] = {
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
-	{ }
-};
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc268_dell_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct hda_verb alc267_quanta_il1_verbs[] = {
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
-	{ }
-};
-
-static void alc267_quanta_il1_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc268_base_init_verbs[] = {
-	/* Unmute DAC0-1 and set vol = 0 */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	/*
-	 * Set up output mixers (0x0c - 0x0e)
-	 */
-	/* set vol=0 to output mixers */
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
-	/* set PCBEEP vol = 0, mute connections */
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
-	/* Unmute Selector 23h,24h and set the default input to mic-in */
-
-	{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	{ }
-};
-
-/* only for model=test */
-#ifdef CONFIG_SND_DEBUG
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc268_volume_init_verbs[] = {
-	/* set output DAC */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{ }
-};
-#endif /* CONFIG_SND_DEBUG */
-
-static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
-	_DEFINE_CAPSRC(1),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc268_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
-	_DEFINE_CAPSRC(2),
-	{ } /* end */
-};
-
-static const struct hda_input_mux alc268_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x1 },
-		{ "Line", 0x2 },
-		{ "CD", 0x3 },
-	},
-};
-
-static const struct hda_input_mux alc268_acer_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-		{ "Line", 0x2 },
-	},
-};
-
-static const struct hda_input_mux alc268_acer_dmic_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x6 },
-		{ "Line", 0x2 },
-	},
-};
-
-#ifdef CONFIG_SND_DEBUG
-static const struct snd_kcontrol_new alc268_test_mixer[] = {
-	/* Volume widgets */
-	HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
-	HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
-	HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
-	HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
-	HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
-	/* The below appears problematic on some hardwares */
-	/*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
-	HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
-
-	/* Modes for retasking pin widgets */
-	ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
-	ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
-	ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
-	ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
-
-	/* Controls for GPIO pins, assuming they are configured as outputs */
-	ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
-	ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
-	ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
-	ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
-
-	/* Switches to allow the digital SPDIF output pin to be enabled.
-	 * The ALC268 does not have an SPDIF input.
-	 */
-	ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
-
-	/* A switch allowing EAPD to be enabled.  Some laptops seem to use
-	 * this output to turn on an external amplifier.
-	 */
-	ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
-	ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
-
-	{ } /* end */
-};
-#endif
-
-/*
- * configuration and preset
- */
-static const char * const alc268_models[ALC268_MODEL_LAST] = {
-	[ALC267_QUANTA_IL1]	= "quanta-il1",
-	[ALC268_3ST]		= "3stack",
-	[ALC268_TOSHIBA]	= "toshiba",
-	[ALC268_ACER]		= "acer",
-	[ALC268_ACER_DMIC]	= "acer-dmic",
-	[ALC268_ACER_ASPIRE_ONE]	= "acer-aspire",
-	[ALC268_DELL]		= "dell",
-	[ALC268_ZEPTO]		= "zepto",
-#ifdef CONFIG_SND_DEBUG
-	[ALC268_TEST]		= "test",
-#endif
-	[ALC268_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk alc268_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
-	SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
-	SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
-	SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
-	SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
-	SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
-						ALC268_ACER_ASPIRE_ONE),
-	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
-	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
-	SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
-			"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
-	/* almost compatible with toshiba but with optional digital outs;
-	 * auto-probing seems working fine
-	 */
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
-			   ALC268_AUTO),
-	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
-	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
-	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
-	{}
-};
-
-/* Toshiba laptops have no unique PCI SSID but only codec SSID */
-static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
-	SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
-	SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
-			   ALC268_TOSHIBA),
-	{}
-};
-
-static const struct alc_config_preset alc268_presets[] = {
-	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc267_quanta_il1_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc267_quanta_il1_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC268_3ST] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
-		.init_verbs = { alc268_base_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-                .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-                .adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x03,
-		.dig_out_nid = ALC268_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.input_mux = &alc268_capture_source,
-	},
-	[ALC268_TOSHIBA] = {
-		.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_toshiba_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.input_mux = &alc268_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc268_toshiba_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_acer_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x02,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.input_mux = &alc268_acer_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc268_acer_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC268_ACER_DMIC] = {
-		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_acer_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x02,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.input_mux = &alc268_acer_dmic_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc268_acer_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC268_ACER_ASPIRE_ONE] = {
-		.mixers = { alc268_acer_aspire_one_mixer,
-			    alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_acer_aspire_one_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc268_acer_lc_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC268_DELL] = {
-		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_dell_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x02,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc268_dell_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC268_ZEPTO] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_toshiba_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x03,
-		.dig_out_nid = ALC268_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.input_mux = &alc268_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc268_toshiba_setup,
-		.init_hook = alc_inithook,
-	},
-#ifdef CONFIG_SND_DEBUG
-	[ALC268_TEST] = {
-		.mixers = { alc268_test_mixer, alc268_capture_mixer },
-		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
-				alc268_volume_init_verbs,
-				alc268_beep_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
-		.dac_nids = alc268_dac_nids,
-		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
-		.adc_nids = alc268_adc_nids_alt,
-		.capsrc_nids = alc268_capsrc_nids,
-		.hp_nid = 0x03,
-		.dig_out_nid = ALC268_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc268_modes),
-		.channel_mode = alc268_modes,
-		.input_mux = &alc268_capture_source,
-	},
-#endif
-};
-
diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
deleted file mode 100644
index 14fdcf2..0000000
--- a/sound/pci/hda/alc269_quirks.c
+++ /dev/null
@@ -1,681 +0,0 @@
-/*
- * ALC269/ALC270/ALC275/ALC276 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC269 models */
-enum {
-	ALC269_AUTO,
-	ALC269_BASIC,
-	ALC269_QUANTA_FL1,
-	ALC269_AMIC,
-	ALC269_DMIC,
-	ALC269VB_AMIC,
-	ALC269VB_DMIC,
-	ALC269_FUJITSU,
-	ALC269_LIFEBOOK,
-	ALC271_ACER,
-	ALC269_MODEL_LAST /* last tag */
-};
-
-/*
- *  ALC269 channel source setting (2 channel)
- */
-#define ALC269_DIGOUT_NID	ALC880_DIGOUT_NID
-
-#define alc269_dac_nids		alc260_dac_nids
-
-static const hda_nid_t alc269_adc_nids[1] = {
-	/* ADC1 */
-	0x08,
-};
-
-static const hda_nid_t alc269_capsrc_nids[1] = {
-	0x23,
-};
-
-static const hda_nid_t alc269vb_adc_nids[1] = {
-	/* ADC1 */
-	0x09,
-};
-
-static const hda_nid_t alc269vb_capsrc_nids[1] = {
-	0x22,
-};
-
-#define alc269_modes		alc260_modes
-#define alc269_capture_source	alc880_lg_lw_capture_source
-
-static const struct snd_kcontrol_new alc269_base_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
-	/* output mixer control */
-	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = alc268_acer_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct snd_kcontrol_new alc269_lifebook_mixer[] = {
-	/* output mixer control */
-	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = alc268_acer_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct snd_kcontrol_new alc269_laptop_mixer[] = {
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc269_asus_mixer[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-/* capture mixer elements */
-static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-/* FSC amilo */
-#define alc269_fujitsu_mixer	alc269_laptop_mixer
-
-static const struct hda_verb alc269_quanta_fl1_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{ }
-};
-
-static const struct hda_verb alc269_lifebook_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{ }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
-{
-	alc_hp_automute(codec);
-
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_COEF_INDEX, 0x0c);
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_PROC_COEF, 0x680);
-
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_COEF_INDEX, 0x0c);
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_PROC_COEF, 0x480);
-}
-
-#define alc269_lifebook_speaker_automute \
-	alc269_quanta_fl1_speaker_automute
-
-static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
-{
-	unsigned int present_laptop;
-	unsigned int present_dock;
-
-	present_laptop	= snd_hda_jack_detect(codec, 0x18);
-	present_dock	= snd_hda_jack_detect(codec, 0x1b);
-
-	/* Laptop mic port overrides dock mic port, design decision */
-	if (present_dock)
-		snd_hda_codec_write(codec, 0x23, 0,
-				AC_VERB_SET_CONNECT_SEL, 0x3);
-	if (present_laptop)
-		snd_hda_codec_write(codec, 0x23, 0,
-				AC_VERB_SET_CONNECT_SEL, 0x0);
-	if (!present_dock && !present_laptop)
-		snd_hda_codec_write(codec, 0x23, 0,
-				AC_VERB_SET_CONNECT_SEL, 0x1);
-}
-
-static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
-				    unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC_HP_EVENT:
-		alc269_quanta_fl1_speaker_automute(codec);
-		break;
-	case ALC_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
-
-static void alc269_lifebook_unsol_event(struct hda_codec *codec,
-					unsigned int res)
-{
-	if ((res >> 26) == ALC_HP_EVENT)
-		alc269_lifebook_speaker_automute(codec);
-	if ((res >> 26) == ALC_MIC_EVENT)
-		alc269_lifebook_mic_autoswitch(codec);
-}
-
-static void alc269_quanta_fl1_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
-{
-	alc269_quanta_fl1_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
-
-static void alc269_lifebook_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.hp_pins[1] = 0x1a;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-}
-
-static void alc269_lifebook_init_hook(struct hda_codec *codec)
-{
-	alc269_lifebook_speaker_automute(codec);
-	alc269_lifebook_mic_autoswitch(codec);
-}
-
-static const struct hda_verb alc269_laptop_dmic_init_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc269_laptop_amic_init_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x22, AC_VERB_SET_CONNECT_SEL, 0x06},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc271_acer_dmic_verbs[] = {
-	{0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
-	{0x20, AC_VERB_SET_PROC_COEF, 0x4000},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x22, AC_VERB_SET_CONNECT_SEL, 6},
-	{ }
-};
-
-static void alc269_laptop_amic_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-static void alc269_laptop_dmic_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x12;
-	spec->auto_mic = 1;
-}
-
-static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x12;
-	spec->auto_mic = 1;
-}
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc269_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/*
-	 * Set up output mixers (0x02 - 0x03)
-	 */
-	/* set vol=0 to output mixers */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	/* set up input amps for analog loopback */
-	/* Amp Indices: DAC = 0, mixer = 1 */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* FIXME: use Mux-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
-	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
-	{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	/* set EAPD */
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
-static const struct hda_verb alc269vb_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/*
-	 * Set up output mixers (0x02 - 0x03)
-	 */
-	/* set vol=0 to output mixers */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	/* set up input amps for analog loopback */
-	/* Amp Indices: DAC = 0, mixer = 1 */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* FIXME: use Mux-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
-	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
-	{0x22, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	/* set EAPD */
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
-/*
- * configuration and preset
- */
-static const char * const alc269_models[ALC269_MODEL_LAST] = {
-	[ALC269_BASIC]			= "basic",
-	[ALC269_QUANTA_FL1]		= "quanta",
-	[ALC269_AMIC]			= "laptop-amic",
-	[ALC269_DMIC]			= "laptop-dmic",
-	[ALC269_FUJITSU]		= "fujitsu",
-	[ALC269_LIFEBOOK]		= "lifebook",
-	[ALC269_AUTO]			= "auto",
-};
-
-static const struct snd_pci_quirk alc269_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
-	SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER),
-	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
-		      ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
-		      ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
-		      ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
-	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
-	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
-	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
-	SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
-	SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC),
-	SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC),
-	SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC),
-	{}
-};
-
-static const struct alc_config_preset alc269_presets[] = {
-	[ALC269_BASIC] = {
-		.mixers = { alc269_base_mixer },
-		.init_verbs = { alc269_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.input_mux = &alc269_capture_source,
-	},
-	[ALC269_QUANTA_FL1] = {
-		.mixers = { alc269_quanta_fl1_mixer },
-		.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.input_mux = &alc269_capture_source,
-		.unsol_event = alc269_quanta_fl1_unsol_event,
-		.setup = alc269_quanta_fl1_setup,
-		.init_hook = alc269_quanta_fl1_init_hook,
-	},
-	[ALC269_AMIC] = {
-		.mixers = { alc269_laptop_mixer },
-		.cap_mixer = alc269_laptop_analog_capture_mixer,
-		.init_verbs = { alc269_init_verbs,
-				alc269_laptop_amic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc269_laptop_amic_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC269_DMIC] = {
-		.mixers = { alc269_laptop_mixer },
-		.cap_mixer = alc269_laptop_digital_capture_mixer,
-		.init_verbs = { alc269_init_verbs,
-				alc269_laptop_dmic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc269_laptop_dmic_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC269VB_AMIC] = {
-		.mixers = { alc269vb_laptop_mixer },
-		.cap_mixer = alc269vb_laptop_analog_capture_mixer,
-		.init_verbs = { alc269vb_init_verbs,
-				alc269vb_laptop_amic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc269vb_laptop_amic_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC269VB_DMIC] = {
-		.mixers = { alc269vb_laptop_mixer },
-		.cap_mixer = alc269vb_laptop_digital_capture_mixer,
-		.init_verbs = { alc269vb_init_verbs,
-				alc269vb_laptop_dmic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc269vb_laptop_dmic_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC269_FUJITSU] = {
-		.mixers = { alc269_fujitsu_mixer },
-		.cap_mixer = alc269_laptop_digital_capture_mixer,
-		.init_verbs = { alc269_init_verbs,
-				alc269_laptop_dmic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc269_laptop_dmic_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC269_LIFEBOOK] = {
-		.mixers = { alc269_lifebook_mixer },
-		.init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.hp_nid = 0x03,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.input_mux = &alc269_capture_source,
-		.unsol_event = alc269_lifebook_unsol_event,
-		.setup = alc269_lifebook_setup,
-		.init_hook = alc269_lifebook_init_hook,
-	},
-	[ALC271_ACER] = {
-		.mixers = { alc269_asus_mixer },
-		.cap_mixer = alc269vb_laptop_digital_capture_mixer,
-		.init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs },
-		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
-		.dac_nids = alc269_dac_nids,
-		.adc_nids = alc262_dmic_adc_nids,
-		.num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids),
-		.capsrc_nids = alc262_dmic_capsrc_nids,
-		.num_channel_mode = ARRAY_SIZE(alc269_modes),
-		.channel_mode = alc269_modes,
-		.input_mux = &alc269_capture_source,
-		.dig_out_nid = ALC880_DIGOUT_NID,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc269vb_laptop_dmic_setup,
-		.init_hook = alc_inithook,
-	},
-};
-
diff --git a/sound/pci/hda/alc662_quirks.c b/sound/pci/hda/alc662_quirks.c
deleted file mode 100644
index e69a6ea..0000000
--- a/sound/pci/hda/alc662_quirks.c
+++ /dev/null
@@ -1,1408 +0,0 @@
-/*
- * ALC662/ALC663/ALC665/ALC670 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC662 models */
-enum {
-	ALC662_AUTO,
-	ALC662_3ST_2ch_DIG,
-	ALC662_3ST_6ch_DIG,
-	ALC662_3ST_6ch,
-	ALC662_5ST_DIG,
-	ALC662_LENOVO_101E,
-	ALC662_ASUS_EEEPC_P701,
-	ALC662_ASUS_EEEPC_EP20,
-	ALC663_ASUS_M51VA,
-	ALC663_ASUS_G71V,
-	ALC663_ASUS_H13,
-	ALC663_ASUS_G50V,
-	ALC662_ECS,
-	ALC663_ASUS_MODE1,
-	ALC662_ASUS_MODE2,
-	ALC663_ASUS_MODE3,
-	ALC663_ASUS_MODE4,
-	ALC663_ASUS_MODE5,
-	ALC663_ASUS_MODE6,
-	ALC663_ASUS_MODE7,
-	ALC663_ASUS_MODE8,
-	ALC272_DELL,
-	ALC272_DELL_ZM1,
-	ALC272_SAMSUNG_NC10,
-	ALC662_MODEL_LAST,
-};
-
-#define ALC662_DIGOUT_NID	0x06
-#define ALC662_DIGIN_NID	0x0a
-
-static const hda_nid_t alc662_dac_nids[3] = {
-	/* front, rear, clfe */
-	0x02, 0x03, 0x04
-};
-
-static const hda_nid_t alc272_dac_nids[2] = {
-	0x02, 0x03
-};
-
-static const hda_nid_t alc662_adc_nids[2] = {
-	/* ADC1-2 */
-	0x09, 0x08
-};
-
-static const hda_nid_t alc272_adc_nids[1] = {
-	/* ADC1-2 */
-	0x08,
-};
-
-static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
-static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
-
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-static const struct hda_input_mux alc662_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x1 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-static const struct hda_input_mux alc662_lenovo_101e_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x1 },
-		{ "Line", 0x2 },
-	},
-};
-
-static const struct hda_input_mux alc663_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x1 },
-		{ "Line", 0x2 },
-	},
-};
-
-#if 0 /* set to 1 for testing other input sources below */
-static const struct hda_input_mux alc272_nc10_capture_source = {
-	.num_items = 16,
-	.items = {
-		{ "Autoselect Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-		{ "In-0x02", 0x2 },
-		{ "In-0x03", 0x3 },
-		{ "In-0x04", 0x4 },
-		{ "In-0x05", 0x5 },
-		{ "In-0x06", 0x6 },
-		{ "In-0x07", 0x7 },
-		{ "In-0x08", 0x8 },
-		{ "In-0x09", 0x9 },
-		{ "In-0x0a", 0x0a },
-		{ "In-0x0b", 0x0b },
-		{ "In-0x0c", 0x0c },
-		{ "In-0x0d", 0x0d },
-		{ "In-0x0e", 0x0e },
-		{ "In-0x0f", 0x0f },
-	},
-};
-#endif
-
-/*
- * 2ch mode
- */
-static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
-	{ 2, NULL }
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc662_3ST_ch2_init[] = {
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc662_3ST_ch6_init[] = {
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
-	{ 2, alc662_3ST_ch2_init },
-	{ 6, alc662_3ST_ch6_init },
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc662_sixstack_ch6_init[] = {
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc662_sixstack_ch8_init[] = {
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode alc662_5stack_modes[2] = {
-	{ 2, alc662_sixstack_ch6_init },
-	{ 6, alc662_sixstack_ch8_init },
-};
-
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-
-static const struct snd_kcontrol_new alc662_base_mixer[] = {
-	/* output mixer control */
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
-	/*Input mixer control */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	ALC262_HIPPO_MASTER_SWITCH,
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
-	ALC262_HIPPO_MASTER_SWITCH,
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls alc663_asus_one_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc663_m51va_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_tree_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
-	{ } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_four_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_1bjd_mixer[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls alc663_asus_two_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume",
-				&alc663_asus_two_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
-	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_g71v_mixer[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_g50v_mixer[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc663_mode7_mixer[] = {
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
-	HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
-	HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
-	HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_mode8_mixer[] = {
-	HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
-	HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
-	HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
-	HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-
-static const struct snd_kcontrol_new alc662_chmode_mixer[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = alc_ch_mode_info,
-		.get = alc_ch_mode_get,
-		.put = alc_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb alc662_init_verbs[] = {
-	/* ADC: mute amp left and right */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* Front Pin: output 0 (0x0c) */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* Rear Pin: output 1 (0x0d) */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* CLFE Pin: output 2 (0x0e) */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* Mic (rear) pin: input vref at 80% */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Front Mic pin: input vref at 80% */
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line In pin: input */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line-2 In: Headphone output (output 0 - 0x0c) */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* CD pin widget for input */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	/* FIXME: use matrix-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
-	/* Input mixer */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	{ }
-};
-
-static const struct hda_verb alc662_eapd_init_verbs[] = {
-	/* always trun on EAPD */
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
-static const struct hda_verb alc662_sue_init_verbs[] = {
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc662_eeepc_sue_init_verbs[] = {
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-/* Set Unsolicited Event*/
-static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_m51va_init_verbs[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_21jd_amic_init_verbs[] = {
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc662_1bjd_amic_init_verbs[] = {
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_15jd_amic_init_verbs[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x0},	/* Headphone */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_g71v_init_verbs[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
-	/* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
-
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Headphone */
-
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_g50v_init_verbs[] = {
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Headphone */
-
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc662_ecs_init_verbs[] = {
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc272_dell_zm1_init_verbs[] = {
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc272_dell_init_verbs[] = {
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_mode7_init_verbs[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
-	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct hda_verb alc663_mode8_init_verbs[] = {
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{}
-};
-
-static const struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static void alc662_lenovo_101e_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x1b;
-	spec->autocfg.line_out_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->detect_line = 1;
-	spec->automute_lines = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc662_eeepc_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	alc262_hippo1_setup(codec);
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[0] = 0x1b;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc663_m51va_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x12;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode1 ******************************/
-static void alc663_mode1_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode2 ******************************/
-static void alc662_mode2_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x1b;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode3 ******************************/
-static void alc663_mode3_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode4 ******************************/
-static void alc663_mode4_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[1] = 0x16;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute_mixer_nid[1] = 0x0e;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode5 ******************************/
-static void alc663_mode5_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[1] = 0x16;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute_mixer_nid[1] = 0x0e;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode6 ******************************/
-static void alc663_mode6_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x1b;
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0c;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode7 ******************************/
-static void alc663_mode7_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x1b;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[0] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x19;
-	spec->auto_mic = 1;
-}
-
-/* ***************** Mode8 ******************************/
-static void alc663_mode8_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.hp_pins[1] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[0] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_PIN;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x12;
-	spec->auto_mic = 1;
-}
-
-static void alc663_g71v_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x21;
-	spec->autocfg.line_out_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-	spec->detect_line = 1;
-	spec->automute_lines = 1;
-	spec->ext_mic_pin = 0x18;
-	spec->int_mic_pin = 0x12;
-	spec->auto_mic = 1;
-}
-
-#define alc663_g50v_setup	alc663_m51va_setup
-
-static const struct snd_kcontrol_new alc662_ecs_mixer[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	ALC262_HIPPO_MASTER_SWITCH,
-
-	HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc272_nc10_mixer[] = {
-	/* Master Playback automatically created from Speaker and Headphone */
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	{ } /* end */
-};
-
-
-/*
- * configuration and preset
- */
-static const char * const alc662_models[ALC662_MODEL_LAST] = {
-	[ALC662_3ST_2ch_DIG]	= "3stack-dig",
-	[ALC662_3ST_6ch_DIG]	= "3stack-6ch-dig",
-	[ALC662_3ST_6ch]	= "3stack-6ch",
-	[ALC662_5ST_DIG]	= "5stack-dig",
-	[ALC662_LENOVO_101E]	= "lenovo-101e",
-	[ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
-	[ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
-	[ALC662_ECS] = "ecs",
-	[ALC663_ASUS_M51VA] = "m51va",
-	[ALC663_ASUS_G71V] = "g71v",
-	[ALC663_ASUS_H13] = "h13",
-	[ALC663_ASUS_G50V] = "g50v",
-	[ALC663_ASUS_MODE1] = "asus-mode1",
-	[ALC662_ASUS_MODE2] = "asus-mode2",
-	[ALC663_ASUS_MODE3] = "asus-mode3",
-	[ALC663_ASUS_MODE4] = "asus-mode4",
-	[ALC663_ASUS_MODE5] = "asus-mode5",
-	[ALC663_ASUS_MODE6] = "asus-mode6",
-	[ALC663_ASUS_MODE7] = "asus-mode7",
-	[ALC663_ASUS_MODE8] = "asus-mode8",
-	[ALC272_DELL]		= "dell",
-	[ALC272_DELL_ZM1]	= "dell-zm1",
-	[ALC272_SAMSUNG_NC10]	= "samsung-nc10",
-	[ALC662_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk alc662_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
-	SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL),
-	SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1),
-	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
-	SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
-	SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
-	SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
-	SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
-	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
-	/*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
-	SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
-	/*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
-	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA),
-	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
-	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
-	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
-	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
-	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
-		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
-	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
-		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
-	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
-	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
-	SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
-					ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
-			   ALC663_ASUS_H13),
-	SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
-	{}
-};
-
-static const struct alc_config_preset alc662_presets[] = {
-	[ALC662_3ST_2ch_DIG] = {
-		.mixers = { alc662_3ST_2ch_mixer },
-		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.dig_in_nid = ALC662_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc662_capture_source,
-	},
-	[ALC662_3ST_6ch_DIG] = {
-		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.dig_in_nid = ALC662_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
-		.channel_mode = alc662_3ST_6ch_modes,
-		.need_dac_fix = 1,
-		.input_mux = &alc662_capture_source,
-	},
-	[ALC662_3ST_6ch] = {
-		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
-		.channel_mode = alc662_3ST_6ch_modes,
-		.need_dac_fix = 1,
-		.input_mux = &alc662_capture_source,
-	},
-	[ALC662_5ST_DIG] = {
-		.mixers = { alc662_base_mixer, alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.dig_in_nid = ALC662_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
-		.channel_mode = alc662_5stack_modes,
-		.input_mux = &alc662_capture_source,
-	},
-	[ALC662_LENOVO_101E] = {
-		.mixers = { alc662_lenovo_101e_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc662_sue_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc662_lenovo_101e_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc662_lenovo_101e_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC662_ASUS_EEEPC_P701] = {
-		.mixers = { alc662_eeepc_p701_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc662_eeepc_sue_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc662_eeepc_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC662_ASUS_EEEPC_EP20] = {
-		.mixers = { alc662_eeepc_ep20_mixer,
-			    alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc662_eeepc_ep20_sue_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
-		.channel_mode = alc662_3ST_6ch_modes,
-		.input_mux = &alc662_lenovo_101e_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc662_eeepc_ep20_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC662_ECS] = {
-		.mixers = { alc662_ecs_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc662_ecs_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc662_eeepc_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_M51VA] = {
-		.mixers = { alc663_m51va_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_m51va_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_m51va_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_G71V] = {
-		.mixers = { alc663_g71v_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_g71v_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_g71v_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_H13] = {
-		.mixers = { alc663_m51va_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_m51va_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.setup = alc663_m51va_setup,
-		.unsol_event = alc_sku_unsol_event,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_G50V] = {
-		.mixers = { alc663_g50v_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_g50v_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
-		.channel_mode = alc662_3ST_6ch_modes,
-		.input_mux = &alc663_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_g50v_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE1] = {
-		.mixers = { alc663_m51va_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_21jd_amic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode1_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC662_ASUS_MODE2] = {
-		.mixers = { alc662_1bjd_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc662_1bjd_amic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc662_mode2_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE3] = {
-		.mixers = { alc663_two_hp_m1_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_two_hp_amic_m1_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode3_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE4] = {
-		.mixers = { alc663_asus_21jd_clfe_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_21jd_amic_init_verbs},
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode4_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE5] = {
-		.mixers = { alc663_asus_15jd_clfe_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_15jd_amic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode5_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE6] = {
-		.mixers = { alc663_two_hp_m2_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_two_hp_amic_m2_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode6_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE7] = {
-		.mixers = { alc663_mode7_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_mode7_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode7_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC663_ASUS_MODE8] = {
-		.mixers = { alc663_mode8_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_mode8_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
-		.hp_nid = 0x03,
-		.dac_nids = alc662_dac_nids,
-		.dig_out_nid = ALC662_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode8_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC272_DELL] = {
-		.mixers = { alc663_m51va_mixer },
-		.cap_mixer = alc272_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc272_dell_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc272_dac_nids),
-		.dac_nids = alc272_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.adc_nids = alc272_adc_nids,
-		.num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
-		.capsrc_nids = alc272_capsrc_nids,
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_m51va_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC272_DELL_ZM1] = {
-		.mixers = { alc663_m51va_mixer },
-		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc272_dell_zm1_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc272_dac_nids),
-		.dac_nids = alc272_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.adc_nids = alc662_adc_nids,
-		.num_adc_nids = 1,
-		.capsrc_nids = alc662_capsrc_nids,
-		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_m51va_setup,
-		.init_hook = alc_inithook,
-	},
-	[ALC272_SAMSUNG_NC10] = {
-		.mixers = { alc272_nc10_mixer },
-		.init_verbs = { alc662_init_verbs,
-				alc662_eapd_init_verbs,
-				alc663_21jd_amic_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc272_dac_nids),
-		.dac_nids = alc272_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
-		.channel_mode = alc662_3ST_2ch_modes,
-		/*.input_mux = &alc272_nc10_capture_source,*/
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc663_mode4_setup,
-		.init_hook = alc_inithook,
-	},
-};
-
-
diff --git a/sound/pci/hda/alc680_quirks.c b/sound/pci/hda/alc680_quirks.c
deleted file mode 100644
index 0eeb227..0000000
--- a/sound/pci/hda/alc680_quirks.c
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- * ALC680 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC680 models */
-enum {
-	ALC680_AUTO,
-	ALC680_BASE,
-	ALC680_MODEL_LAST,
-};
-
-#define ALC680_DIGIN_NID	ALC880_DIGIN_NID
-#define ALC680_DIGOUT_NID	ALC880_DIGOUT_NID
-#define alc680_modes		alc260_modes
-
-static const hda_nid_t alc680_dac_nids[3] = {
-	/* Lout1, Lout2, hp */
-	0x02, 0x03, 0x04
-};
-
-static const hda_nid_t alc680_adc_nids[3] = {
-	/* ADC0-2 */
-	/* DMIC, MIC, Line-in*/
-	0x07, 0x08, 0x09
-};
-
-/*
- * Analog capture ADC cgange
- */
-static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec)
-{
-	static hda_nid_t pins[] = {0x18, 0x19};
-	static hda_nid_t adcs[] = {0x08, 0x09};
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(pins); i++) {
-		if (!is_jack_detectable(codec, pins[i]))
-			continue;
-		if (snd_hda_jack_detect(codec, pins[i]))
-			return adcs[i];
-	}
-	return 0x07;
-}
-
-static void alc680_rec_autoswitch(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t nid = alc680_get_cur_adc(codec);
-	if (spec->cur_adc && nid != spec->cur_adc) {
-		__snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
-		spec->cur_adc = nid;
-		snd_hda_codec_setup_stream(codec, nid,
-					   spec->cur_adc_stream_tag, 0,
-					   spec->cur_adc_format);
-	}
-}
-
-static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      unsigned int stream_tag,
-				      unsigned int format,
-				      struct snd_pcm_substream *substream)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t nid = alc680_get_cur_adc(codec);
-
-	spec->cur_adc = nid;
-	spec->cur_adc_stream_tag = stream_tag;
-	spec->cur_adc_format = format;
-	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
-	return 0;
-}
-
-static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      struct snd_pcm_substream *substream)
-{
-	struct alc_spec *spec = codec->spec;
-	snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
-	spec->cur_adc = 0;
-	return 0;
-}
-
-static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
-	.substreams = 1, /* can be overridden */
-	.channels_min = 2,
-	.channels_max = 2,
-	/* NID is set in alc_build_pcms */
-	.ops = {
-		.prepare = alc680_capture_pcm_prepare,
-		.cleanup = alc680_capture_pcm_cleanup
-	},
-};
-
-static const struct snd_kcontrol_new alc680_base_mixer[] = {
-	/* output mixer control */
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
-	{ }
-};
-
-static const struct hda_bind_ctls alc680_bind_cap_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
-		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
-		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls alc680_bind_cap_switch = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
-		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
-		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
-		0
-	},
-};
-
-static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
-	HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
-	HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
-	{ } /* end */
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc680_init_verbs[] = {
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
-	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT   | AC_USRSP_EN},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT  | AC_USRSP_EN},
-	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT  | AC_USRSP_EN},
-
-	{ }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc680_base_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x16;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[1] = 0x15;
-	spec->autocfg.num_inputs = 2;
-	spec->autocfg.inputs[0].pin = 0x18;
-	spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
-	spec->autocfg.inputs[1].pin = 0x19;
-	spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc680_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	if ((res >> 26) == ALC_HP_EVENT)
-		alc_hp_automute(codec);
-	if ((res >> 26) == ALC_MIC_EVENT)
-		alc680_rec_autoswitch(codec);
-}
-
-static void alc680_inithook(struct hda_codec *codec)
-{
-	alc_hp_automute(codec);
-	alc680_rec_autoswitch(codec);
-}
-
-/*
- * configuration and preset
- */
-static const char * const alc680_models[ALC680_MODEL_LAST] = {
-	[ALC680_BASE]		= "base",
-	[ALC680_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk alc680_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
-	{}
-};
-
-static const struct alc_config_preset alc680_presets[] = {
-	[ALC680_BASE] = {
-		.mixers = { alc680_base_mixer },
-		.cap_mixer =  alc680_master_capture_mixer,
-		.init_verbs = { alc680_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc680_dac_nids),
-		.dac_nids = alc680_dac_nids,
-		.dig_out_nid = ALC680_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc680_modes),
-		.channel_mode = alc680_modes,
-		.unsol_event = alc680_unsol_event,
-		.setup = alc680_base_setup,
-		.init_hook = alc680_inithook,
-
-	},
-};
diff --git a/sound/pci/hda/alc861_quirks.c b/sound/pci/hda/alc861_quirks.c
deleted file mode 100644
index d719ec6..0000000
--- a/sound/pci/hda/alc861_quirks.c
+++ /dev/null
@@ -1,725 +0,0 @@
-/*
- * ALC660/ALC861 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC861 models */
-enum {
-	ALC861_AUTO,
-	ALC861_3ST,
-	ALC660_3ST,
-	ALC861_3ST_DIG,
-	ALC861_6ST_DIG,
-	ALC861_UNIWILL_M31,
-	ALC861_TOSHIBA,
-	ALC861_ASUS,
-	ALC861_ASUS_LAPTOP,
-	ALC861_MODEL_LAST,
-};
-
-/*
- *  ALC861 channel source setting (2/6 channel selection for 3-stack)
- */
-
-/*
- * set the path ways for 2 channel output
- * need to set the codec line out and mic 1 pin widgets to inputs
- */
-static const struct hda_verb alc861_threestack_ch2_init[] = {
-	/* set pin widget 1Ah (line in) for input */
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* set pin widget 18h (mic1/2) for input, for mic also enable
-	 * the vref
-	 */
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
-#if 0
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
-#endif
-	{ } /* end */
-};
-/*
- * 6ch mode
- * need to set the codec line out and mic 1 pin widgets to outputs
- */
-static const struct hda_verb alc861_threestack_ch6_init[] = {
-	/* set pin widget 1Ah (line in) for output (Back Surround)*/
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* set pin widget 18h (mic1) for output (CLFE)*/
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-
-	{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
-
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-#if 0
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
-#endif
-	{ } /* end */
-};
-
-static const struct hda_channel_mode alc861_threestack_modes[2] = {
-	{ 2, alc861_threestack_ch2_init },
-	{ 6, alc861_threestack_ch6_init },
-};
-/* Set mic1 as input and unmute the mixer */
-static const struct hda_verb alc861_uniwill_m31_ch2_init[] = {
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
-	{ } /* end */
-};
-/* Set mic1 as output and mute mixer */
-static const struct hda_verb alc861_uniwill_m31_ch4_init[] = {
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
-	{ } /* end */
-};
-
-static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
-	{ 2, alc861_uniwill_m31_ch2_init },
-	{ 4, alc861_uniwill_m31_ch4_init },
-};
-
-/* Set mic1 and line-in as input and unmute the mixer */
-static const struct hda_verb alc861_asus_ch2_init[] = {
-	/* set pin widget 1Ah (line in) for input */
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* set pin widget 18h (mic1/2) for input, for mic also enable
-	 * the vref
-	 */
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
-#if 0
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
-#endif
-	{ } /* end */
-};
-/* Set mic1 nad line-in as output and mute mixer */
-static const struct hda_verb alc861_asus_ch6_init[] = {
-	/* set pin widget 1Ah (line in) for output (Back Surround)*/
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
-	/* set pin widget 18h (mic1) for output (CLFE)*/
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
-	{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
-
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-#if 0
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
-#endif
-	{ } /* end */
-};
-
-static const struct hda_channel_mode alc861_asus_modes[2] = {
-	{ 2, alc861_asus_ch2_init },
-	{ 6, alc861_asus_ch6_init },
-};
-
-/* patch-ALC861 */
-
-static const struct snd_kcontrol_new alc861_base_mixer[] = {
-        /* output mixer control */
-	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
-
-        /*Input mixer control */
-	/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
-
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_3ST_mixer[] = {
-        /* output mixer control */
-	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
-	/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
-
-	/* Input mixer control */
-	/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = alc_ch_mode_info,
-		.get = alc_ch_mode_get,
-		.put = alc_ch_mode_put,
-                .private_value = ARRAY_SIZE(alc861_threestack_modes),
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_toshiba_mixer[] = {
-        /* output mixer control */
-	HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
-
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
-        /* output mixer control */
-	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
-	/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
-
-	/* Input mixer control */
-	/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = alc_ch_mode_info,
-		.get = alc_ch_mode_get,
-		.put = alc_ch_mode_put,
-                .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_asus_mixer[] = {
-        /* output mixer control */
-	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
-
-	/* Input mixer control */
-	HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = alc_ch_mode_info,
-		.get = alc_ch_mode_get,
-		.put = alc_ch_mode_put,
-                .private_value = ARRAY_SIZE(alc861_asus_modes),
-	},
-	{ }
-};
-
-/* additional mixer */
-static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	{ }
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc861_base_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	/* port-A for surround (rear panel) */
-	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-B for mic-in (rear panel) with vref */
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-C for line-in (rear panel) */
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* port-D for Front */
-	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-E for HP out (front panel) */
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* route front PCM to HP */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-F for mic-in (front panel) with vref */
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-G for CLFE (rear panel) */
-	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-H for side (rear panel) */
-	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* CD-in */
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* route front mic to ADC1*/
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/* Unmute DAC0~3 & spdif out*/
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* Unmute Mixer 14 (mic) 1c (Line in)*/
-	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* Unmute Stereo Mixer 15 */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* hp used DAC 3 (Front) */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-
-	{ }
-};
-
-static const struct hda_verb alc861_threestack_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	/* port-A for surround (rear panel) */
-	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	/* port-B for mic-in (rear panel) with vref */
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-C for line-in (rear panel) */
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* port-D for Front */
-	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-E for HP out (front panel) */
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* route front PCM to HP */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-F for mic-in (front panel) with vref */
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-G for CLFE (rear panel) */
-	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	/* port-H for side (rear panel) */
-	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	/* CD-in */
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* route front mic to ADC1*/
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Unmute DAC0~3 & spdif out*/
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* Unmute Mixer 14 (mic) 1c (Line in)*/
-	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* Unmute Stereo Mixer 15 */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* hp used DAC 3 (Front) */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{ }
-};
-
-static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	/* port-A for surround (rear panel) */
-	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	/* port-B for mic-in (rear panel) with vref */
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-C for line-in (rear panel) */
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* port-D for Front */
-	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-E for HP out (front panel) */
-	/* this has to be set to VREF80 */
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* route front PCM to HP */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-F for mic-in (front panel) with vref */
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-G for CLFE (rear panel) */
-	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	/* port-H for side (rear panel) */
-	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	/* CD-in */
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* route front mic to ADC1*/
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Unmute DAC0~3 & spdif out*/
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	/* Unmute Mixer 14 (mic) 1c (Line in)*/
-	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* Unmute Stereo Mixer 15 */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* hp used DAC 3 (Front) */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{ }
-};
-
-static const struct hda_verb alc861_asus_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	/* port-A for surround (rear panel)
-	 * according to codec#0 this is the HP jack
-	 */
-	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
-	/* route front PCM to HP */
-	{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	/* port-B for mic-in (rear panel) with vref */
-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-C for line-in (rear panel) */
-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* port-D for Front */
-	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-E for HP out (front panel) */
-	/* this has to be set to VREF80 */
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* route front PCM to HP */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
-	/* port-F for mic-in (front panel) with vref */
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* port-G for CLFE (rear panel) */
-	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* port-H for side (rear panel) */
-	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* CD-in */
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* route front mic to ADC1*/
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Unmute DAC0~3 & spdif out*/
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Unmute Mixer 14 (mic) 1c (Line in)*/
-	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* Unmute Stereo Mixer 15 */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* hp used DAC 3 (Front) */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{ }
-};
-
-/* additional init verbs for ASUS laptops */
-static const struct hda_verb alc861_asus_laptop_init_verbs[] = {
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
-	{ }
-};
-
-static const struct hda_verb alc861_toshiba_init_verbs[] = {
-	{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-
-	{ }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc861_toshiba_automute(struct hda_codec *codec)
-{
-	unsigned int present = snd_hda_jack_detect(codec, 0x0f);
-
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
-				 HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
-}
-
-static void alc861_toshiba_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) == ALC_HP_EVENT)
-		alc861_toshiba_automute(codec);
-}
-
-#define ALC861_DIGOUT_NID	0x07
-
-static const struct hda_channel_mode alc861_8ch_modes[1] = {
-	{ 8, NULL }
-};
-
-static const hda_nid_t alc861_dac_nids[4] = {
-	/* front, surround, clfe, side */
-	0x03, 0x06, 0x05, 0x04
-};
-
-static const hda_nid_t alc660_dac_nids[3] = {
-	/* front, clfe, surround */
-	0x03, 0x05, 0x06
-};
-
-static const hda_nid_t alc861_adc_nids[1] = {
-	/* ADC0-2 */
-	0x08,
-};
-
-static const struct hda_input_mux alc861_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x3 },
-		{ "Line", 0x1 },
-		{ "CD", 0x4 },
-		{ "Mixer", 0x5 },
-	},
-};
-
-/*
- * configuration and preset
- */
-static const char * const alc861_models[ALC861_MODEL_LAST] = {
-	[ALC861_3ST]		= "3stack",
-	[ALC660_3ST]		= "3stack-660",
-	[ALC861_3ST_DIG]	= "3stack-dig",
-	[ALC861_6ST_DIG]	= "6stack-dig",
-	[ALC861_UNIWILL_M31]	= "uniwill-m31",
-	[ALC861_TOSHIBA]	= "toshiba",
-	[ALC861_ASUS]		= "asus",
-	[ALC861_ASUS_LAPTOP]	= "asus-laptop",
-	[ALC861_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk alc861_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
-	SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
-	SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
-	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
-	/* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
-	 *        Any other models that need this preset?
-	 */
-	/* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
-	SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
-	SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
-	SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
-	SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
-	SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
-	/* FIXME: the below seems conflict */
-	/* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
-	SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
-	SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
-	{}
-};
-
-static const struct alc_config_preset alc861_presets[] = {
-	[ALC861_3ST] = {
-		.mixers = { alc861_3ST_mixer },
-		.init_verbs = { alc861_threestack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
-		.channel_mode = alc861_threestack_modes,
-		.need_dac_fix = 1,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-	[ALC861_3ST_DIG] = {
-		.mixers = { alc861_base_mixer },
-		.init_verbs = { alc861_threestack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.dig_out_nid = ALC861_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
-		.channel_mode = alc861_threestack_modes,
-		.need_dac_fix = 1,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-	[ALC861_6ST_DIG] = {
-		.mixers = { alc861_base_mixer },
-		.init_verbs = { alc861_base_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.dig_out_nid = ALC861_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
-		.channel_mode = alc861_8ch_modes,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-	[ALC660_3ST] = {
-		.mixers = { alc861_3ST_mixer },
-		.init_verbs = { alc861_threestack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc660_dac_nids),
-		.dac_nids = alc660_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
-		.channel_mode = alc861_threestack_modes,
-		.need_dac_fix = 1,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-	[ALC861_UNIWILL_M31] = {
-		.mixers = { alc861_uniwill_m31_mixer },
-		.init_verbs = { alc861_uniwill_m31_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.dig_out_nid = ALC861_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
-		.channel_mode = alc861_uniwill_m31_modes,
-		.need_dac_fix = 1,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-	[ALC861_TOSHIBA] = {
-		.mixers = { alc861_toshiba_mixer },
-		.init_verbs = { alc861_base_init_verbs,
-				alc861_toshiba_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
-		.channel_mode = alc883_3ST_2ch_modes,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-		.unsol_event = alc861_toshiba_unsol_event,
-		.init_hook = alc861_toshiba_automute,
-	},
-	[ALC861_ASUS] = {
-		.mixers = { alc861_asus_mixer },
-		.init_verbs = { alc861_asus_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.dig_out_nid = ALC861_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
-		.channel_mode = alc861_asus_modes,
-		.need_dac_fix = 1,
-		.hp_nid = 0x06,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-	[ALC861_ASUS_LAPTOP] = {
-		.mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
-		.init_verbs = { alc861_asus_init_verbs,
-				alc861_asus_laptop_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
-		.dac_nids = alc861_dac_nids,
-		.dig_out_nid = ALC861_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
-		.channel_mode = alc883_3ST_2ch_modes,
-		.need_dac_fix = 1,
-		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
-		.adc_nids = alc861_adc_nids,
-		.input_mux = &alc861_capture_source,
-	},
-};
-
diff --git a/sound/pci/hda/alc861vd_quirks.c b/sound/pci/hda/alc861vd_quirks.c
deleted file mode 100644
index 8f28450..0000000
--- a/sound/pci/hda/alc861vd_quirks.c
+++ /dev/null
@@ -1,605 +0,0 @@
-/*
- * ALC660-VD/ALC861-VD quirk models
- * included by patch_realtek.c
- */
-
-/* ALC861-VD models */
-enum {
-	ALC861VD_AUTO,
-	ALC660VD_3ST,
-	ALC660VD_3ST_DIG,
-	ALC660VD_ASUS_V1S,
-	ALC861VD_3ST,
-	ALC861VD_3ST_DIG,
-	ALC861VD_6ST_DIG,
-	ALC861VD_LENOVO,
-	ALC861VD_DALLAS,
-	ALC861VD_HP,
-	ALC861VD_MODEL_LAST,
-};
-
-#define ALC861VD_DIGOUT_NID	0x06
-
-static const hda_nid_t alc861vd_dac_nids[4] = {
-	/* front, surr, clfe, side surr */
-	0x02, 0x03, 0x04, 0x05
-};
-
-/* dac_nids for ALC660vd are in a different order - according to
- * Realtek's driver.
- * This should probably result in a different mixer for 6stack models
- * of ALC660vd codecs, but for now there is only 3stack mixer
- * - and it is the same as in 861vd.
- * adc_nids in ALC660vd are (is) the same as in 861vd
- */
-static const hda_nid_t alc660vd_dac_nids[3] = {
-	/* front, rear, clfe, rear_surr */
-	0x02, 0x04, 0x03
-};
-
-static const hda_nid_t alc861vd_adc_nids[1] = {
-	/* ADC0 */
-	0x09,
-};
-
-static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-static const struct hda_input_mux alc861vd_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x1 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-static const struct hda_input_mux alc861vd_dallas_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-	},
-};
-
-static const struct hda_input_mux alc861vd_hp_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "ATAPI Mic", 0x1 },
-	},
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
-	{ 2, NULL }
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc861vd_6stack_ch6_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc861vd_6stack_ch8_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
-	{ 6, alc861vd_6stack_ch6_init },
-	{ 8, alc861vd_6stack_ch8_init },
-};
-
-static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = alc_ch_mode_info,
-		.get = alc_ch_mode_get,
-		.put = alc_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-static const struct snd_kcontrol_new alc861vd_6st_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
-				HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
-				HDA_OUTPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
-
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc861vd_3st_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
-
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	/*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
-	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-
-	{ } /* end */
-};
-
-/* Pin assignment: Speaker=0x14, HP = 0x15,
- *                 Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
- */
-static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
-
-/* Pin assignment: Speaker=0x14, Line-out = 0x15,
- *                 Front Mic=0x18, ATAPI Mic = 0x19,
- */
-static const struct snd_kcontrol_new alc861vd_hp_mixer[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
-	{ } /* end */
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc861vd_volume_init_verbs[] = {
-	/*
-	 * Unmute ADC0 and set the default input to mic-in
-	 */
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
-	 * the analog-loopback mixer widget
-	 */
-	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
-	/* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
-	/*
-	 * Set up output mixers (0x02 - 0x05)
-	 */
-	/* set vol=0 to output mixers */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	/* set up input amps for analog loopback */
-	/* Amp Indices: DAC = 0, mixer = 1 */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
-	{ }
-};
-
-/*
- * 3-stack pin configuration:
- * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
- */
-static const struct hda_verb alc861vd_3stack_init_verbs[] = {
-	/*
-	 * Set pin mode and muting
-	 */
-	/* set front pin widgets 0x14 for output */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	/* Mic (rear) pin: input vref at 80% */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Front Mic pin: input vref at 80% */
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line In pin: input */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line-2 In: Headphone output (output 0 - 0x0c) */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* CD pin widget for input */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	{ }
-};
-
-/*
- * 6-stack pin configuration:
- */
-static const struct hda_verb alc861vd_6stack_init_verbs[] = {
-	/*
-	 * Set pin mode and muting
-	 */
-	/* set front pin widgets 0x14 for output */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-
-	/* Rear Pin: output 1 (0x0d) */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* CLFE Pin: output 2 (0x0e) */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
-	/* Side Pin: output 3 (0x0f) */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
-
-	/* Mic (rear) pin: input vref at 80% */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Front Mic pin: input vref at 80% */
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line In pin: input */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line-2 In: Headphone output (output 0 - 0x0c) */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* CD pin widget for input */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	{ }
-};
-
-static const struct hda_verb alc861vd_eapd_verbs[] = {
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
-static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-	{}
-};
-
-static void alc861vd_lenovo_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x1b;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
-{
-	alc_hp_automute(codec);
-	alc88x_simple_mic_automute(codec);
-}
-
-static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
-					unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC_MIC_EVENT:
-		alc88x_simple_mic_automute(codec);
-		break;
-	default:
-		alc_sku_unsol_event(codec, res);
-		break;
-	}
-}
-
-static const struct hda_verb alc861vd_dallas_verbs[] = {
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
-
-	{ } /* end */
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc861vd_dallas_setup(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/*
- * configuration and preset
- */
-static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
-	[ALC660VD_3ST]		= "3stack-660",
-	[ALC660VD_3ST_DIG]	= "3stack-660-digout",
-	[ALC660VD_ASUS_V1S]	= "asus-v1s",
-	[ALC861VD_3ST]		= "3stack",
-	[ALC861VD_3ST_DIG]	= "3stack-digout",
-	[ALC861VD_6ST_DIG]	= "6stack-digout",
-	[ALC861VD_LENOVO]	= "lenovo",
-	[ALC861VD_DALLAS]	= "dallas",
-	[ALC861VD_HP]		= "hp",
-	[ALC861VD_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk alc861vd_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
-	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
-	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
-	/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
-	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
-	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
-	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
-	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
-	/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
-	SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
-	SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
-	{}
-};
-
-static const struct alc_config_preset alc861vd_presets[] = {
-	[ALC660VD_3ST] = {
-		.mixers = { alc861vd_3st_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-				 alc861vd_3stack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
-		.dac_nids = alc660vd_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_capture_source,
-	},
-	[ALC660VD_3ST_DIG] = {
-		.mixers = { alc861vd_3st_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-				 alc861vd_3stack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
-		.dac_nids = alc660vd_dac_nids,
-		.dig_out_nid = ALC861VD_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_capture_source,
-	},
-	[ALC861VD_3ST] = {
-		.mixers = { alc861vd_3st_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-				 alc861vd_3stack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
-		.dac_nids = alc861vd_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_capture_source,
-	},
-	[ALC861VD_3ST_DIG] = {
-		.mixers = { alc861vd_3st_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-		 		 alc861vd_3stack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
-		.dac_nids = alc861vd_dac_nids,
-		.dig_out_nid = ALC861VD_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_capture_source,
-	},
-	[ALC861VD_6ST_DIG] = {
-		.mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-				alc861vd_6stack_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
-		.dac_nids = alc861vd_dac_nids,
-		.dig_out_nid = ALC861VD_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
-		.channel_mode = alc861vd_6stack_modes,
-		.input_mux = &alc861vd_capture_source,
-	},
-	[ALC861VD_LENOVO] = {
-		.mixers = { alc861vd_lenovo_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-				alc861vd_3stack_init_verbs,
-				alc861vd_eapd_verbs,
-				alc861vd_lenovo_unsol_verbs },
-		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
-		.dac_nids = alc660vd_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_capture_source,
-		.unsol_event = alc861vd_lenovo_unsol_event,
-		.setup = alc861vd_lenovo_setup,
-		.init_hook = alc861vd_lenovo_init_hook,
-	},
-	[ALC861VD_DALLAS] = {
-		.mixers = { alc861vd_dallas_mixer },
-		.init_verbs = { alc861vd_dallas_verbs },
-		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
-		.dac_nids = alc861vd_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_dallas_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc861vd_dallas_setup,
-		.init_hook = alc_hp_automute,
-	},
-	[ALC861VD_HP] = {
-		.mixers = { alc861vd_hp_mixer },
-		.init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
-		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
-		.dac_nids = alc861vd_dac_nids,
-		.dig_out_nid = ALC861VD_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_hp_capture_source,
-		.unsol_event = alc_sku_unsol_event,
-		.setup = alc861vd_dallas_setup,
-		.init_hook = alc_hp_automute,
-	},
-	[ALC660VD_ASUS_V1S] = {
-		.mixers = { alc861vd_lenovo_mixer },
-		.init_verbs = { alc861vd_volume_init_verbs,
-				alc861vd_3stack_init_verbs,
-				alc861vd_eapd_verbs,
-				alc861vd_lenovo_unsol_verbs },
-		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
-		.dac_nids = alc660vd_dac_nids,
-		.dig_out_nid = ALC861VD_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
-		.channel_mode = alc861vd_3stack_2ch_modes,
-		.input_mux = &alc861vd_capture_source,
-		.unsol_event = alc861vd_lenovo_unsol_event,
-		.setup = alc861vd_lenovo_setup,
-		.init_hook = alc861vd_lenovo_init_hook,
-	},
-};
-
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
index c844d2b..bea22ed 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -749,8 +749,7 @@
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x16;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc880_uniwill_init_hook(struct hda_codec *codec)
@@ -781,8 +780,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -1051,8 +1049,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 /*
@@ -1137,8 +1134,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
@@ -1188,7 +1184,7 @@
 	struct alc_spec *spec = codec->spec;
 	alc_hp_automute(codec);
 	/* toggle EAPD */
-	if (spec->jack_present)
+	if (spec->hp_jack_present)
 		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
 	else
 		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
@@ -1210,8 +1206,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
index 617d047..e251514 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -173,8 +173,7 @@
 	spec->autocfg.speaker_pins[2] = 0x17;
 	spec->autocfg.speaker_pins[3] = 0x19;
 	spec->autocfg.speaker_pins[4] = 0x1a;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc889_intel_init_hook(struct hda_codec *codec)
@@ -191,8 +190,7 @@
 	spec->autocfg.hp_pins[1] = 0x1b; /* hp */
 	spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
 	spec->autocfg.speaker_pins[1] = 0x15; /* bass */
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 /*
@@ -475,8 +473,7 @@
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
@@ -487,8 +484,7 @@
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
@@ -499,8 +495,7 @@
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
@@ -511,8 +506,7 @@
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x1b;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 #define ALC882_DIGOUT_NID	0x06
@@ -1711,8 +1705,7 @@
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x18;
 	spec->autocfg.speaker_pins[1] = 0x1a;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 #define alc885_mb5_setup	alc885_imac24_setup
@@ -1721,12 +1714,11 @@
 /* Macbook Air 2,1 */
 static void alc885_mba21_setup(struct hda_codec *codec)
 {
-       struct alc_spec *spec = codec->spec;
+	struct alc_spec *spec = codec->spec;
 
-       spec->autocfg.hp_pins[0] = 0x14;
-       spec->autocfg.speaker_pins[0] = 0x18;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	spec->autocfg.hp_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x18;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 
@@ -1737,8 +1729,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc885_imac91_setup(struct hda_codec *codec)
@@ -1748,8 +1739,7 @@
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x18;
 	spec->autocfg.speaker_pins[1] = 0x1a;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct hda_verb alc882_targa_verbs[] = {
@@ -1773,7 +1763,7 @@
 	struct alc_spec *spec = codec->spec;
 	alc_hp_automute(codec);
 	snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
-				  spec->jack_present ? 1 : 3);
+				  spec->hp_jack_present ? 1 : 3);
 }
 
 static void alc882_targa_setup(struct hda_codec *codec)
@@ -1782,8 +1772,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -2187,8 +2176,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x1a;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
@@ -2341,8 +2329,7 @@
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct hda_verb alc883_mitac_verbs[] = {
@@ -2507,8 +2494,7 @@
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x18;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct hda_verb alc888_3st_hp_verbs[] = {
@@ -2568,8 +2554,7 @@
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.line_out_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -2579,8 +2564,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -2593,8 +2577,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
@@ -2623,8 +2606,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc883_haier_w66_setup(struct hda_codec *codec)
@@ -2633,8 +2615,7 @@
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc883_lenovo_101e_setup(struct hda_codec *codec)
@@ -2644,10 +2625,7 @@
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.line_out_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	spec->automute = 1;
-	spec->detect_line = 1;
-	spec->automute_lines = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -2658,8 +2636,7 @@
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[1] = 0x16;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct hda_verb alc883_acer_eapd_verbs[] = {
@@ -2689,8 +2666,7 @@
 	spec->autocfg.speaker_pins[1] = 0x15;
 	spec->autocfg.speaker_pins[2] = 0x16;
 	spec->autocfg.speaker_pins[3] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc888_lenovo_sky_setup(struct hda_codec *codec)
@@ -2703,8 +2679,7 @@
 	spec->autocfg.speaker_pins[2] = 0x16;
 	spec->autocfg.speaker_pins[3] = 0x17;
 	spec->autocfg.speaker_pins[4] = 0x1a;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static void alc883_vaiott_setup(struct hda_codec *codec)
@@ -2714,8 +2689,7 @@
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x17;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct hda_verb alc888_asus_m90v_verbs[] = {
@@ -2739,8 +2713,7 @@
 	spec->ext_mic_pin = 0x18;
 	spec->int_mic_pin = 0x19;
 	spec->auto_mic = 1;
-	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
 }
 
 static const struct hda_verb alc888_asus_eee1601_verbs[] = {
diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c
index 2be1129..a18952e 100644
--- a/sound/pci/hda/alc_quirks.c
+++ b/sound/pci/hda/alc_quirks.c
@@ -453,6 +453,19 @@
 	alc_fixup_autocfg_pin_nums(codec);
 }
 
+static void alc_simple_setup_automute(struct alc_spec *spec, int mode)
+{
+	int lo_pin = spec->autocfg.line_out_pins[0];
+
+	if (lo_pin == spec->autocfg.speaker_pins[0] ||
+		lo_pin == spec->autocfg.hp_pins[0])
+		lo_pin = 0;
+	spec->automute_mode = mode;
+	spec->detect_hp = !!spec->autocfg.hp_pins[0];
+	spec->detect_lo = !!lo_pin;
+	spec->automute_lo = spec->automute_lo_possible = !!lo_pin;
+	spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0];
+}
 
 /* auto-toggle front mic */
 static void alc88x_simple_mic_automute(struct hda_codec *codec)
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c..1715e8b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -34,6 +34,9 @@
 #include "hda_beep.h"
 #include <sound/hda_hwdep.h>
 
+#define CREATE_TRACE_POINTS
+#include "hda_trace.h"
+
 /*
  * vendor / preset table
  */
@@ -208,15 +211,19 @@
  again:
 	snd_hda_power_up(codec);
 	mutex_lock(&bus->cmd_mutex);
+	trace_hda_send_cmd(codec, cmd);
 	err = bus->ops.command(bus, cmd);
-	if (!err && res)
+	if (!err && res) {
 		*res = bus->ops.get_response(bus, codec->addr);
+		trace_hda_get_response(codec, *res);
+	}
 	mutex_unlock(&bus->cmd_mutex);
 	snd_hda_power_down(codec);
 	if (res && *res == -1 && bus->rirb_error) {
 		if (bus->response_reset) {
 			snd_printd("hda_codec: resetting BUS due to "
 				   "fatal communication error\n");
+			trace_hda_bus_reset(bus);
 			bus->ops.bus_reset(bus);
 		}
 		goto again;
@@ -579,9 +586,13 @@
 		return -1;
 	}
 	recursive++;
-	for (i = 0; i < nums; i++)
+	for (i = 0; i < nums; i++) {
+		unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+		if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+			continue;
 		if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
 			return i;
+	}
 	return -1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
@@ -603,6 +614,7 @@
 	struct hda_bus_unsolicited *unsol;
 	unsigned int wp;
 
+	trace_hda_unsol_event(bus, res, res_ex);
 	unsol = bus->unsol;
 	if (!unsol)
 		return 0;
@@ -1479,8 +1491,11 @@
 				  struct hda_cvt_setup *q)
 {
 	hda_nid_t nid = q->nid;
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
+	if (q->stream_tag || q->channel_id)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+	if (q->format_id)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0
+);
 	memset(q, 0, sizeof(*q));
 	q->nid = nid;
 }
@@ -1685,6 +1700,29 @@
 EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
 
 /**
+ * snd_hda_override_pin_caps - Override the pin capabilities
+ * @codec: the CODEC
+ * @nid: the NID to override
+ * @caps: the capability bits to set
+ *
+ * Override the cached PIN capabilitiy bits value by the given one.
+ *
+ * Returns zero if successful or a negative error code.
+ */
+int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
+			      unsigned int caps)
+{
+	struct hda_amp_info *info;
+	info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
+	if (!info)
+		return -ENOMEM;
+	info->amp_caps = caps;
+	info->head.val |= INFO_AMP_CAPS;
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps);
+
+/**
  * snd_hda_pin_sense - execute pin sense measurement
  * @codec: the CODEC to sense
  * @nid: the pin NID to sense
@@ -4083,6 +4121,7 @@
 		return;
 	}
 
+	trace_hda_power_down(codec);
 	hda_call_codec_suspend(codec);
 	if (bus->ops.pm_notify)
 		bus->ops.pm_notify(bus);
@@ -4121,6 +4160,7 @@
 	if (codec->power_on || codec->power_transition)
 		return;
 
+	trace_hda_power_up(codec);
 	snd_hda_update_power_acct(codec);
 	codec->power_on = 1;
 	codec->power_jiffies = jiffies;
@@ -4533,6 +4573,11 @@
 		snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
 					   0, format);
 	/* extra outputs copied from front */
+	for (i = 0; i < ARRAY_SIZE(mout->hp_out_nid); i++)
+		if (!mout->no_share_stream && mout->hp_out_nid[i])
+			snd_hda_codec_setup_stream(codec,
+						   mout->hp_out_nid[i],
+						   stream_tag, 0, format);
 	for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
 		if (!mout->no_share_stream && mout->extra_out_nid[i])
 			snd_hda_codec_setup_stream(codec,
@@ -4565,6 +4610,10 @@
 		snd_hda_codec_cleanup_stream(codec, nids[i]);
 	if (mout->hp_nid)
 		snd_hda_codec_cleanup_stream(codec, mout->hp_nid);
+	for (i = 0; i < ARRAY_SIZE(mout->hp_out_nid); i++)
+		if (mout->hp_out_nid[i])
+			snd_hda_codec_cleanup_stream(codec,
+						     mout->hp_out_nid[i]);
 	for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
 		if (mout->extra_out_nid[i])
 			snd_hda_codec_cleanup_stream(codec,
@@ -4645,6 +4694,27 @@
 	}
 }
 
+/* Reorder the surround channels
+ * ALSA sequence is front/surr/clfe/side
+ * HDA sequence is:
+ *    4-ch: front/surr  =>  OK as it is
+ *    6-ch: front/clfe/surr
+ *    8-ch: front/clfe/rear/side|fc
+ */
+static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
+{
+	hda_nid_t nid;
+
+	switch (nums) {
+	case 3:
+	case 4:
+		nid = pins[1];
+		pins[1] = pins[2];
+		pins[2] = nid;
+		break;
+	}
+}
+
 /*
  * Parse all pin widgets and store the useful pin nids to cfg
  *
@@ -4662,12 +4732,13 @@
  * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
  * respectively.
  */
-int snd_hda_parse_pin_def_config(struct hda_codec *codec,
-				 struct auto_pin_cfg *cfg,
-				 const hda_nid_t *ignore_nids)
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+			     struct auto_pin_cfg *cfg,
+			     const hda_nid_t *ignore_nids,
+			     unsigned int cond_flags)
 {
 	hda_nid_t nid, end_nid;
-	short seq, assoc_line_out, assoc_speaker;
+	short seq, assoc_line_out;
 	short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
 	short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
 	short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
@@ -4678,7 +4749,7 @@
 	memset(sequences_line_out, 0, sizeof(sequences_line_out));
 	memset(sequences_speaker, 0, sizeof(sequences_speaker));
 	memset(sequences_hp, 0, sizeof(sequences_hp));
-	assoc_line_out = assoc_speaker = 0;
+	assoc_line_out = 0;
 
 	end_nid = codec->start_nid + codec->num_nodes;
 	for (nid = codec->start_nid; nid < end_nid; nid++) {
@@ -4730,16 +4801,10 @@
 		case AC_JACK_SPEAKER:
 			seq = get_defcfg_sequence(def_conf);
 			assoc = get_defcfg_association(def_conf);
-			if (!assoc)
-				continue;
-			if (!assoc_speaker)
-				assoc_speaker = assoc;
-			else if (assoc_speaker != assoc)
-				continue;
 			if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
 				continue;
 			cfg->speaker_pins[cfg->speaker_outs] = nid;
-			sequences_speaker[cfg->speaker_outs] = seq;
+			sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
 			cfg->speaker_outs++;
 			break;
 		case AC_JACK_HP_OUT:
@@ -4788,7 +4853,8 @@
 	 * If no line-out is defined but multiple HPs are found,
 	 * some of them might be the real line-outs.
 	 */
-	if (!cfg->line_outs && cfg->hp_outs > 1) {
+	if (!cfg->line_outs && cfg->hp_outs > 1 &&
+	    !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
 		int i = 0;
 		while (i < cfg->hp_outs) {
 			/* The real HPs should have the sequence 0x0f */
@@ -4825,7 +4891,8 @@
 	 * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
 	 * as a primary output
 	 */
-	if (!cfg->line_outs) {
+	if (!cfg->line_outs &&
+	    !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
 		if (cfg->speaker_outs) {
 			cfg->line_outs = cfg->speaker_outs;
 			memcpy(cfg->line_out_pins, cfg->speaker_pins,
@@ -4843,21 +4910,9 @@
 		}
 	}
 
-	/* Reorder the surround channels
-	 * ALSA sequence is front/surr/clfe/side
-	 * HDA sequence is:
-	 *    4-ch: front/surr  =>  OK as it is
-	 *    6-ch: front/clfe/surr
-	 *    8-ch: front/clfe/rear/side|fc
-	 */
-	switch (cfg->line_outs) {
-	case 3:
-	case 4:
-		nid = cfg->line_out_pins[1];
-		cfg->line_out_pins[1] = cfg->line_out_pins[2];
-		cfg->line_out_pins[2] = nid;
-		break;
-	}
+	reorder_outputs(cfg->line_outs, cfg->line_out_pins);
+	reorder_outputs(cfg->hp_outs, cfg->hp_pins);
+	reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
 
 	sort_autocfg_input_pins(cfg);
 
@@ -4895,7 +4950,7 @@
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config);
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
 
 int snd_hda_get_input_pin_attr(unsigned int def_conf)
 {
@@ -5154,30 +5209,6 @@
 EXPORT_SYMBOL_HDA(snd_array_free);
 
 /**
- * snd_print_pcm_rates - Print the supported PCM rates to the string buffer
- * @pcm: PCM caps bits
- * @buf: the string buffer to write
- * @buflen: the max buffer length
- *
- * used by hda_proc.c and hda_eld.c
- */
-void snd_print_pcm_rates(int pcm, char *buf, int buflen)
-{
-	static unsigned int rates[] = {
-		8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
-		96000, 176400, 192000, 384000
-	};
-	int i, j;
-
-	for (i = 0, j = 0; i < ARRAY_SIZE(rates); i++)
-		if (pcm & (1 << i))
-			j += snprintf(buf + j, buflen - j,  " %d", rates[i]);
-
-	buf[j] = '\0'; /* necessary when j == 0 */
-}
-EXPORT_SYMBOL_HDA(snd_print_pcm_rates);
-
-/**
  * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer
  * @pcm: PCM caps bits
  * @buf: the string buffer to write
@@ -5218,6 +5249,8 @@
 		return "Mic";
 	case SND_JACK_LINEOUT:
 		return "Line-out";
+	case SND_JACK_LINEIN:
+		return "Line-in";
 	case SND_JACK_HEADSET:
 		return "Headset";
 	case SND_JACK_VIDEOOUT:
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d..1c8ddf5 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@
 	SNDRV_PCM_RATE_192000,	/* 7: 192000Hz */
 };
 
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
 					int byte_index)
 {
 	unsigned int val;
 
 	val = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_HDMI_ELDD, byte_index);
-
 #ifdef BE_PARANOID
 	printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
 #endif
-
-	if ((val & AC_ELDD_ELD_VALID) == 0) {
-		snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
-								byte_index);
-		val = 0;
-	}
-
-	return val & AC_ELDD_ELD_DATA;
+	return val;
 }
 
 #define GRAB_BITS(buf, byte, lowbit, bits) 		\
@@ -326,6 +318,11 @@
 	int size;
 	unsigned char *buf;
 
+	/*
+	 * ELD size is initialized to zero in caller function. If no errors and
+	 * ELD is valid, actual eld_size is assigned in hdmi_update_eld()
+	 */
+
 	if (!eld->eld_valid)
 		return -ENOENT;
 
@@ -335,24 +332,59 @@
 		snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n");
 		size = 128;
 	}
-	if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) {
+	if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) {
 		snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size);
 		return -ERANGE;
 	}
 
-	buf = kmalloc(size, GFP_KERNEL);
-	if (!buf)
-		return -ENOMEM;
+	/* set ELD buffer */
+	buf = eld->eld_buffer;
 
-	for (i = 0; i < size; i++)
-		buf[i] = hdmi_get_eld_byte(codec, nid, i);
+	for (i = 0; i < size; i++) {
+		unsigned int val = hdmi_get_eld_data(codec, nid, i);
+		if (!(val & AC_ELDD_ELD_VALID)) {
+			if (!i) {
+				snd_printd(KERN_INFO
+					   "HDMI: invalid ELD data\n");
+				ret = -EINVAL;
+				goto error;
+			}
+			snd_printd(KERN_INFO
+				  "HDMI: invalid ELD data byte %d\n", i);
+			val = 0;
+		} else
+			val &= AC_ELDD_ELD_DATA;
+		buf[i] = val;
+	}
 
 	ret = hdmi_update_eld(eld, buf, size);
 
-	kfree(buf);
+error:
 	return ret;
 }
 
+/**
+ * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with
+ * hdmi-specific routine.
+ */
+static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen)
+{
+	static unsigned int alsa_rates[] = {
+		5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
+		96000, 176400, 192000, 384000
+	};
+	int i, j;
+
+	for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++)
+		if (pcm & (1 << i))
+			j += snprintf(buf + j, buflen - j,  " %d",
+				alsa_rates[i]);
+
+	buf[j] = '\0'; /* necessary when j == 0 */
+}
+
+#define SND_PRINT_RATES_ADVISED_BUFSIZE	80
+
 static void hdmi_show_short_audio_desc(struct cea_sad *a)
 {
 	char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
@@ -361,7 +393,7 @@
 	if (!a->format)
 		return;
 
-	snd_print_pcm_rates(a->rates, buf, sizeof(buf));
+	hdmi_print_pcm_rates(a->rates, buf, sizeof(buf));
 
 	if (a->format == AUDIO_CODING_TYPE_LPCM)
 		snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
@@ -420,7 +452,7 @@
 			i, a->format, cea_audio_coding_type_names[a->format]);
 	snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels);
 
-	snd_print_pcm_rates(a->rates, buf, sizeof(buf));
+	hdmi_print_pcm_rates(a->rates, buf, sizeof(buf));
 	snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf);
 
 	if (a->format == AUDIO_CODING_TYPE_LPCM) {
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index bf3ced5..7e7d078 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -643,14 +643,14 @@
 static void parse_codec_mode(char *buf, struct hda_bus *bus,
 			     struct hda_codec **codecp)
 {
-	unsigned int vendorid, subid, caddr;
+	int vendorid, subid, caddr;
 	struct hda_codec *codec;
 
 	*codecp = NULL;
 	if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) {
 		list_for_each_entry(codec, &bus->codec_list, list) {
-			if (codec->vendor_id == vendorid &&
-			    codec->subsystem_id == subid &&
+			if ((vendorid <= 0 || codec->vendor_id == vendorid) &&
+			    (subid <= 0 || codec->subsystem_id == subid) &&
 			    codec->addr == caddr) {
 				*codecp = codec;
 				break;
@@ -756,8 +756,6 @@
 	}
 	if (!fw->size)
 		return 0;
-	if (size < fw->size)
-		size = fw->size;
 
 	for (len = 0; len < fw->size; len++) {
 		if (!*p)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index be69822..bd7fc99 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -34,7 +34,6 @@
  * 
  */
 
-#include <asm/io.h>
 #include <linux/delay.h>
 #include <linux/interrupt.h>
 #include <linux/kernel.h>
@@ -46,6 +45,12 @@
 #include <linux/pci.h>
 #include <linux/mutex.h>
 #include <linux/reboot.h>
+#include <linux/io.h>
+#ifdef CONFIG_X86
+/* for snoop control */
+#include <asm/pgtable.h>
+#include <asm/cacheflush.h>
+#endif
 #include <sound/core.h>
 #include <sound/initval.h>
 #include "hda_codec.h"
@@ -116,6 +121,22 @@
 MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
 #endif
 
+static int align_buffer_size = 1;
+module_param(align_buffer_size, bool, 0644);
+MODULE_PARM_DESC(align_buffer_size,
+		"Force buffer and period sizes to be multiple of 128 bytes.");
+
+#ifdef CONFIG_X86
+static bool hda_snoop = true;
+module_param_named(snoop, hda_snoop, bool, 0444);
+MODULE_PARM_DESC(snoop, "Enable/disable snooping");
+#define azx_snoop(chip)		(chip)->snoop
+#else
+#define hda_snoop		true
+#define azx_snoop(chip)		true
+#endif
+
+
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, ICH6M},"
@@ -360,7 +381,7 @@
 					 */
 	unsigned char stream_tag;	/* assigned stream */
 	unsigned char index;		/* stream index */
-	int device;			/* last device number assigned to */
+	int assigned_key;		/* last device# key assigned to */
 
 	unsigned int opened :1;
 	unsigned int running :1;
@@ -371,6 +392,7 @@
 	 *  when link position is not greater than FIFO size
 	 */
 	unsigned int insufficient :1;
+	unsigned int wc_marked:1;
 };
 
 /* CORB/RIRB */
@@ -438,6 +460,7 @@
 	unsigned int msi :1;
 	unsigned int irq_pending_warned :1;
 	unsigned int probing :1; /* codec probing phase */
+	unsigned int snoop:1;
 
 	/* for debugging */
 	unsigned int last_cmd[AZX_MAX_CODECS];
@@ -481,6 +504,7 @@
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
+#define AZX_DCAPS_BUFSIZE	(1 << 21)	/* no buffer size alignment */
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -542,6 +566,45 @@
 /* for pcm support */
 #define get_azx_dev(substream) (substream->runtime->private_data)
 
+#ifdef CONFIG_X86
+static void __mark_pages_wc(struct azx *chip, void *addr, size_t size, bool on)
+{
+	if (azx_snoop(chip))
+		return;
+	if (addr && size) {
+		int pages = (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
+		if (on)
+			set_memory_wc((unsigned long)addr, pages);
+		else
+			set_memory_wb((unsigned long)addr, pages);
+	}
+}
+
+static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
+				 bool on)
+{
+	__mark_pages_wc(chip, buf->area, buf->bytes, on);
+}
+static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
+				   struct snd_pcm_runtime *runtime, bool on)
+{
+	if (azx_dev->wc_marked != on) {
+		__mark_pages_wc(chip, runtime->dma_area, runtime->dma_bytes, on);
+		azx_dev->wc_marked = on;
+	}
+}
+#else
+/* NOP for other archs */
+static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
+				 bool on)
+{
+}
+static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
+				   struct snd_pcm_runtime *runtime, bool on)
+{
+}
+#endif
+
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
 static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
 /*
@@ -563,6 +626,7 @@
 		snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n");
 		return err;
 	}
+	mark_pages_wc(chip, &chip->rb, true);
 	return 0;
 }
 
@@ -1079,7 +1143,15 @@
 
 static void azx_init_pci(struct azx *chip)
 {
-	unsigned short snoop;
+	/* force to non-snoop mode for a new VIA controller when BIOS is set */
+	if (chip->snoop && chip->driver_type == AZX_DRIVER_VIA) {
+		u8 snoop;
+		pci_read_config_byte(chip->pci, 0x42, &snoop);
+		if (!(snoop & 0x80) && chip->pci->revision == 0x30) {
+			chip->snoop = 0;
+			snd_printdd(SFX "Force to non-snoop mode\n");
+		}
+	}
 
 	/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
 	 * TCSEL == Traffic Class Select Register, which sets PCI express QOS
@@ -1096,15 +1168,15 @@
 	 * we need to enable snoop.
 	 */
 	if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) {
-		snd_printdd(SFX "Enabling ATI snoop\n");
+		snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip));
 		update_pci_byte(chip->pci,
-				ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 
-				0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+				ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07,
+				azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0);
 	}
 
 	/* For NVIDIA HDA, enable snoop */
 	if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) {
-		snd_printdd(SFX "Enabling Nvidia snoop\n");
+		snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip));
 		update_pci_byte(chip->pci,
 				NVIDIA_HDA_TRANSREG_ADDR,
 				0x0f, NVIDIA_HDA_ENABLE_COHBITS);
@@ -1118,16 +1190,20 @@
 
 	/* Enable SCH/PCH snoop if needed */
 	if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) {
+		unsigned short snoop;
 		pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
-		if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
-			pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC,
-				snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP));
+		if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) ||
+		    (azx_snoop(chip) && (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP))) {
+			snoop &= ~INTEL_SCH_HDA_DEVC_NOSNOOP;
+			if (!azx_snoop(chip))
+				snoop |= INTEL_SCH_HDA_DEVC_NOSNOOP;
+			pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, snoop);
 			pci_read_config_word(chip->pci,
 				INTEL_SCH_HDA_DEVC, &snoop);
-			snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n",
-				(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
-				? "Failed" : "OK");
 		}
+		snd_printdd(SFX "SCH snoop: %s\n",
+				(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
+				? "Disabled" : "Enabled");
         }
 }
 
@@ -1334,12 +1410,16 @@
  */
 static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 {
+	unsigned int val;
 	/* make sure the run bit is zero for SD */
 	azx_stream_clear(chip, azx_dev);
 	/* program the stream_tag */
-	azx_sd_writel(azx_dev, SD_CTL,
-		      (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)|
-		      (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT));
+	val = azx_sd_readl(azx_dev, SD_CTL);
+	val = (val & ~SD_CTL_STREAM_TAG_MASK) |
+		(azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT);
+	if (!azx_snoop(chip))
+		val |= SD_CTL_TRAFFIC_PRIO;
+	azx_sd_writel(azx_dev, SD_CTL, val);
 
 	/* program the length of samples in cyclic buffer */
 	azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize);
@@ -1533,6 +1613,9 @@
 {
 	int dev, i, nums;
 	struct azx_dev *res = NULL;
+	/* make a non-zero unique key for the substream */
+	int key = (substream->pcm->device << 16) | (substream->number << 2) |
+		(substream->stream + 1);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		dev = chip->playback_index_offset;
@@ -1544,12 +1627,12 @@
 	for (i = 0; i < nums; i++, dev++)
 		if (!chip->azx_dev[dev].opened) {
 			res = &chip->azx_dev[dev];
-			if (res->device == substream->pcm->device)
+			if (res->assigned_key == key)
 				break;
 		}
 	if (res) {
 		res->opened = 1;
-		res->device = substream->pcm->device;
+		res->assigned_key = key;
 	}
 	return res;
 }
@@ -1599,6 +1682,7 @@
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned long flags;
 	int err;
+	int buff_step;
 
 	mutex_lock(&chip->open_mutex);
 	azx_dev = azx_assign_device(chip, substream);
@@ -1613,10 +1697,25 @@
 	runtime->hw.rates = hinfo->rates;
 	snd_pcm_limit_hw_rates(runtime);
 	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if (align_buffer_size)
+		/* constrain buffer sizes to be multiple of 128
+		   bytes. This is more efficient in terms of memory
+		   access but isn't required by the HDA spec and
+		   prevents users from specifying exact period/buffer
+		   sizes. For example for 44.1kHz, a period size set
+		   to 20ms will be rounded to 19.59ms. */
+		buff_step = 128;
+	else
+		/* Don't enforce steps on buffer sizes, still need to
+		   be multiple of 4 bytes (HDA spec). Tested on Intel
+		   HDA controllers, may not work on all devices where
+		   option needs to be disabled */
+		buff_step = 4;
+
 	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
-				   128);
+				   buff_step);
 	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
-				   128);
+				   buff_step);
 	snd_hda_power_up(apcm->codec);
 	err = hinfo->ops.open(hinfo, apcm->codec, substream);
 	if (err < 0) {
@@ -1671,19 +1770,30 @@
 static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *hw_params)
 {
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	struct azx *chip = apcm->chip;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct azx_dev *azx_dev = get_azx_dev(substream);
+	int ret;
 
+	mark_runtime_wc(chip, azx_dev, runtime, false);
 	azx_dev->bufsize = 0;
 	azx_dev->period_bytes = 0;
 	azx_dev->format_val = 0;
-	return snd_pcm_lib_malloc_pages(substream,
+	ret = snd_pcm_lib_malloc_pages(substream,
 					params_buffer_bytes(hw_params));
+	if (ret < 0)
+		return ret;
+	mark_runtime_wc(chip, azx_dev, runtime, true);
+	return ret;
 }
 
 static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
 	struct azx_dev *azx_dev = get_azx_dev(substream);
+	struct azx *chip = apcm->chip;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
 
 	/* reset BDL address */
@@ -1696,6 +1806,7 @@
 
 	snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
 
+	mark_runtime_wc(chip, azx_dev, runtime, false);
 	return snd_pcm_lib_free_pages(substream);
 }
 
@@ -1924,7 +2035,8 @@
 }
 
 static unsigned int azx_get_position(struct azx *chip,
-				     struct azx_dev *azx_dev)
+				     struct azx_dev *azx_dev,
+				     bool with_check)
 {
 	unsigned int pos;
 	int stream = azx_dev->substream->stream;
@@ -1940,7 +2052,7 @@
 	default:
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
-		if (chip->position_fix[stream] == POS_FIX_AUTO) {
+		if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
 			if (!pos || pos == (u32)-1) {
 				printk(KERN_WARNING
 				       "hda-intel: Invalid position buffer, "
@@ -1964,7 +2076,7 @@
 	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));
+			       azx_get_position(chip, azx_dev, false));
 }
 
 /*
@@ -1987,7 +2099,7 @@
 		return -1;	/* bogus (too early) interrupt */
 
 	stream = azx_dev->substream->stream;
-	pos = azx_get_position(chip, azx_dev);
+	pos = azx_get_position(chip, azx_dev, true);
 
 	if (WARN_ONCE(!azx_dev->period_bytes,
 		      "hda-intel: zero azx_dev->period_bytes"))
@@ -2054,6 +2166,20 @@
 	spin_unlock_irq(&chip->reg_lock);
 }
 
+#ifdef CONFIG_X86
+static int azx_pcm_mmap(struct snd_pcm_substream *substream,
+			struct vm_area_struct *area)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	struct azx *chip = apcm->chip;
+	if (!azx_snoop(chip))
+		area->vm_page_prot = pgprot_writecombine(area->vm_page_prot);
+	return snd_pcm_lib_default_mmap(substream, area);
+}
+#else
+#define azx_pcm_mmap	NULL
+#endif
+
 static struct snd_pcm_ops azx_pcm_ops = {
 	.open = azx_pcm_open,
 	.close = azx_pcm_close,
@@ -2063,6 +2189,7 @@
 	.prepare = azx_pcm_prepare,
 	.trigger = azx_pcm_trigger,
 	.pointer = azx_pcm_pointer,
+	.mmap = azx_pcm_mmap,
 	.page = snd_pcm_sgbuf_ops_page,
 };
 
@@ -2343,13 +2470,19 @@
 
 	if (chip->azx_dev) {
 		for (i = 0; i < chip->num_streams; i++)
-			if (chip->azx_dev[i].bdl.area)
+			if (chip->azx_dev[i].bdl.area) {
+				mark_pages_wc(chip, &chip->azx_dev[i].bdl, false);
 				snd_dma_free_pages(&chip->azx_dev[i].bdl);
+			}
 	}
-	if (chip->rb.area)
+	if (chip->rb.area) {
+		mark_pages_wc(chip, &chip->rb, false);
 		snd_dma_free_pages(&chip->rb);
-	if (chip->posbuf.area)
+	}
+	if (chip->posbuf.area) {
+		mark_pages_wc(chip, &chip->posbuf, false);
 		snd_dma_free_pages(&chip->posbuf);
+	}
 	pci_release_regions(chip->pci);
 	pci_disable_device(chip->pci);
 	kfree(chip->azx_dev);
@@ -2369,6 +2502,7 @@
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1028, 0x02c6, "Dell Inspiron 1010", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
@@ -2544,6 +2678,7 @@
 	check_probe_mask(chip, dev);
 
 	chip->single_cmd = single_cmd;
+	chip->snoop = hda_snoop;
 
 	if (bdl_pos_adj[dev] < 0) {
 		switch (chip->driver_type) {
@@ -2616,6 +2751,10 @@
 		gcap &= ~ICH6_GCAP_64OK;
 	}
 
+	/* disable buffer size rounding to 128-byte multiples if supported */
+	if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
+		align_buffer_size = 0;
+
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
 		pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
@@ -2667,6 +2806,7 @@
 			snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
 			goto errout;
 		}
+		mark_pages_wc(chip, &chip->azx_dev[i].bdl, true);
 	}
 	/* allocate memory for the position buffer */
 	err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
@@ -2676,6 +2816,7 @@
 		snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
 		goto errout;
 	}
+	mark_pages_wc(chip, &chip->posbuf, true);
 	/* allocate CORB/RIRB */
 	err = azx_alloc_cmd_io(chip);
 	if (err < 0)
@@ -2817,37 +2958,49 @@
 static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* CPT */
 	{ PCI_DEVICE(0x8086, 0x1c20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE },
 	/* PBG */
 	{ PCI_DEVICE(0x8086, 0x1d20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE},
 	/* Panther Point */
 	{ PCI_DEVICE(0x8086, 0x1e20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE},
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
-	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
+	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE},
 	{ PCI_DEVICE(0x8086, 0x2668),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH6 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH6 */
 	{ PCI_DEVICE(0x8086, 0x27d8),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH7 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH7 */
 	{ PCI_DEVICE(0x8086, 0x269a),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ESB2 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ESB2 */
 	{ PCI_DEVICE(0x8086, 0x284b),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH8 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH8 */
 	{ PCI_DEVICE(0x8086, 0x293e),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH9 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH9 */
 	{ PCI_DEVICE(0x8086, 0x293f),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH9 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH9 */
 	{ PCI_DEVICE(0x8086, 0x3a3e),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH10 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH10 */
 	{ PCI_DEVICE(0x8086, 0x3a6e),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH10 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+	  AZX_DCAPS_BUFSIZE },  /* ICH10 */
 	/* Generic Intel */
 	{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
-	  .driver_data = AZX_DRIVER_ICH },
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE },
 	/* ATI SB 450/600/700/800/900 */
 	{ PCI_DEVICE(0x1002, 0x437b),
 	  .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 2e7ac31..79f49e2 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -267,11 +267,14 @@
 enum { HDA_FRONT, HDA_REAR, HDA_CLFE, HDA_SIDE }; /* index for dac_nidx */
 enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */
 
+#define HDA_MAX_OUTS	5
+
 struct hda_multi_out {
 	int num_dacs;		/* # of DACs, must be more than 1 */
 	const hda_nid_t *dac_nids;	/* DAC list */
 	hda_nid_t hp_nid;	/* optional DAC for HP, 0 when not exists */
-	hda_nid_t extra_out_nid[3];	/* optional DACs, 0 when not exists */
+	hda_nid_t hp_out_nid[HDA_MAX_OUTS];	/* DACs for multiple HPs */
+	hda_nid_t extra_out_nid[HDA_MAX_OUTS];	/* other (e.g. speaker) DACs */
 	hda_nid_t dig_out_nid;	/* digital out audio widget */
 	const hda_nid_t *slave_dig_outs;
 	int max_channels;	/* currently supported analog channels */
@@ -333,9 +336,6 @@
 static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
 #endif
 
-#define SND_PRINT_RATES_ADVISED_BUFSIZE	80
-void snd_print_pcm_rates(int pcm, char *buf, int buflen);
-
 #define SND_PRINT_BITS_ADVISED_BUFSIZE	16
 void snd_print_pcm_bits(int pcm, char *buf, int buflen);
 
@@ -385,7 +385,7 @@
 	AUTO_PIN_HP_OUT
 };
 
-#define AUTO_CFG_MAX_OUTS	5
+#define AUTO_CFG_MAX_OUTS	HDA_MAX_OUTS
 #define AUTO_CFG_MAX_INS	8
 
 struct auto_pin_cfg_item {
@@ -442,10 +442,21 @@
 	(cfg & AC_DEFCFG_SEQUENCE)
 #define get_defcfg_device(cfg) \
 	((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+#define get_defcfg_misc(cfg) \
+	((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT)
 
-int snd_hda_parse_pin_def_config(struct hda_codec *codec,
-				 struct auto_pin_cfg *cfg,
-				 const hda_nid_t *ignore_nids);
+/* bit-flags for snd_hda_parse_pin_def_config() behavior */
+#define HDA_PINCFG_NO_HP_FIXUP	(1 << 0) /* no HP-split */
+#define HDA_PINCFG_NO_LO_FIXUP	(1 << 1) /* don't take other outs as LO */
+
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+			     struct auto_pin_cfg *cfg,
+			     const hda_nid_t *ignore_nids,
+			     unsigned int cond_flags);
+
+/* older function */
+#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
+	snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
 
 /* amp values */
 #define AMP_IN_MUTE(idx)	(0x7080 | ((idx)<<8))
@@ -492,12 +503,16 @@
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps);
 u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
+			      unsigned int caps);
 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
 int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
 
 static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
 {
 	return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) &&
+		!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid) &
+		  AC_DEFCFG_MISC_NO_PRESENCE)) &&
 		(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP);
 }
 
@@ -589,7 +604,8 @@
 #define get_amp_nid_(pv)	((pv) & 0xffff)
 #define get_amp_nid(kc)		get_amp_nid_((kc)->private_value)
 #define get_amp_channels(kc)	(((kc)->private_value >> 16) & 0x3)
-#define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
+#define get_amp_direction_(pv)	(((pv) >> 18) & 0x1)
+#define get_amp_direction(kc)	get_amp_direction_((kc)->private_value)
 #define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
 #define get_amp_offset(kc)	(((kc)->private_value >> 23) & 0x3f)
 #define get_amp_min_mute(kc)	(((kc)->private_value >> 29) & 0x1)
@@ -607,6 +623,7 @@
 };
 
 #define ELD_FIXED_BYTES	20
+#define ELD_MAX_SIZE    256
 #define ELD_MAX_MNL	16
 #define ELD_MAX_SAD	16
 
@@ -631,6 +648,7 @@
 	int	spk_alloc;
 	int	sad_count;
 	struct cea_sad sad[ELD_MAX_SAD];
+	char    eld_buffer[ELD_MAX_SIZE];
 #ifdef CONFIG_PROC_FS
 	struct snd_info_entry *proc_entry;
 #endif
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 2be57b0..2c981b5 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -152,12 +152,18 @@
 
 static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm)
 {
-	char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
+	static unsigned int rates[] = {
+		8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
+		96000, 176400, 192000, 384000
+	};
+	int i;
 
 	pcm &= AC_SUPPCM_RATES;
 	snd_iprintf(buffer, "    rates [0x%x]:", pcm);
-	snd_print_pcm_rates(pcm, buf, sizeof(buf));
-	snd_iprintf(buffer, "%s\n", buf);
+	for (i = 0; i < ARRAY_SIZE(rates); i++)
+		if (pcm & (1 << i))
+			snd_iprintf(buffer,  " %d", rates[i]);
+	snd_iprintf(buffer, "\n");
 }
 
 static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm)
diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h
new file mode 100644
index 0000000..9884871
--- /dev/null
+++ b/sound/pci/hda/hda_trace.h
@@ -0,0 +1,117 @@
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM hda
+#define TRACE_INCLUDE_FILE hda_trace
+
+#if !defined(_TRACE_HDA_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _TRACE_HDA_H
+
+#include <linux/tracepoint.h>
+
+struct hda_bus;
+struct hda_codec;
+
+DECLARE_EVENT_CLASS(hda_cmd,
+
+	TP_PROTO(struct hda_codec *codec, unsigned int val),
+
+	TP_ARGS(codec, val),
+
+	TP_STRUCT__entry(
+		__field( unsigned int, card )
+		__field( unsigned int, addr )
+		__field( unsigned int, val )
+	),
+
+	TP_fast_assign(
+		__entry->card = (codec)->bus->card->number;
+		__entry->addr = (codec)->addr;
+		__entry->val = (val);
+	),
+
+	TP_printk("[%d:%d] val=%x", __entry->card, __entry->addr, __entry->val)
+);
+
+DEFINE_EVENT(hda_cmd, hda_send_cmd,
+	TP_PROTO(struct hda_codec *codec, unsigned int val),
+	TP_ARGS(codec, val)
+);
+
+DEFINE_EVENT(hda_cmd, hda_get_response,
+	TP_PROTO(struct hda_codec *codec, unsigned int val),
+	TP_ARGS(codec, val)
+);
+
+TRACE_EVENT(hda_bus_reset,
+
+	TP_PROTO(struct hda_bus *bus),
+
+	TP_ARGS(bus),
+
+	TP_STRUCT__entry(
+		__field( unsigned int, card )
+	),
+
+	TP_fast_assign(
+		__entry->card = (bus)->card->number;
+	),
+
+	TP_printk("[%d]", __entry->card)
+);
+
+DECLARE_EVENT_CLASS(hda_power,
+
+	TP_PROTO(struct hda_codec *codec),
+
+	TP_ARGS(codec),
+
+	TP_STRUCT__entry(
+		__field( unsigned int, card )
+		__field( unsigned int, addr )
+	),
+
+	TP_fast_assign(
+		__entry->card = (codec)->bus->card->number;
+		__entry->addr = (codec)->addr;
+	),
+
+	TP_printk("[%d:%d]", __entry->card, __entry->addr)
+);
+
+DEFINE_EVENT(hda_power, hda_power_down,
+	TP_PROTO(struct hda_codec *codec),
+	TP_ARGS(codec)
+);
+
+DEFINE_EVENT(hda_power, hda_power_up,
+	TP_PROTO(struct hda_codec *codec),
+	TP_ARGS(codec)
+);
+
+TRACE_EVENT(hda_unsol_event,
+
+	TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex),
+
+	TP_ARGS(bus, res, res_ex),
+
+	TP_STRUCT__entry(
+		__field( unsigned int, card )
+		__field( u32, res )
+		__field( u32, res_ex )
+	),
+
+	TP_fast_assign(
+		__entry->card = (bus)->card->number;
+		__entry->res = res;
+		__entry->res_ex = res_ex;
+	),
+
+	TP_printk("[%d] res=%x, res_ex=%x", __entry->card,
+		  __entry->res, __entry->res_ex)
+);
+
+#endif /* _TRACE_HDA_H */
+
+/* This part must be outside protection */
+#undef TRACE_INCLUDE_PATH
+#define TRACE_INCLUDE_PATH .
+#include <trace/define_trace.h>
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 8648917..d8aac58 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -48,6 +48,8 @@
 
 	const hda_nid_t *alt_dac_nid;
 	const struct hda_pcm_stream *stream_analog_alt_playback;
+	int independent_hp;
+	int num_active_streams;
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -302,6 +304,72 @@
 }
 #endif
 
+static void activate_ctl(struct hda_codec *codec, const char *name, int active)
+{
+	struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name);
+	if (ctl) {
+		ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+		ctl->vd[0].access |= active ? 0 :
+			SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+		ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE;
+		ctl->vd[0].access |= active ?
+			SNDRV_CTL_ELEM_ACCESS_WRITE : 0;
+		snd_ctl_notify(codec->bus->card,
+			       SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
+	}
+}
+
+static void set_stream_active(struct hda_codec *codec, bool active)
+{
+	struct ad198x_spec *spec = codec->spec;
+	if (active)
+		spec->num_active_streams++;
+	else
+		spec->num_active_streams--;
+	activate_ctl(codec, "Independent HP", spec->num_active_streams == 0);
+}
+
+static int ad1988_independent_hp_info(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[] = { "OFF", "ON", NULL};
+	int index;
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 2;
+	index = uinfo->value.enumerated.item;
+	if (index >= 2)
+		index = 1;
+	strcpy(uinfo->value.enumerated.name, texts[index]);
+	return 0;
+}
+
+static int ad1988_independent_hp_get(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad198x_spec *spec = codec->spec;
+	ucontrol->value.enumerated.item[0] = spec->independent_hp;
+	return 0;
+}
+
+static int ad1988_independent_hp_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad198x_spec *spec = codec->spec;
+	unsigned int select = ucontrol->value.enumerated.item[0];
+	if (spec->independent_hp != select) {
+		spec->independent_hp = select;
+		if (spec->independent_hp)
+			spec->multiout.hp_nid = 0;
+		else
+			spec->multiout.hp_nid = spec->alt_dac_nid[0];
+		return 1;
+	}
+	return 0;
+}
+
 /*
  * Analog playback callbacks
  */
@@ -310,8 +378,15 @@
 				    struct snd_pcm_substream *substream)
 {
 	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+	int err;
+	set_stream_active(codec, true);
+	err = snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
 					     hinfo);
+	if (err < 0) {
+		set_stream_active(codec, false);
+		return err;
+	}
+	return 0;
 }
 
 static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -333,11 +408,41 @@
 	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
+static int ad198x_playback_pcm_close(struct hda_pcm_stream *hinfo,
+				 struct hda_codec *codec,
+				 struct snd_pcm_substream *substream)
+{
+	set_stream_active(codec, false);
+	return 0;
+}
+
+static int ad1988_alt_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				 struct hda_codec *codec,
+				 struct snd_pcm_substream *substream)
+{
+	struct ad198x_spec *spec = codec->spec;
+	if (!spec->independent_hp)
+		return -EBUSY;
+	set_stream_active(codec, true);
+	return 0;
+}
+
+static int ad1988_alt_playback_pcm_close(struct hda_pcm_stream *hinfo,
+				 struct hda_codec *codec,
+				 struct snd_pcm_substream *substream)
+{
+	set_stream_active(codec, false);
+	return 0;
+}
+
 static const struct hda_pcm_stream ad198x_pcm_analog_alt_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	/* NID is set in ad198x_build_pcms */
+	.ops = {
+		.open  = ad1988_alt_playback_pcm_open,
+		.close = ad1988_alt_playback_pcm_close
+	},
 };
 
 /*
@@ -402,7 +507,6 @@
 	return 0;
 }
 
-
 /*
  */
 static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
@@ -413,7 +517,8 @@
 	.ops = {
 		.open = ad198x_playback_pcm_open,
 		.prepare = ad198x_playback_pcm_prepare,
-		.cleanup = ad198x_playback_pcm_cleanup
+		.cleanup = ad198x_playback_pcm_cleanup,
+		.close = ad198x_playback_pcm_close
 	},
 };
 
@@ -2058,7 +2163,6 @@
 enum {
 	AD1988_6STACK,
 	AD1988_6STACK_DIG,
-	AD1988_6STACK_DIG_FP,
 	AD1988_3STACK,
 	AD1988_3STACK_DIG,
 	AD1988_LAPTOP,
@@ -2168,6 +2272,17 @@
 	return err;
 }
 
+static const struct snd_kcontrol_new ad1988_hp_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Independent HP",
+		.info = ad1988_independent_hp_info,
+		.get = ad1988_independent_hp_get,
+		.put = ad1988_independent_hp_put,
+	},
+	{ } /* end */
+};
+
 /* 6-stack mode */
 static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
@@ -2188,6 +2303,7 @@
 };
 
 static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
 	HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
 	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
@@ -2210,13 +2326,6 @@
 
 	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-
 	{ } /* end */
 };
 
@@ -2238,6 +2347,7 @@
 };
 
 static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
 	HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
 	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
@@ -2272,6 +2382,7 @@
 
 /* laptop mode */
 static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
 	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
@@ -2446,7 +2557,7 @@
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
+	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
 	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2594,7 +2705,7 @@
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
+	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
 	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2669,7 +2780,7 @@
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
+	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
 	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2782,11 +2893,11 @@
 {
 	static const hda_nid_t idx_to_dac[8] = {
 		/* A     B     C     D     E     F     G     H */
-		0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
+		0x03, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
 	};
 	static const hda_nid_t idx_to_dac_rev2[8] = {
 		/* A     B     C     D     E     F     G     H */
-		0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
+		0x03, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
 	};
 	if (is_rev2(codec))
 		return idx_to_dac_rev2[idx];
@@ -3023,8 +3134,8 @@
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
 	switch (nid) {
-	case 0x11: /* port-A - DAC 04 */
-		snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x01);
+	case 0x11: /* port-A - DAC 03 */
+		snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x00);
 		break;
 	case 0x14: /* port-B - DAC 06 */
 		snd_hda_codec_write(codec, 0x30, 0, AC_VERB_SET_CONNECT_SEL, 0x02);
@@ -3150,7 +3261,6 @@
 static const char * const ad1988_models[AD1988_MODEL_LAST] = {
 	[AD1988_6STACK]		= "6stack",
 	[AD1988_6STACK_DIG]	= "6stack-dig",
-	[AD1988_6STACK_DIG_FP]	= "6stack-dig-fp",
 	[AD1988_3STACK]		= "3stack",
 	[AD1988_3STACK_DIG]	= "3stack-dig",
 	[AD1988_LAPTOP]		= "laptop",
@@ -3208,10 +3318,11 @@
 	}
 	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
 
+	if (!spec->multiout.hp_nid)
+		spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
 	switch (board_config) {
 	case AD1988_6STACK:
 	case AD1988_6STACK_DIG:
-	case AD1988_6STACK_DIG_FP:
 		spec->multiout.max_channels = 8;
 		spec->multiout.num_dacs = 4;
 		if (is_rev2(codec))
@@ -3227,19 +3338,7 @@
 		spec->mixers[1] = ad1988_6stack_mixers2;
 		spec->num_init_verbs = 1;
 		spec->init_verbs[0] = ad1988_6stack_init_verbs;
-		if (board_config == AD1988_6STACK_DIG_FP) {
-			spec->num_mixers++;
-			spec->mixers[2] = ad1988_6stack_fp_mixers;
-			spec->num_init_verbs++;
-			spec->init_verbs[1] = ad1988_6stack_fp_init_verbs;
-			spec->slave_vols = ad1988_6stack_fp_slave_vols;
-			spec->slave_sws = ad1988_6stack_fp_slave_sws;
-			spec->alt_dac_nid = ad1988_alt_dac_nid;
-			spec->stream_analog_alt_playback =
-				&ad198x_pcm_analog_alt_playback;
-		}
-		if ((board_config == AD1988_6STACK_DIG) ||
-			(board_config == AD1988_6STACK_DIG_FP)) {
+		if (board_config == AD1988_6STACK_DIG) {
 			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
 			spec->dig_in_nid = AD1988_SPDIF_IN;
 		}
@@ -3282,6 +3381,15 @@
 		break;
 	}
 
+	if (spec->autocfg.hp_pins[0]) {
+		spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
+		spec->slave_vols = ad1988_6stack_fp_slave_vols;
+		spec->slave_sws = ad1988_6stack_fp_slave_sws;
+		spec->alt_dac_nid = ad1988_alt_dac_nid;
+		spec->stream_analog_alt_playback =
+			&ad198x_pcm_analog_alt_playback;
+	}
+
 	spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
 	spec->adc_nids = ad1988_adc_nids;
 	spec->capsrc_nids = ad1988_capsrc_nids;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc..c45f3e6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@
 static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int *idxp)
 {
-	int i;
+	int i, idx;
 	hda_nid_t nid;
 
 	nid = codec->start_nid;
@@ -384,9 +384,11 @@
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
-		if (*idxp >= 0)
+		idx = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (idx >= 0) {
+			*idxp = idx;
 			return nid;
+		}
 	}
 	return 0;
 }
@@ -533,7 +535,7 @@
 		      int index, unsigned int pval, int dir,
 		      struct snd_kcontrol **kctlp)
 {
-	char tmp[32];
+	char tmp[44];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..0c8b5a1 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -136,6 +136,8 @@
 	unsigned int thinkpad:1;
 	unsigned int hp_laptop:1;
 	unsigned int asus:1;
+	unsigned int pin_eapd_ctrls:1;
+	unsigned int single_adc_amp:1;
 
 	unsigned int adc_switching:1;
 
@@ -1867,39 +1869,6 @@
 	{ } /* end */
 };
 
-static const struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
-	/* Line in, Mic */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
-	/* SPK  */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* HP, Amp  */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Docking HP */
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* DAC1 */
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Record selector: Internal mic */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
-	/* SPDIF route: PCM */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* needed for W500 Advanced Mini Dock 250410 */
-	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* EAPD */
-	{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
-	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
-	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
-	{ } /* end */
-};
-
 static const struct hda_verb cxt5051_f700_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
@@ -1968,7 +1937,6 @@
 	CXT5051_LAPTOP,	 /* Laptops w/ EAPD support */
 	CXT5051_HP,	/* no docking */
 	CXT5051_HP_DV6736,	/* HP without mic switch */
-	CXT5051_LENOVO_X200,	/* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */
 	CXT5051_F700,       /* HP Compaq Presario F700 */
 	CXT5051_TOSHIBA,	/* Toshiba M300 & co */
 	CXT5051_IDEAPAD,	/* Lenovo IdeaPad Y430 */
@@ -1980,7 +1948,6 @@
 	[CXT5051_LAPTOP]	= "laptop",
 	[CXT5051_HP]		= "hp",
 	[CXT5051_HP_DV6736]	= "hp-dv6736",
-	[CXT5051_LENOVO_X200]	= "lenovo-x200",
 	[CXT5051_F700]          = "hp-700",
 	[CXT5051_TOSHIBA]	= "toshiba",
 	[CXT5051_IDEAPAD]	= "ideapad",
@@ -1995,7 +1962,6 @@
 	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
-	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD),
 	{}
 };
@@ -2053,13 +2019,6 @@
 		spec->mixers[0] = cxt5051_hp_dv6736_mixers;
 		spec->auto_mic = 0;
 		break;
-	case CXT5051_LENOVO_X200:
-		spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
-		/* Thinkpad X301 does not have S/PDIF wired and no ability
-		   to use a docking station. */
-		if (codec->subsystem_id == 0x17aa211f)
-			spec->multiout.dig_out_nid = 0;
-		break;
 	case CXT5051_F700:
 		spec->init_verbs[0] = cxt5051_f700_init_verbs;
 		spec->mixers[0] = cxt5051_f700_mixers;
@@ -3110,6 +3069,7 @@
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
@@ -3348,6 +3308,8 @@
 
 #define MAX_AUTO_DACS	5
 
+#define DAC_SLAVE_FLAG	0x8000	/* filled dac is a slave */
+
 /* fill analog DAC list from the widget tree */
 static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 {
@@ -3370,16 +3332,26 @@
 /* fill pin_dac_pair list from the pin and dac list */
 static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 			      int num_pins, hda_nid_t *dacs, int *rest,
-			      struct pin_dac_pair *filled, int type)
+			      struct pin_dac_pair *filled, int nums, 
+			      int type)
 {
-	int i, nums;
+	int i, start = nums;
 
-	nums = 0;
-	for (i = 0; i < num_pins; i++) {
+	for (i = 0; i < num_pins; i++, nums++) {
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
-		nums++;
+		if (filled[nums].dac) 
+			continue;
+		if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+			filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
 	}
 	return nums;
 }
@@ -3395,19 +3367,19 @@
 	rest = fill_cx_auto_dacs(codec, dacs);
 	/* parse all analog output pins */
 	nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
-				  dacs, &rest, spec->dac_info,
-				  AUTO_PIN_LINE_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_HP_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_SPEAKER_OUT);
+			  dacs, &rest, spec->dac_info, 0,
+			  AUTO_PIN_LINE_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_HP_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_SPEAKER_OUT);
 	spec->dac_info_filled = nums;
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac)
+		if (!dac || (dac & DAC_SLAVE_FLAG))
 			continue;
 		switch (spec->dac_info[i].type) {
 		case AUTO_PIN_LINE_OUT:
@@ -3460,12 +3432,14 @@
 static void do_automute(struct hda_codec *codec, int num_pins,
 			hda_nid_t *pins, bool on)
 {
+	struct conexant_spec *spec = codec->spec;
 	int i;
 	for (i = 0; i < num_pins; i++)
 		snd_hda_codec_write(codec, pins[i], 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL,
 				    on ? PIN_OUT : 0);
-	cx_auto_turn_eapd(codec, num_pins, pins, on);
+	if (spec->pin_eapd_ctrls)
+		cx_auto_turn_eapd(codec, num_pins, pins, on);
 }
 
 static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
@@ -3490,9 +3464,12 @@
 	int on = 1;
 
 	/* turn on HP EAPD when HP jacks are present */
-	if (spec->auto_mute)
-		on = spec->hp_present;
-	cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on);
+	if (spec->pin_eapd_ctrls) {
+		if (spec->auto_mute)
+			on = spec->hp_present;
+		cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on);
+	}
+
 	/* mute speakers in auto-mode if HP or LO jacks are plugged */
 	if (spec->auto_mute)
 		on = !(spec->hp_present ||
@@ -3862,7 +3839,7 @@
 	}
 	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1 && !spec->auto_mic) {
+	if (imux->num_items > 1) {
 		for (i = 1; i < imux->num_items; i++) {
 			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
 				spec->adc_switching = 1;
@@ -3919,20 +3896,10 @@
 #define cx_auto_parse_beep(codec)
 #endif
 
-static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
-{
-	int i;
-	for (i = 0; i < nums; i++)
-		if (list[i] == nid)
-			return true;
-	return false;
-}
-
-/* parse extra-EAPD that aren't assigned to any pins */
+/* parse EAPDs */
 static void cx_auto_parse_eapd(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
 	hda_nid_t nid, end_nid;
 
 	end_nid = codec->start_nid + codec->num_nodes;
@@ -3941,14 +3908,18 @@
 			continue;
 		if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD))
 			continue;
-		if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
-		    found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
-		    found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs))
-			continue;
 		spec->eapds[spec->num_eapds++] = nid;
 		if (spec->num_eapds >= ARRAY_SIZE(spec->eapds))
 			break;
 	}
+
+	/* NOTE: below is a wild guess; if we have more than two EAPDs,
+	 * it's a new chip, where EAPDs are supposed to be associated to
+	 * pins, and we can control EAPD per pin.
+	 * OTOH, if only one or two EAPDs are found, it's an old chip,
+	 * thus it might control over all pins.
+	 */
+	spec->pin_eapd_ctrls = spec->num_eapds > 2;
 }
 
 static int cx_auto_parse_auto_config(struct hda_codec *codec)
@@ -4035,6 +4006,8 @@
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
+		else if (nid & DAC_SLAVE_FLAG)
+			nid &= ~DAC_SLAVE_FLAG;
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
 	if (spec->auto_mute) {
@@ -4052,8 +4025,9 @@
 		}
 	}
 	cx_auto_update_speakers(codec);
-	/* turn on/off extra EAPDs, too */
-	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+	/* turn on all EAPDs if no individual EAPD control is available */
+	if (!spec->pin_eapd_ctrls)
+		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
 }
 
 static void cx_auto_init_input(struct hda_codec *codec)
@@ -4167,9 +4141,11 @@
 			     hda_nid_t pin, const char *name, int idx)
 {
 	unsigned int caps;
-	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
-	if (caps & AC_AMPCAP_NUM_STEPS)
-		return cx_auto_add_pb_volume(codec, dac, name, idx);
+	if (dac && !(dac & DAC_SLAVE_FLAG)) {
+		caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+		if (caps & AC_AMPCAP_NUM_STEPS)
+			return cx_auto_add_pb_volume(codec, dac, name, idx);
+	}
 	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
 	if (caps & AC_AMPCAP_NUM_STEPS)
 		return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4167,7 @@
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
-		if (!spec->dac_info[i].dac)
-			continue;
+		hda_nid_t dac = spec->dac_info[i].dac;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
 			type = spec->autocfg.line_out_type;
@@ -4211,7 +4186,7 @@
 			idx = num_spk++;
 			break;
 		}
-		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+		err = try_add_pb_volume(codec, dac,
 					spec->dac_info[i].pin,
 					label, idx);
 		if (err < 0)
@@ -4239,6 +4214,8 @@
 		int idx = get_input_connection(codec, adc_nid, nid);
 		if (idx < 0)
 			continue;
+		if (spec->single_adc_amp)
+			idx = 0;
 		return cx_auto_add_volume_idx(codec, label, pfx,
 					      cidx, adc_nid, HDA_INPUT, idx);
 	}
@@ -4279,14 +4256,21 @@
 	struct hda_input_mux *imux = &spec->private_imux;
 	const char *prev_label;
 	int input_conn[HDA_MAX_NUM_INPUTS];
-	int i, err, cidx;
+	int i, j, err, cidx;
 	int multi_connection;
 
+	if (!imux->num_items)
+		return 0;
+
 	multi_connection = 0;
 	for (i = 0; i < imux->num_items; i++) {
 		cidx = get_input_connection(codec, spec->imux_info[i].adc,
 					    spec->imux_info[i].pin);
-		input_conn[i] = (spec->imux_info[i].adc << 8) | cidx;
+		if (cidx < 0)
+			continue;
+		input_conn[i] = spec->imux_info[i].adc;
+		if (!spec->single_adc_amp)
+			input_conn[i] |= cidx << 8;
 		if (i > 0 && input_conn[i] != input_conn[0])
 			multi_connection = 1;
 	}
@@ -4315,6 +4299,15 @@
 			err = cx_auto_add_capture_volume(codec, nid,
 							 "Capture", "", cidx);
 		} else {
+			bool dup_found = false;
+			for (j = 0; j < i; j++) {
+				if (input_conn[j] == input_conn[i]) {
+					dup_found = true;
+					break;
+				}
+			}
+			if (dup_found)
+				continue;
 			err = cx_auto_add_capture_volume(codec, nid,
 							 label, " Capture", cidx);
 		}
@@ -4378,6 +4371,53 @@
 	.reboot_notify = snd_hda_shutup_pins,
 };
 
+/*
+ * pin fix-up
+ */
+struct cxt_pincfg {
+	hda_nid_t nid;
+	u32 val;
+};
+
+static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg)
+{
+	for (; cfg->nid; cfg++)
+		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+
+}
+
+static void apply_pin_fixup(struct hda_codec *codec,
+			    const struct snd_pci_quirk *quirk,
+			    const struct cxt_pincfg **table)
+{
+	quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+	if (quirk) {
+		snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n",
+			    quirk->name);
+		apply_pincfg(codec, table[quirk->value]);
+	}
+}
+
+enum {
+	CXT_PINCFG_LENOVO_X200,
+};
+
+static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
+	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
+	{ 0x17, 0x21a11000 }, /* dock-mic */
+	{ 0x19, 0x2121103f }, /* dock-HP */
+	{}
+};
+
+static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
+	[CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
+};
+
+static const struct snd_pci_quirk cxt_fixups[] = {
+	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
+	{}
+};
+
 static int patch_conexant_auto(struct hda_codec *codec)
 {
 	struct conexant_spec *spec;
@@ -4391,6 +4431,15 @@
 		return -ENOMEM;
 	codec->spec = spec;
 	codec->pin_amp_workaround = 1;
+
+	switch (codec->vendor_id) {
+	case 0x14f15045:
+		spec->single_adc_amp = 1;
+		break;
+	}
+
+	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
+
 	err = cx_auto_search_adcs(codec);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 19cb72d..aac3bfa 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -324,6 +324,66 @@
 	return -EINVAL;
 }
 
+static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hdmi_spec *spec;
+	int pin_idx;
+
+	spec = codec->spec;
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+
+	pin_idx = kcontrol->private_value;
+	uinfo->count = spec->pins[pin_idx].sink_eld.eld_size;
+
+	return 0;
+}
+
+static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hdmi_spec *spec;
+	int pin_idx;
+
+	spec = codec->spec;
+	pin_idx = kcontrol->private_value;
+
+	memcpy(ucontrol->value.bytes.data,
+		spec->pins[pin_idx].sink_eld.eld_buffer, ELD_MAX_SIZE);
+
+	return 0;
+}
+
+static struct snd_kcontrol_new eld_bytes_ctl = {
+	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.name = "ELD",
+	.info = hdmi_eld_ctl_info,
+	.get = hdmi_eld_ctl_get,
+};
+
+static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx,
+			int device)
+{
+	struct snd_kcontrol *kctl;
+	struct hdmi_spec *spec = codec->spec;
+	int err;
+
+	kctl = snd_ctl_new1(&eld_bytes_ctl, codec);
+	if (!kctl)
+		return -ENOMEM;
+	kctl->private_value = pin_idx;
+	kctl->id.device = device;
+
+	err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
 #ifdef BE_PARANOID
 static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid,
 				int *packet_index, int *byte_index)
@@ -946,7 +1006,6 @@
 	unsigned int caps, config;
 	int pin_idx;
 	struct hdmi_spec_per_pin *per_pin;
-	struct hdmi_eld *eld;
 	int err;
 
 	caps = snd_hda_param_read(codec, pin_nid, AC_PAR_PIN_CAP);
@@ -963,23 +1022,15 @@
 
 	pin_idx = spec->num_pins;
 	per_pin = &spec->pins[pin_idx];
-	eld = &per_pin->sink_eld;
 
 	per_pin->pin_nid = pin_nid;
 
-	err = snd_hda_input_jack_add(codec, pin_nid,
-				     SND_JACK_VIDEOOUT, NULL);
-	if (err < 0)
-		return err;
-
 	err = hdmi_read_pin_conn(codec, pin_idx);
 	if (err < 0)
 		return err;
 
 	spec->num_pins++;
 
-	hdmi_present_sense(codec, pin_nid, eld);
-
 	return 0;
 }
 
@@ -1162,6 +1213,25 @@
 	return 0;
 }
 
+static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx)
+{
+	int err;
+	char hdmi_str[32];
+	struct hdmi_spec *spec = codec->spec;
+	struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+	int pcmdev = spec->pcm_rec[pin_idx].device;
+
+	snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP,pcm=%d", pcmdev);
+
+	err = snd_hda_input_jack_add(codec, per_pin->pin_nid,
+			     SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL);
+	if (err < 0)
+		return err;
+
+	hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld);
+	return 0;
+}
+
 static int generic_hdmi_build_controls(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -1170,12 +1240,25 @@
 
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+
+		err = generic_hdmi_build_jack(codec, pin_idx);
+		if (err < 0)
+			return err;
+
 		err = snd_hda_create_spdif_out_ctls(codec,
 						    per_pin->pin_nid,
 						    per_pin->mux_nids[0]);
 		if (err < 0)
 			return err;
 		snd_hda_spdif_ctls_unassign(codec, pin_idx);
+
+		/* add control for ELD Bytes */
+		err = hdmi_create_eld_ctl(codec,
+					pin_idx,
+					spec->pcm_rec[pin_idx].device);
+
+		if (err < 0)
+			return err;
 	}
 
 	return 0;
@@ -1491,7 +1574,7 @@
 				     struct snd_pcm_substream *substream)
 {
 	int chs;
-	unsigned int dataDCC1, dataDCC2, channel_id;
+	unsigned int dataDCC2, channel_id;
 	int i;
 	struct hdmi_spec *spec = codec->spec;
 	struct hda_spdif_out *spdif =
@@ -1501,7 +1584,6 @@
 
 	chs = substream->runtime->channels;
 
-	dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT;
 	dataDCC2 = 0x2;
 
 	/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60..80d6add 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -116,6 +116,8 @@
 	const hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 	hda_nid_t mixer_nid;		/* analog-mixer NID */
+	DECLARE_BITMAP(vol_ctls, 0x20 << 1);
+	DECLARE_BITMAP(sw_ctls, 0x20 << 1);
 
 	/* capture setup for dynamic dual-adc switch */
 	hda_nid_t cur_adc;
@@ -159,23 +161,27 @@
 	void (*power_hook)(struct hda_codec *codec);
 #endif
 	void (*shutup)(struct hda_codec *codec);
+	void (*automute_hook)(struct hda_codec *codec);
 
 	/* for pin sensing */
-	unsigned int jack_present: 1;
+	unsigned int hp_jack_present:1;
 	unsigned int line_jack_present:1;
 	unsigned int master_mute:1;
 	unsigned int auto_mic:1;
 	unsigned int auto_mic_valid_imux:1;	/* valid imux for auto-mic */
-	unsigned int automute:1;	/* HP automute enabled */
-	unsigned int detect_line:1;	/* Line-out detection enabled */
-	unsigned int automute_lines:1;	/* automute line-out as well */
-	unsigned int automute_hp_lo:1;	/* both HP and LO available */
+	unsigned int automute_speaker:1; /* automute speaker outputs */
+	unsigned int automute_lo:1; /* automute LO outputs */
+	unsigned int detect_hp:1;	/* Headphone detection enabled */
+	unsigned int detect_lo:1;	/* Line-out detection enabled */
+	unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */
+	unsigned int automute_lo_possible:1;	  /* there are line outs and HP */
 
 	/* other flags */
 	unsigned int no_analog :1; /* digital I/O only */
 	unsigned int dyn_adc_switch:1; /* switch ADCs (for ALC275) */
 	unsigned int single_input_src:1;
 	unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
+	unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
 
 	/* auto-mute control */
 	int automute_mode;
@@ -193,6 +199,7 @@
 	/* for PLL fix */
 	hda_nid_t pll_nid;
 	unsigned int pll_coef_idx, pll_coef_bit;
+	unsigned int coef0;
 
 	/* fix-up list */
 	int fixup_id;
@@ -202,6 +209,9 @@
 	/* multi-io */
 	int multi_ios;
 	struct alc_multi_io multi_io[4];
+
+	/* bind volumes */
+	struct snd_array bind_ctls;
 };
 
 #define ALC_MODEL_AUTO		0	/* common for all chips */
@@ -525,8 +535,8 @@
 	}
 }
 
-/* Toggle internal speakers muting */
-static void update_speakers(struct hda_codec *codec)
+/* Toggle outputs muting */
+static void update_outputs(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int on;
@@ -538,10 +548,10 @@
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 		    spec->autocfg.hp_pins, spec->master_mute, true);
 
-	if (!spec->automute)
+	if (!spec->automute_speaker)
 		on = 0;
 	else
-		on = spec->jack_present | spec->line_jack_present;
+		on = spec->hp_jack_present | spec->line_jack_present;
 	on |= spec->master_mute;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
 		    spec->autocfg.speaker_pins, on, false);
@@ -551,26 +561,35 @@
 	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
 	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
 		return;
-	if (!spec->automute_lines || !spec->automute)
+	if (!spec->automute_lo)
 		on = 0;
 	else
-		on = spec->jack_present;
+		on = spec->hp_jack_present;
 	on |= spec->master_mute;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 		    spec->autocfg.line_out_pins, on, false);
 }
 
+static void call_update_outputs(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	if (spec->automute_hook)
+		spec->automute_hook(codec);
+	else
+		update_outputs(codec);
+}
+
 /* standard HP-automute helper */
 static void alc_hp_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute)
-		return;
-	spec->jack_present =
+	spec->hp_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 			     spec->autocfg.hp_pins);
-	update_speakers(codec);
+	if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo))
+		return;
+	call_update_outputs(codec);
 }
 
 /* standard line-out-automute helper */
@@ -578,12 +597,16 @@
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute || !spec->detect_line)
+	/* check LO jack only when it's different from HP */
+	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
 		return;
+
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
-	update_speakers(codec);
+	if (!spec->automute_speaker || !spec->detect_lo)
+		return;
+	call_update_outputs(codec);
 }
 
 #define get_connection_index(codec, mux, nid) \
@@ -781,7 +804,7 @@
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
-	if (spec->automute_hp_lo) {
+	if (spec->automute_speaker_possible && spec->automute_lo_possible) {
 		uinfo->value.enumerated.items = 3;
 		texts = texts3;
 	} else {
@@ -800,13 +823,12 @@
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	unsigned int val;
-	if (!spec->automute)
-		val = 0;
-	else if (!spec->automute_lines)
-		val = 1;
-	else
-		val = 2;
+	unsigned int val = 0;
+	if (spec->automute_speaker)
+		val++;
+	if (spec->automute_lo)
+		val++;
+
 	ucontrol->value.enumerated.item[0] = val;
 	return 0;
 }
@@ -819,28 +841,36 @@
 
 	switch (ucontrol->value.enumerated.item[0]) {
 	case 0:
-		if (!spec->automute)
+		if (!spec->automute_speaker && !spec->automute_lo)
 			return 0;
-		spec->automute = 0;
+		spec->automute_speaker = 0;
+		spec->automute_lo = 0;
 		break;
 	case 1:
-		if (spec->automute && !spec->automute_lines)
-			return 0;
-		spec->automute = 1;
-		spec->automute_lines = 0;
+		if (spec->automute_speaker_possible) {
+			if (!spec->automute_lo && spec->automute_speaker)
+				return 0;
+			spec->automute_speaker = 1;
+			spec->automute_lo = 0;
+		} else if (spec->automute_lo_possible) {
+			if (spec->automute_lo)
+				return 0;
+			spec->automute_lo = 1;
+		} else
+			return -EINVAL;
 		break;
 	case 2:
-		if (!spec->automute_hp_lo)
+		if (!spec->automute_lo_possible || !spec->automute_speaker_possible)
 			return -EINVAL;
-		if (spec->automute && spec->automute_lines)
+		if (spec->automute_speaker && spec->automute_lo)
 			return 0;
-		spec->automute = 1;
-		spec->automute_lines = 1;
+		spec->automute_speaker = 1;
+		spec->automute_lo = 1;
 		break;
 	default:
 		return -EINVAL;
 	}
-	update_speakers(codec);
+	call_update_outputs(codec);
 	return 1;
 }
 
@@ -877,7 +907,7 @@
  * Check the availability of HP/line-out auto-mute;
  * Set up appropriately if really supported
  */
-static void alc_init_auto_hp(struct hda_codec *codec)
+static void alc_init_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
@@ -892,8 +922,6 @@
 		present++;
 	if (present < 2) /* need two different output types */
 		return;
-	if (present == 3)
-		spec->automute_hp_lo = 1; /* both HP and LO automute */
 
 	if (!cfg->speaker_pins[0] &&
 	    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
@@ -909,6 +937,8 @@
 		cfg->hp_outs = cfg->line_outs;
 	}
 
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+
 	for (i = 0; i < cfg->hp_outs; i++) {
 		hda_nid_t nid = cfg->hp_pins[i];
 		if (!is_jack_detectable(codec, nid))
@@ -918,28 +948,32 @@
 		snd_hda_codec_write_cache(codec, nid, 0,
 				  AC_VERB_SET_UNSOLICITED_ENABLE,
 				  AC_USRSP_EN | ALC_HP_EVENT);
-		spec->automute = 1;
-		spec->automute_mode = ALC_AUTOMUTE_PIN;
-	}
-	if (spec->automute && cfg->line_out_pins[0] &&
-	    cfg->speaker_pins[0] &&
-	    cfg->line_out_pins[0] != cfg->hp_pins[0] &&
-	    cfg->line_out_pins[0] != cfg->speaker_pins[0]) {
-		for (i = 0; i < cfg->line_outs; i++) {
-			hda_nid_t nid = cfg->line_out_pins[i];
-			if (!is_jack_detectable(codec, nid))
-				continue;
-			snd_printdd("realtek: Enable Line-Out auto-muting "
-				    "on NID 0x%x\n", nid);
-			snd_hda_codec_write_cache(codec, nid, 0,
-					AC_VERB_SET_UNSOLICITED_ENABLE,
-					AC_USRSP_EN | ALC_FRONT_EVENT);
-			spec->detect_line = 1;
-		}
-		spec->automute_lines = spec->detect_line;
+		spec->detect_hp = 1;
 	}
 
-	if (spec->automute) {
+	if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) {
+		if (cfg->speaker_outs)
+			for (i = 0; i < cfg->line_outs; i++) {
+				hda_nid_t nid = cfg->line_out_pins[i];
+				if (!is_jack_detectable(codec, nid))
+					continue;
+				snd_printdd("realtek: Enable Line-Out "
+					    "auto-muting on NID 0x%x\n", nid);
+				snd_hda_codec_write_cache(codec, nid, 0,
+						AC_VERB_SET_UNSOLICITED_ENABLE,
+						AC_USRSP_EN | ALC_FRONT_EVENT);
+				spec->detect_lo = 1;
+		}
+		spec->automute_lo_possible = spec->detect_hp;
+	}
+
+	spec->automute_speaker_possible = cfg->speaker_outs &&
+		(spec->detect_hp || spec->detect_lo);
+
+	spec->automute_lo = spec->automute_lo_possible;
+	spec->automute_speaker = spec->automute_speaker_possible;
+
+	if (spec->automute_speaker_possible || spec->automute_lo_possible) {
 		/* create a control for automute mode */
 		alc_add_automute_mode_enum(codec);
 		spec->unsol_event = alc_sku_unsol_event;
@@ -1140,7 +1174,7 @@
 /* check the availabilities of auto-mute and auto-mic switches */
 static void alc_auto_check_switches(struct hda_codec *codec)
 {
-	alc_init_auto_hp(codec);
+	alc_init_automute(codec);
 	alc_init_auto_mic(codec);
 }
 
@@ -1320,7 +1354,9 @@
 	 * 15   : 1 --> enable the function "Mute internal speaker
 	 *	        when the external headphone out jack is plugged"
 	 */
-	if (!spec->autocfg.hp_pins[0]) {
+	if (!spec->autocfg.hp_pins[0] &&
+	    !(spec->autocfg.line_out_pins[0] &&
+	      spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
 		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
@@ -1521,6 +1557,15 @@
 			    coef_val);
 }
 
+/* a special bypass for COEF 0; read the cached value at the second time */
+static unsigned int alc_get_coef0(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	if (!spec->coef0)
+		spec->coef0 = alc_read_coef_idx(codec, 0);
+	return spec->coef0;
+}
+
 /*
  * Digital I/O handling
  */
@@ -1559,27 +1604,29 @@
 static void alc_auto_parse_digital(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int i, err;
+	int i, err, nums;
 	hda_nid_t dig_nid;
 
 	/* support multiple SPDIFs; the secondary is set up as a slave */
+	nums = 0;
 	for (i = 0; i < spec->autocfg.dig_outs; i++) {
 		hda_nid_t conn[4];
 		err = snd_hda_get_connections(codec,
 					      spec->autocfg.dig_out_pins[i],
 					      conn, ARRAY_SIZE(conn));
-		if (err < 0)
+		if (err <= 0)
 			continue;
 		dig_nid = conn[0]; /* assume the first element is audio-out */
-		if (!i) {
+		if (!nums) {
 			spec->multiout.dig_out_nid = dig_nid;
 			spec->dig_out_type = spec->autocfg.dig_out_type[0];
 		} else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
-			spec->slave_dig_outs[i - 1] = dig_nid;
+			spec->slave_dig_outs[nums - 1] = dig_nid;
 		}
+		nums++;
 	}
 
 	if (spec->autocfg.dig_in_pin) {
@@ -1784,6 +1831,7 @@
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
+	"PCM Playback Volume",
 	NULL,
 };
 
@@ -1798,6 +1846,7 @@
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
 	"Line-Out Playback Switch",
+	"PCM Playback Switch",
 	NULL,
 };
 
@@ -2223,6 +2272,7 @@
 	struct alc_spec *spec = codec->spec;
 	struct hda_pcm *info = spec->pcm_rec;
 	const struct hda_pcm_stream *p;
+	bool have_multi_adcs;
 	int i;
 
 	codec->num_pcms = 1;
@@ -2301,8 +2351,11 @@
 	/* If the use of more than one ADC is requested for the current
 	 * model, configure a second analog capture-only PCM.
 	 */
+	have_multi_adcs = (spec->num_adc_nids > 1) &&
+		!spec->dyn_adc_switch && !spec->auto_mic &&
+		(!spec->input_mux || spec->input_mux->num_items > 1);
 	/* Additional Analaog capture for index #2 */
-	if (spec->alt_dac_nid || spec->num_adc_nids > 1) {
+	if (spec->alt_dac_nid || have_multi_adcs) {
 		codec->num_pcms = 3;
 		info = spec->pcm_rec + 2;
 		info->name = spec->stream_name_analog;
@@ -2318,7 +2371,7 @@
 				alc_pcm_null_stream;
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
 		}
-		if (spec->num_adc_nids > 1) {
+		if (have_multi_adcs) {
 			p = spec->stream_analog_alt_capture;
 			if (!p)
 				p = &alc_pcm_analog_alt_capture;
@@ -2359,6 +2412,18 @@
 	snd_array_free(&spec->kctls);
 }
 
+static void alc_free_bind_ctls(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	if (spec->bind_ctls.list) {
+		struct hda_bind_ctls **ctl = spec->bind_ctls.list;
+		int i;
+		for (i = 0; i < spec->bind_ctls.used; i++)
+			kfree(ctl[i]);
+	}
+	snd_array_free(&spec->bind_ctls);
+}
+
 static void alc_free(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -2369,6 +2434,7 @@
 	alc_shutup(codec);
 	snd_hda_input_jack_free(codec);
 	alc_free_kctls(codec);
+	alc_free_bind_ctls(codec);
 	kfree(spec);
 	snd_hda_detach_beep_device(codec);
 }
@@ -2432,6 +2498,47 @@
 }
 
 /*
+ * Rename codecs appropriately from COEF value
+ */
+struct alc_codec_rename_table {
+	unsigned int vendor_id;
+	unsigned short coef_mask;
+	unsigned short coef_bits;
+	const char *name;
+};
+
+static struct alc_codec_rename_table rename_tbl[] = {
+	{ 0x10ec0269, 0xfff0, 0x3010, "ALC277" },
+	{ 0x10ec0269, 0xf0f0, 0x2010, "ALC259" },
+	{ 0x10ec0269, 0xf0f0, 0x3010, "ALC258" },
+	{ 0x10ec0269, 0x00f0, 0x0010, "ALC269VB" },
+	{ 0x10ec0269, 0xffff, 0xa023, "ALC259" },
+	{ 0x10ec0269, 0xffff, 0x6023, "ALC281X" },
+	{ 0x10ec0269, 0x00f0, 0x0020, "ALC269VC" },
+	{ 0x10ec0887, 0x00f0, 0x0030, "ALC887-VD" },
+	{ 0x10ec0888, 0x00f0, 0x0030, "ALC888-VD" },
+	{ 0x10ec0888, 0xf0f0, 0x3020, "ALC886" },
+	{ 0x10ec0899, 0x2000, 0x2000, "ALC899" },
+	{ 0x10ec0892, 0xffff, 0x8020, "ALC661" },
+	{ 0x10ec0892, 0xffff, 0x8011, "ALC661" },
+	{ 0x10ec0892, 0xffff, 0x4011, "ALC656" },
+	{ } /* terminator */
+};
+
+static int alc_codec_rename_from_preset(struct hda_codec *codec)
+{
+	const struct alc_codec_rename_table *p;
+
+	for (p = rename_tbl; p->vendor_id; p++) {
+		if (p->vendor_id != codec->vendor_id)
+			continue;
+		if ((alc_get_coef0(codec) & p->coef_mask) == p->coef_bits)
+			return alc_codec_rename(codec, p->name);
+	}
+	return 0;
+}
+
+/*
  * Automatic parse of I/O pins from the BIOS configuration
  */
 
@@ -2439,11 +2546,15 @@
 	ALC_CTL_WIDGET_VOL,
 	ALC_CTL_WIDGET_MUTE,
 	ALC_CTL_BIND_MUTE,
+	ALC_CTL_BIND_VOL,
+	ALC_CTL_BIND_SW,
 };
 static const struct snd_kcontrol_new alc_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
 	HDA_BIND_MUTE(NULL, 0, 0, 0),
+	HDA_BIND_VOL(NULL, 0),
+	HDA_BIND_SW(NULL, 0),
 };
 
 /* add dynamic controls */
@@ -2484,13 +2595,14 @@
 #define __add_pb_sw_ctrl(spec, type, pfx, cidx, val)			\
 	add_control_with_pfx(spec, type, pfx, "Playback", "Switch", cidx, val)
 
+static const char * const channel_name[4] = {
+	"Front", "Surround", "CLFE", "Side"
+};
+
 static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
 					bool can_be_master, int *index)
 {
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	static const char * const chname[4] = {
-		"Front", "Surround", NULL /*CLFE*/, "Side"
-	};
 
 	*index = 0;
 	if (cfg->line_outs == 1 && !spec->multi_ios &&
@@ -2513,7 +2625,10 @@
 			return "PCM";
 		break;
 	}
-	return chname[ch];
+	if (snd_BUG_ON(ch >= ARRAY_SIZE(channel_name)))
+		return "PCM";
+
+	return channel_name[ch];
 }
 
 /* create input playback/capture controls for the given pin */
@@ -2548,7 +2663,6 @@
 	hda_nid_t *adc_nids = spec->private_adc_nids;
 	hda_nid_t *cap_nids = spec->private_capsrc_nids;
 	int max_nums = ARRAY_SIZE(spec->private_adc_nids);
-	bool indep_capsrc = false;
 	int i, nums = 0;
 
 	nid = codec->start_nid;
@@ -2570,13 +2684,11 @@
 				break;
 			if (type == AC_WID_AUD_SEL) {
 				cap_nids[nums] = src;
-				indep_capsrc = true;
 				break;
 			}
 			n = snd_hda_get_conn_list(codec, src, &list);
 			if (n > 1) {
 				cap_nids[nums] = src;
-				indep_capsrc = true;
 				break;
 			} else if (n != 1)
 				break;
@@ -2777,8 +2889,9 @@
 		if (found_in_nid_list(nid, spec->multiout.dac_nids,
 				      spec->multiout.num_dacs))
 			continue;
-		if (spec->multiout.hp_nid == nid)
-			continue;
+		if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
+				      ARRAY_SIZE(spec->multiout.hp_out_nid)))
+		    continue;
 		if (found_in_nid_list(nid, spec->multiout.extra_out_nid,
 				      ARRAY_SIZE(spec->multiout.extra_out_nid)))
 		    continue;
@@ -2795,6 +2908,29 @@
 	return 0;
 }
 
+static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
+				    const hda_nid_t *pins, hda_nid_t *dacs)
+{
+	int i;
+
+	if (num_outs && !dacs[0]) {
+		dacs[0] = alc_auto_look_for_dac(codec, pins[0]);
+		if (!dacs[0])
+			return 0;
+	}
+
+	for (i = 1; i < num_outs; i++)
+		dacs[i] = get_dac_if_single(codec, pins[i]);
+	for (i = 1; i < num_outs; i++) {
+		if (!dacs[i])
+			dacs[i] = alc_auto_look_for_dac(codec, pins[i]);
+	}
+	return 0;
+}
+
+static int alc_auto_fill_multi_ios(struct hda_codec *codec,
+				   unsigned int location);
+
 /* fill in the dac_nids table from the parsed pin configuration */
 static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 {
@@ -2806,7 +2942,7 @@
  again:
 	/* set num_dacs once to full for alc_auto_look_for_dac() */
 	spec->multiout.num_dacs = cfg->line_outs;
-	spec->multiout.hp_nid = 0;
+	spec->multiout.hp_out_nid[0] = 0;
 	spec->multiout.extra_out_nid[0] = 0;
 	memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
 	spec->multiout.dac_nids = spec->private_dac_nids;
@@ -2817,7 +2953,7 @@
 			spec->private_dac_nids[i] =
 				get_dac_if_single(codec, cfg->line_out_pins[i]);
 		if (cfg->hp_outs)
-			spec->multiout.hp_nid =
+			spec->multiout.hp_out_nid[0] =
 				get_dac_if_single(codec, cfg->hp_pins[0]);
 		if (cfg->speaker_outs)
 			spec->multiout.extra_out_nid[0] =
@@ -2849,24 +2985,58 @@
 				sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
 	}
 
-	if (cfg->hp_outs && !spec->multiout.hp_nid)
-		spec->multiout.hp_nid =
-			alc_auto_look_for_dac(codec, cfg->hp_pins[0]);
-	if (cfg->speaker_outs && !spec->multiout.extra_out_nid[0])
-		spec->multiout.extra_out_nid[0] =
-			alc_auto_look_for_dac(codec, cfg->speaker_pins[0]);
+	if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+		/* try to fill multi-io first */
+		unsigned int location, defcfg;
+		int num_pins;
+
+		defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
+		location = get_defcfg_location(defcfg);
+
+		num_pins = alc_auto_fill_multi_ios(codec, location);
+		if (num_pins > 0) {
+			spec->multi_ios = num_pins;
+			spec->ext_channel_count = 2;
+			spec->multiout.num_dacs = num_pins + 1;
+		}
+	}
+
+	if (cfg->line_out_type != AUTO_PIN_HP_OUT)
+		alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins,
+				 spec->multiout.hp_out_nid);
+	if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT)
+		alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins,
+				 spec->multiout.extra_out_nid);
 
 	return 0;
 }
 
+static inline unsigned int get_ctl_pos(unsigned int data)
+{
+	hda_nid_t nid = get_amp_nid_(data);
+	unsigned int dir = get_amp_direction_(data);
+	return (nid << 1) | dir;
+}
+
+#define is_ctl_used(bits, data) \
+	test_bit(get_ctl_pos(data), bits)
+#define mark_ctl_usage(bits, data) \
+	set_bit(get_ctl_pos(data), bits)
+
 static int alc_auto_add_vol_ctl(struct hda_codec *codec,
 			      const char *pfx, int cidx,
 			      hda_nid_t nid, unsigned int chs)
 {
+	struct alc_spec *spec = codec->spec;
+	unsigned int val;
 	if (!nid)
 		return 0;
+	val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT);
+	if (is_ctl_used(spec->vol_ctls, val) && chs != 2) /* exclude LFE */
+		return 0;
+	mark_ctl_usage(spec->vol_ctls, val);
 	return __add_pb_vol_ctrl(codec->spec, ALC_CTL_WIDGET_VOL, pfx, cidx,
-				 HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+				 val);
 }
 
 #define alc_auto_add_stereo_vol(codec, pfx, cidx, nid)	\
@@ -2879,6 +3049,7 @@
 			     const char *pfx, int cidx,
 			     hda_nid_t nid, unsigned int chs)
 {
+	struct alc_spec *spec = codec->spec;
 	int wid_type;
 	int type;
 	unsigned long val;
@@ -2895,6 +3066,9 @@
 		type = ALC_CTL_BIND_MUTE;
 		val = HDA_COMPOSE_AMP_VAL(nid, chs, 2, HDA_INPUT);
 	}
+	if (is_ctl_used(spec->sw_ctls, val) && chs != 2) /* exclude LFE */
+		return 0;
+	mark_ctl_usage(spec->sw_ctls, val);
 	return __add_pb_sw_ctrl(codec->spec, type, pfx, cidx, val);
 }
 
@@ -2955,7 +3129,7 @@
 		sw = alc_look_for_out_mute_nid(codec, pin, dac);
 		vol = alc_look_for_out_vol_nid(codec, pin, dac);
 		name = alc_get_line_out_pfx(spec, i, true, &index);
-		if (!name) {
+		if (!name || !strcmp(name, "CLFE")) {
 			/* Center/LFE */
 			err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1);
 			if (err < 0)
@@ -2981,23 +3155,24 @@
 	return 0;
 }
 
-/* add playback controls for speaker and HP outputs */
 static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
-					hda_nid_t dac, const char *pfx)
+				     hda_nid_t dac, const char *pfx)
 {
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t sw, vol;
 	int err;
 
-	if (!pin)
-		return 0;
 	if (!dac) {
+		unsigned int val;
 		/* the corresponding DAC is already occupied */
 		if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
 			return 0; /* no way */
 		/* create a switch only */
-		return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
-				   HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+		val = HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT);
+		if (is_ctl_used(spec->sw_ctls, val))
+			return 0; /* already created */
+		mark_ctl_usage(spec->sw_ctls, val);
+		return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val);
 	}
 
 	sw = alc_look_for_out_mute_nid(codec, pin, dac);
@@ -3011,20 +3186,112 @@
 	return 0;
 }
 
+static struct hda_bind_ctls *new_bind_ctl(struct hda_codec *codec,
+					  unsigned int nums,
+					  struct hda_ctl_ops *ops)
+{
+	struct alc_spec *spec = codec->spec;
+	struct hda_bind_ctls **ctlp, *ctl;
+	snd_array_init(&spec->bind_ctls, sizeof(ctl), 8);
+	ctlp = snd_array_new(&spec->bind_ctls);
+	if (!ctlp)
+		return NULL;
+	ctl = kzalloc(sizeof(*ctl) + sizeof(long) * (nums + 1), GFP_KERNEL);
+	*ctlp = ctl;
+	if (ctl)
+		ctl->ops = ops;
+	return ctl;
+}
+
+/* add playback controls for speaker and HP outputs */
+static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins,
+				      const hda_nid_t *pins,
+				      const hda_nid_t *dacs,
+				      const char *pfx)
+{
+	struct alc_spec *spec = codec->spec;
+	struct hda_bind_ctls *ctl;
+	char name[32];
+	int i, n, err;
+
+	if (!num_pins || !pins[0])
+		return 0;
+
+	if (num_pins == 1) {
+		hda_nid_t dac = *dacs;
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		return alc_auto_create_extra_out(codec, *pins, dac, pfx);
+	}
+
+	if (dacs[num_pins - 1]) {
+		/* OK, we have a multi-output system with individual volumes */
+		for (i = 0; i < num_pins; i++) {
+			snprintf(name, sizeof(name), "%s %s",
+				 pfx, channel_name[i]);
+			err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+							name);
+			if (err < 0)
+				return err;
+		}
+		return 0;
+	}
+
+	/* Let's create a bind-controls */
+	ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_sw);
+	if (!ctl)
+		return -ENOMEM;
+	n = 0;
+	for (i = 0; i < num_pins; i++) {
+		if (get_wcaps(codec, pins[i]) & AC_WCAP_OUT_AMP)
+			ctl->values[n++] =
+				HDA_COMPOSE_AMP_VAL(pins[i], 3, 0, HDA_OUTPUT);
+	}
+	if (n) {
+		snprintf(name, sizeof(name), "%s Playback Switch", pfx);
+		err = add_control(spec, ALC_CTL_BIND_SW, name, 0, (long)ctl);
+		if (err < 0)
+			return err;
+	}
+
+	ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_vol);
+	if (!ctl)
+		return -ENOMEM;
+	n = 0;
+	for (i = 0; i < num_pins; i++) {
+		hda_nid_t vol;
+		if (!pins[i] || !dacs[i])
+			continue;
+		vol = alc_look_for_out_vol_nid(codec, pins[i], dacs[i]);
+		if (vol)
+			ctl->values[n++] =
+				HDA_COMPOSE_AMP_VAL(vol, 3, 0, HDA_OUTPUT);
+	}
+	if (n) {
+		snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+		err = add_control(spec, ALC_CTL_BIND_VOL, name, 0, (long)ctl);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
 static int alc_auto_create_hp_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	return alc_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
-					 spec->multiout.hp_nid,
-					 "Headphone");
+	return alc_auto_create_extra_outs(codec, spec->autocfg.hp_outs,
+					  spec->autocfg.hp_pins,
+					  spec->multiout.hp_out_nid,
+					  "Headphone");
 }
 
 static int alc_auto_create_speaker_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	return alc_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0],
-					 spec->multiout.extra_out_nid[0],
-					 "Speaker");
+	return alc_auto_create_extra_outs(codec, spec->autocfg.speaker_outs,
+					  spec->autocfg.speaker_pins,
+					  spec->multiout.extra_out_nid,
+					  "Speaker");
 }
 
 static void alc_auto_set_output_and_unmute(struct hda_codec *codec,
@@ -3081,16 +3348,39 @@
 static void alc_auto_init_extra_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
+	int i;
+	hda_nid_t pin, dac;
 
-	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
-						  spec->multiout.hp_nid);
-	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
-					spec->multiout.extra_out_nid[0]);
+	for (i = 0; i < spec->autocfg.hp_outs; i++) {
+		if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+			break;
+		pin = spec->autocfg.hp_pins[i];
+		if (!pin)
+			break;
+		dac = spec->multiout.hp_out_nid[i];
+		if (!dac) {
+			if (i > 0 && spec->multiout.hp_out_nid[0])
+				dac = spec->multiout.hp_out_nid[0];
+			else
+				dac = spec->multiout.dac_nids[0];
+		}
+		alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+	}
+	for (i = 0; i < spec->autocfg.speaker_outs; i++) {
+		if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+			break;
+		pin = spec->autocfg.speaker_pins[i];
+		if (!pin)
+			break;
+		dac = spec->multiout.extra_out_nid[i];
+		if (!dac) {
+			if (i > 0 && spec->multiout.extra_out_nid[0])
+				dac = spec->multiout.extra_out_nid[0];
+			else
+				dac = spec->multiout.dac_nids[0];
+		}
+		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+	}
 }
 
 /*
@@ -3101,6 +3391,7 @@
 {
 	struct alc_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t prime_dac = spec->private_dac_nids[0];
 	int type, i, num_pins = 0;
 
 	for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) {
@@ -3128,8 +3419,13 @@
 		}
 	}
 	spec->multiout.num_dacs = 1;
-	if (num_pins < 2)
+	if (num_pins < 2) {
+		/* clear up again */
+		memset(spec->private_dac_nids, 0,
+		       sizeof(spec->private_dac_nids));
+		spec->private_dac_nids[0] = prime_dac;
 		return 0;
+	}
 	return num_pins;
 }
 
@@ -3215,36 +3511,11 @@
 	.put = alc_auto_ch_mode_put,
 };
 
-static int alc_auto_add_multi_channel_mode(struct hda_codec *codec,
-					   int (*fill_dac)(struct hda_codec *))
+static int alc_auto_add_multi_channel_mode(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	unsigned int location, defcfg;
-	int num_pins;
 
-	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) {
-		/* use HP as primary out */
-		cfg->speaker_outs = cfg->line_outs;
-		memcpy(cfg->speaker_pins, cfg->line_out_pins,
-		       sizeof(cfg->speaker_pins));
-		cfg->line_outs = cfg->hp_outs;
-		memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins));
-		cfg->hp_outs = 0;
-		memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
-		cfg->line_out_type = AUTO_PIN_HP_OUT;
-		if (fill_dac)
-			fill_dac(codec);
-	}
-	if (cfg->line_outs != 1 ||
-	    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
-		return 0;
-
-	defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
-	location = get_defcfg_location(defcfg);
-
-	num_pins = alc_auto_fill_multi_ios(codec, location);
-	if (num_pins > 0) {
+	if (spec->multi_ios > 0) {
 		struct snd_kcontrol_new *knew;
 
 		knew = alc_kcontrol_new(spec);
@@ -3254,10 +3525,6 @@
 		knew->name = kstrdup("Channel Mode", GFP_KERNEL);
 		if (!knew->name)
 			return -ENOMEM;
-
-		spec->multi_ios = num_pins;
-		spec->ext_channel_count = 2;
-		spec->multiout.num_dacs = num_pins + 1;
 	}
 	return 0;
 }
@@ -3540,27 +3807,42 @@
 				 const hda_nid_t *ssid_nids)
 {
 	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
 	int err;
 
-	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
-					   ignore_nids);
+	err = snd_hda_parse_pin_defcfg(codec, cfg, ignore_nids,
+				       spec->parse_flags);
 	if (err < 0)
 		return err;
-	if (!spec->autocfg.line_outs) {
-		if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+	if (!cfg->line_outs) {
+		if (cfg->dig_outs || cfg->dig_in_pin) {
 			spec->multiout.max_channels = 2;
 			spec->no_analog = 1;
 			goto dig_only;
 		}
 		return 0; /* can't find valid BIOS pin config */
 	}
+
+	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
+	    cfg->line_outs <= cfg->hp_outs) {
+		/* use HP as primary out */
+		cfg->speaker_outs = cfg->line_outs;
+		memcpy(cfg->speaker_pins, cfg->line_out_pins,
+		       sizeof(cfg->speaker_pins));
+		cfg->line_outs = cfg->hp_outs;
+		memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins));
+		cfg->hp_outs = 0;
+		memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+		cfg->line_out_type = AUTO_PIN_HP_OUT;
+	}
+
 	err = alc_auto_fill_dac_nids(codec);
 	if (err < 0)
 		return err;
-	err = alc_auto_add_multi_channel_mode(codec, alc_auto_fill_dac_nids);
+	err = alc_auto_add_multi_channel_mode(codec);
 	if (err < 0)
 		return err;
-	err = alc_auto_create_multi_out_ctls(codec, &spec->autocfg);
+	err = alc_auto_create_multi_out_ctls(codec, cfg);
 	if (err < 0)
 		return err;
 	err = alc_auto_create_hp_out(codec);
@@ -3663,10 +3945,8 @@
 	if (board_config == ALC_MODEL_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc880_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
 		else if (!err) {
 			printk(KERN_INFO
@@ -3691,10 +3971,8 @@
 
 	if (!spec->no_analog) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 	}
 
@@ -3709,6 +3987,10 @@
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 
@@ -3790,10 +4072,8 @@
 	if (board_config == ALC_MODEL_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc260_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
 		else if (!err) {
 			printk(KERN_INFO
@@ -3818,10 +4098,8 @@
 
 	if (!spec->no_analog) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
 	}
 
@@ -3839,6 +4117,10 @@
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 
@@ -3865,6 +4147,7 @@
 	PINFIX_LENOVO_Y530,
 	PINFIX_PB_M5210,
 	PINFIX_ACER_ASPIRE_7736,
+	PINFIX_ASUS_W90V,
 };
 
 static const struct alc_fixup alc882_fixups[] = {
@@ -3896,10 +4179,18 @@
 		.type = ALC_FIXUP_SKU,
 		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
+	[PINFIX_ASUS_W90V] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x16, 0x99130110 }, /* fix sequence for CLFE */
+			{ }
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
+	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
@@ -3946,6 +4237,10 @@
 		break;
 	}
 
+	err = alc_codec_rename_from_preset(codec);
+	if (err < 0)
+		goto error;
+
 	board_config = alc_board_config(codec, ALC882_MODEL_LAST,
 					alc882_models, alc882_cfg_tbl);
 
@@ -3969,10 +4264,8 @@
 	if (board_config == ALC_MODEL_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc882_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
 		else if (!err) {
 			printk(KERN_INFO
@@ -3997,10 +4290,8 @@
 
 	if (!spec->no_analog && has_cdefine_beep(codec)) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 	}
 
@@ -4019,6 +4310,10 @@
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 
@@ -4123,10 +4418,8 @@
 	if (board_config == ALC_MODEL_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc262_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
 		else if (!err) {
 			printk(KERN_INFO
@@ -4151,10 +4444,8 @@
 
 	if (!spec->no_analog && has_cdefine_beep(codec)) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 	}
 
@@ -4174,6 +4465,10 @@
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 /*
@@ -4222,14 +4517,9 @@
 
 /*
  */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc268_quirks.c"
-#endif
-
 static int patch_alc268(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config;
 	int i, has_beep, err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -4240,38 +4530,10 @@
 
 	/* ALC268 has no aa-loopback mixer */
 
-	board_config = alc_board_config(codec, ALC268_MODEL_LAST,
-					alc268_models, alc268_cfg_tbl);
-
-	if (board_config < 0)
-		board_config = alc_board_codec_sid_config(codec,
-			ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
-
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = ALC_MODEL_AUTO;
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		/* automatic parse from the BIOS config */
-		err = alc268_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-			board_config = ALC268_3ST;
-		}
-#endif
-	}
-
-	if (board_config != ALC_MODEL_AUTO)
-		setup_preset(codec, &alc268_presets[board_config]);
+	/* automatic parse from the BIOS config */
+	err = alc268_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
 
 	has_beep = 0;
 	for (i = 0; i < spec->num_mixers; i++) {
@@ -4283,10 +4545,8 @@
 
 	if (has_beep) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
 			/* override the amp caps for beep generator */
 			snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
@@ -4308,13 +4568,16 @@
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC_MODEL_AUTO)
-		spec->init_hook = alc_auto_init_std;
+	spec->init_hook = alc_auto_init_std;
 	spec->shutup = alc_eapd_shutup;
 
 	alc_init_jacks(codec);
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 /*
@@ -4408,9 +4671,9 @@
 
 static void alc269_shutup(struct hda_codec *codec)
 {
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017)
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x017)
 		alc269_toggle_power_output(codec, 0);
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		alc269_toggle_power_output(codec, 0);
 		msleep(150);
 	}
@@ -4419,19 +4682,19 @@
 #ifdef CONFIG_PM
 static int alc269_resume(struct hda_codec *codec)
 {
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		alc269_toggle_power_output(codec, 0);
 		msleep(150);
 	}
 
 	codec->patch_ops.init(codec);
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
 		alc269_toggle_power_output(codec, 1);
 		msleep(200);
 	}
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018)
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x018)
 		alc269_toggle_power_output(codec, 1);
 
 	snd_hda_codec_resume_amp(codec);
@@ -4484,6 +4747,46 @@
 	spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
 }
 
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+				     const struct alc_fixup *fix, int action)
+{
+	int coef;
+
+	if (action != ALC_FIXUP_ACT_INIT)
+		return;
+	/* The digital-mic unit sends PDM (differential signal) instead of
+	 * the standard PCM, thus you can't record a valid mono stream as is.
+	 * Below is a workaround specific to ALC269 to control the dmic
+	 * signal source as mono.
+	 */
+	coef = alc_read_coef_idx(codec, 0x07);
+	alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
+static void alc269_quanta_automute(struct hda_codec *codec)
+{
+	update_outputs(codec);
+
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_COEF_INDEX, 0x0c);
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_PROC_COEF, 0x680);
+
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_COEF_INDEX, 0x0c);
+	snd_hda_codec_write(codec, 0x20, 0,
+			AC_VERB_SET_PROC_COEF, 0x480);
+}
+
+static void alc269_fixup_quanta_mute(struct hda_codec *codec,
+				     const struct alc_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action != ALC_FIXUP_ACT_PROBE)
+		return;
+	spec->automute_hook = alc269_quanta_automute;
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4797,13 @@
 	ALC275_FIXUP_SONY_HWEQ,
 	ALC271_FIXUP_DMIC,
 	ALC269_FIXUP_PCM_44K,
+	ALC269_FIXUP_STEREO_DMIC,
+	ALC269_FIXUP_QUANTA_MUTE,
+	ALC269_FIXUP_LIFEBOOK,
+	ALC269_FIXUP_AMIC,
+	ALC269_FIXUP_DMIC,
+	ALC269VB_FIXUP_AMIC,
+	ALC269VB_FIXUP_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -4556,23 +4866,144 @@
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
 	},
+	[ALC269_FIXUP_STEREO_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_stereo_dmic,
+	},
+	[ALC269_FIXUP_QUANTA_MUTE] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_quanta_mute,
+	},
+	[ALC269_FIXUP_LIFEBOOK] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x1a, 0x2101103f }, /* dock line-out */
+			{ 0x1b, 0x23a11040 }, /* dock mic-in */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_QUANTA_MUTE
+	},
+	[ALC269_FIXUP_AMIC] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x0121401f }, /* HP out */
+			{ 0x18, 0x01a19c20 }, /* mic */
+			{ 0x19, 0x99a3092f }, /* int-mic */
+			{ }
+		},
+	},
+	[ALC269_FIXUP_DMIC] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x12, 0x99a3092f }, /* int-mic */
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x0121401f }, /* HP out */
+			{ 0x18, 0x01a19c20 }, /* mic */
+			{ }
+		},
+	},
+	[ALC269VB_FIXUP_AMIC] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x18, 0x01a19c20 }, /* mic */
+			{ 0x19, 0x99a3092f }, /* int-mic */
+			{ 0x21, 0x0121401f }, /* HP out */
+			{ }
+		},
+	},
+	[ALC269_FIXUP_DMIC] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x12, 0x99a3092f }, /* int-mic */
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x18, 0x01a19c20 }, /* mic */
+			{ 0x21, 0x0121401f }, /* HP out */
+			{ }
+		},
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
+
+#if 1
+	/* Below is a quirk table taken from the old code.
+	 * Basically the device should work as is without the fixup table.
+	 * If BIOS doesn't give a proper info, enable the corresponding
+	 * fixup entry.
+	 */ 
+	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
+		      ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_AMIC),
+	SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_FIXUP_DMIC),
+#endif
+	{}
+};
+
+static const struct alc_model_fixup alc269_fixup_models[] = {
+	{.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"},
+	{.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"},
 	{}
 };
 
@@ -4581,23 +5012,23 @@
 {
 	int val;
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) {
+	if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
 		alc_write_coef_idx(codec, 0xf, 0x960b);
 		alc_write_coef_idx(codec, 0xe, 0x8817);
 	}
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) {
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x016) {
 		alc_write_coef_idx(codec, 0xf, 0x960b);
 		alc_write_coef_idx(codec, 0xe, 0x8814);
 	}
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
 		val = alc_read_coef_idx(codec, 0x04);
 		/* Power up output pin */
 		alc_write_coef_idx(codec, 0x04, val | (1<<11));
 	}
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+	if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		val = alc_read_coef_idx(codec, 0xd);
 		if ((val & 0x0c00) >> 10 != 0x1) {
 			/* Capless ramp up clock control */
@@ -4621,15 +5052,10 @@
 
 /*
  */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc269_quirks.c"
-#endif
-
 static int patch_alc269(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config, coef;
-	int err;
+	int err = 0;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4641,72 +5067,41 @@
 
 	alc_auto_parse_customize_define(codec);
 
+	err = alc_codec_rename_from_preset(codec);
+	if (err < 0)
+		goto error;
+
 	if (codec->vendor_id == 0x10ec0269) {
 		spec->codec_variant = ALC269_TYPE_ALC269VA;
-		coef = alc_read_coef_idx(codec, 0);
-		if ((coef & 0x00f0) == 0x0010) {
+		switch (alc_get_coef0(codec) & 0x00f0) {
+		case 0x0010:
 			if (codec->bus->pci->subsystem_vendor == 0x1025 &&
-			    spec->cdefine.platform_type == 1) {
-				alc_codec_rename(codec, "ALC271X");
-			} else if ((coef & 0xf000) == 0x2000) {
-				alc_codec_rename(codec, "ALC259");
-			} else if ((coef & 0xf000) == 0x3000) {
-				alc_codec_rename(codec, "ALC258");
-			} else if ((coef & 0xfff0) == 0x3010) {
-				alc_codec_rename(codec, "ALC277");
-			} else {
-				alc_codec_rename(codec, "ALC269VB");
-			}
+			    spec->cdefine.platform_type == 1)
+				err = alc_codec_rename(codec, "ALC271X");
 			spec->codec_variant = ALC269_TYPE_ALC269VB;
-		} else if ((coef & 0x00f0) == 0x0020) {
-			if (coef == 0xa023)
-				alc_codec_rename(codec, "ALC259");
-			else if (coef == 0x6023)
-				alc_codec_rename(codec, "ALC281X");
-			else if (codec->bus->pci->subsystem_vendor == 0x17aa &&
-				 codec->bus->pci->subsystem_device == 0x21f3)
-				alc_codec_rename(codec, "ALC3202");
-			else
-				alc_codec_rename(codec, "ALC269VC");
+			break;
+		case 0x0020:
+			if (codec->bus->pci->subsystem_vendor == 0x17aa &&
+			    codec->bus->pci->subsystem_device == 0x21f3)
+				err = alc_codec_rename(codec, "ALC3202");
 			spec->codec_variant = ALC269_TYPE_ALC269VC;
-		} else
+			break;
+		default:
 			alc_fix_pll_init(codec, 0x20, 0x04, 15);
+		}
+		if (err < 0)
+			goto error;
 		alc269_fill_coef(codec);
 	}
 
-	board_config = alc_board_config(codec, ALC269_MODEL_LAST,
-					alc269_models, alc269_cfg_tbl);
+	alc_pick_fixup(codec, alc269_fixup_models,
+		       alc269_fixup_tbl, alc269_fixups);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = ALC_MODEL_AUTO;
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups);
-		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		/* automatic parse from the BIOS config */
-		err = alc269_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-			board_config = ALC269_BASIC;
-		}
-#endif
-	}
-
-	if (board_config != ALC_MODEL_AUTO)
-		setup_preset(codec, &alc269_presets[board_config]);
+	/* automatic parse from the BIOS config */
+	err = alc269_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
 
 	if (!spec->no_analog && !spec->adc_nids) {
 		alc_auto_fill_adc_caps(codec);
@@ -4719,10 +5114,8 @@
 
 	if (!spec->no_analog && has_cdefine_beep(codec)) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 	}
 
@@ -4734,8 +5127,7 @@
 #ifdef CONFIG_PM
 	codec->patch_ops.resume = alc269_resume;
 #endif
-	if (board_config == ALC_MODEL_AUTO)
-		spec->init_hook = alc_auto_init_std;
+	spec->init_hook = alc_auto_init_std;
 	spec->shutup = alc269_shutup;
 
 	alc_init_jacks(codec);
@@ -4747,6 +5139,10 @@
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 /*
@@ -4794,14 +5190,9 @@
 
 /*
  */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc861_quirks.c"
-#endif
-
 static int patch_alc861(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config;
 	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -4812,39 +5203,13 @@
 
 	spec->mixer_nid = 0x15;
 
-        board_config = alc_board_config(codec, ALC861_MODEL_LAST,
-					alc861_models, alc861_cfg_tbl);
+	alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = ALC_MODEL_AUTO;
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
-		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		/* automatic parse from the BIOS config */
-		err = alc861_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-		   board_config = ALC861_3ST_DIG;
-		}
-#endif
-	}
-
-	if (board_config != ALC_MODEL_AUTO)
-		setup_preset(codec, &alc861_presets[board_config]);
+	/* automatic parse from the BIOS config */
+	err = alc861_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
 
 	if (!spec->no_analog && !spec->adc_nids) {
 		alc_auto_fill_adc_caps(codec);
@@ -4857,10 +5222,8 @@
 
 	if (!spec->no_analog) {
 		err = snd_hda_attach_beep_device(codec, 0x23);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
 	}
 
@@ -4869,18 +5232,18 @@
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC_MODEL_AUTO) {
-		spec->init_hook = alc_auto_init_std;
+	spec->init_hook = alc_auto_init_std;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-		spec->power_hook = alc_power_eapd;
-#endif
-	}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
+	spec->power_hook = alc_power_eapd;
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc861_loopbacks;
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 /*
@@ -4902,24 +5265,41 @@
 }
 
 enum {
-	ALC660VD_FIX_ASUS_GPIO1
+	ALC660VD_FIX_ASUS_GPIO1,
+	ALC861VD_FIX_DALLAS,
 };
 
-/* reset GPIO1 */
+/* exclude VREF80 */
+static void alc861vd_fixup_dallas(struct hda_codec *codec,
+				  const struct alc_fixup *fix, int action)
+{
+	if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+		snd_hda_override_pin_caps(codec, 0x18, 0x00001714);
+		snd_hda_override_pin_caps(codec, 0x19, 0x0000171c);
+	}
+}
+
 static const struct alc_fixup alc861vd_fixups[] = {
 	[ALC660VD_FIX_ASUS_GPIO1] = {
 		.type = ALC_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
+			/* reset GPIO1 */
 			{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
 			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
 			{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
 			{ }
 		}
 	},
+	[ALC861VD_FIX_DALLAS] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc861vd_fixup_dallas,
+	},
 };
 
 static const struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_FIX_DALLAS),
 	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
+	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_FIX_DALLAS),
 	{}
 };
 
@@ -4931,14 +5311,10 @@
 
 /*
  */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc861vd_quirks.c"
-#endif
-
 static int patch_alc861vd(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int err, board_config;
+	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4948,39 +5324,13 @@
 
 	spec->mixer_nid = 0x0b;
 
-	board_config = alc_board_config(codec, ALC861VD_MODEL_LAST,
-					alc861vd_models, alc861vd_cfg_tbl);
+	alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = ALC_MODEL_AUTO;
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
-		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		/* automatic parse from the BIOS config */
-		err = alc861vd_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-			board_config = ALC861VD_3ST;
-		}
-#endif
-	}
-
-	if (board_config != ALC_MODEL_AUTO)
-		setup_preset(codec, &alc861vd_presets[board_config]);
+	/* automatic parse from the BIOS config */
+	err = alc861vd_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
 
 	if (codec->vendor_id == 0x10ec0660) {
 		/* always turn on EAPD */
@@ -4998,10 +5348,8 @@
 
 	if (!spec->no_analog) {
 		err = snd_hda_attach_beep_device(codec, 0x23);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 	}
 
@@ -5011,8 +5359,7 @@
 
 	codec->patch_ops = alc_patch_ops;
 
-	if (board_config == ALC_MODEL_AUTO)
-		spec->init_hook = alc_auto_init_std;
+	spec->init_hook = alc_auto_init_std;
 	spec->shutup = alc_eapd_shutup;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
@@ -5020,6 +5367,10 @@
 #endif
 
 	return 0;
+
+ error:
+	alc_free(codec);
+	return err;
 }
 
 /*
@@ -5077,6 +5428,14 @@
 	ALC662_FIXUP_CZC_P10T,
 	ALC662_FIXUP_SKU_IGNORE,
 	ALC662_FIXUP_HP_RP5800,
+	ALC662_FIXUP_ASUS_MODE1,
+	ALC662_FIXUP_ASUS_MODE2,
+	ALC662_FIXUP_ASUS_MODE3,
+	ALC662_FIXUP_ASUS_MODE4,
+	ALC662_FIXUP_ASUS_MODE5,
+	ALC662_FIXUP_ASUS_MODE6,
+	ALC662_FIXUP_ASUS_MODE7,
+	ALC662_FIXUP_ASUS_MODE8,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -5118,37 +5477,204 @@
 		.chained = true,
 		.chain_id = ALC662_FIXUP_SKU_IGNORE
 	},
+	[ALC662_FIXUP_ASUS_MODE1] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x18, 0x01a19c20 }, /* mic */
+			{ 0x19, 0x99a3092f }, /* int-mic */
+			{ 0x21, 0x0121401f }, /* HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE2] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x18, 0x01a19820 }, /* mic */
+			{ 0x19, 0x99a3092f }, /* int-mic */
+			{ 0x1b, 0x0121401f }, /* HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE3] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x0121441f }, /* HP */
+			{ 0x18, 0x01a19840 }, /* mic */
+			{ 0x19, 0x99a3094f }, /* int-mic */
+			{ 0x21, 0x01211420 }, /* HP2 */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE4] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x16, 0x99130111 }, /* speaker */
+			{ 0x18, 0x01a19840 }, /* mic */
+			{ 0x19, 0x99a3094f }, /* int-mic */
+			{ 0x21, 0x0121441f }, /* HP */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE5] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x0121441f }, /* HP */
+			{ 0x16, 0x99130111 }, /* speaker */
+			{ 0x18, 0x01a19840 }, /* mic */
+			{ 0x19, 0x99a3094f }, /* int-mic */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE6] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x01211420 }, /* HP2 */
+			{ 0x18, 0x01a19840 }, /* mic */
+			{ 0x19, 0x99a3094f }, /* int-mic */
+			{ 0x1b, 0x0121441f }, /* HP */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE7] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x17, 0x99130111 }, /* speaker */
+			{ 0x18, 0x01a19840 }, /* mic */
+			{ 0x19, 0x99a3094f }, /* int-mic */
+			{ 0x1b, 0x01214020 }, /* HP */
+			{ 0x21, 0x0121401f }, /* HP */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
+	[ALC662_FIXUP_ASUS_MODE8] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x12, 0x99a30970 }, /* int-mic */
+			{ 0x15, 0x01214020 }, /* HP */
+			{ 0x17, 0x99130111 }, /* speaker */
+			{ 0x18, 0x01a19840 }, /* mic */
+			{ 0x21, 0x0121401f }, /* HP */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
+	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
+
+#if 0
+	/* Below is a quirk table taken from the old code.
+	 * Basically the device should work as is without the fixup table.
+	 * If BIOS doesn't give a proper info, enable the corresponding
+	 * fixup entry.
+	 */ 
+	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC662_FIXUP_ASUS_MODE7),
+	SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC662_FIXUP_ASUS_MODE7),
+	SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC662_FIXUP_ASUS_MODE8),
+	SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC662_FIXUP_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC662_FIXUP_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC662_FIXUP_ASUS_MODE5),
+	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC662_FIXUP_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC662_FIXUP_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC662_FIXUP_ASUS_MODE4),
+#endif
 	{}
 };
 
 static const struct alc_model_fixup alc662_fixup_models[] = {
 	{.id = ALC272_FIXUP_MARIO, .name = "mario"},
+	{.id = ALC662_FIXUP_ASUS_MODE1, .name = "asus-mode1"},
+	{.id = ALC662_FIXUP_ASUS_MODE2, .name = "asus-mode2"},
+	{.id = ALC662_FIXUP_ASUS_MODE3, .name = "asus-mode3"},
+	{.id = ALC662_FIXUP_ASUS_MODE4, .name = "asus-mode4"},
+	{.id = ALC662_FIXUP_ASUS_MODE5, .name = "asus-mode5"},
+	{.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"},
+	{.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
+	{.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
 	{}
 };
 
 
 /*
  */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc662_quirks.c"
-#endif
-
 static int patch_alc662(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int err, board_config;
-	int coef;
+	int err = 0;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
@@ -5158,50 +5684,31 @@
 
 	spec->mixer_nid = 0x0b;
 
+	/* handle multiple HPs as is */
+	spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+
 	alc_auto_parse_customize_define(codec);
 
 	alc_fix_pll_init(codec, 0x20, 0x04, 15);
 
-	coef = alc_read_coef_idx(codec, 0);
-	if (coef == 0x8020 || coef == 0x8011)
-		alc_codec_rename(codec, "ALC661");
-	else if (coef & (1 << 14) &&
-		codec->bus->pci->subsystem_vendor == 0x1025 &&
-		spec->cdefine.platform_type == 1)
-		alc_codec_rename(codec, "ALC272X");
-	else if (coef == 0x4011)
-		alc_codec_rename(codec, "ALC656");
+	err = alc_codec_rename_from_preset(codec);
+	if (err < 0)
+		goto error;
 
-	board_config = alc_board_config(codec, ALC662_MODEL_LAST,
-					alc662_models, alc662_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = ALC_MODEL_AUTO;
+	if ((alc_get_coef0(codec) & (1 << 14)) &&
+	    codec->bus->pci->subsystem_vendor == 0x1025 &&
+	    spec->cdefine.platform_type == 1) {
+		if (alc_codec_rename(codec, "ALC272X") < 0)
+			goto error;
 	}
 
-	if (board_config == ALC_MODEL_AUTO) {
-		alc_pick_fixup(codec, alc662_fixup_models,
-			       alc662_fixup_tbl, alc662_fixups);
-		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
-		/* automatic parse from the BIOS config */
-		err = alc662_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-			board_config = ALC662_3ST_2ch_DIG;
-		}
-#endif
-	}
-
-	if (board_config != ALC_MODEL_AUTO)
-		setup_preset(codec, &alc662_presets[board_config]);
+	alc_pick_fixup(codec, alc662_fixup_models,
+		       alc662_fixup_tbl, alc662_fixups);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	/* automatic parse from the BIOS config */
+	err = alc662_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
 
 	if (!spec->no_analog && !spec->adc_nids) {
 		alc_auto_fill_adc_caps(codec);
@@ -5214,10 +5721,8 @@
 
 	if (!spec->no_analog && has_cdefine_beep(codec)) {
 		err = snd_hda_attach_beep_device(codec, 0x1);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 		switch (codec->vendor_id) {
 		case 0x10ec0662:
 			set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -5237,8 +5742,7 @@
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC_MODEL_AUTO)
-		spec->init_hook = alc_auto_init_std;
+	spec->init_hook = alc_auto_init_std;
 	spec->shutup = alc_eapd_shutup;
 
 	alc_init_jacks(codec);
@@ -5249,32 +5753,10 @@
 #endif
 
 	return 0;
-}
 
-static int patch_alc888(struct hda_codec *codec)
-{
-	if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
-		kfree(codec->chip_name);
-		if (codec->vendor_id == 0x10ec0887)
-			codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
-		else
-			codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
-		if (!codec->chip_name) {
-			alc_free(codec);
-			return -ENOMEM;
-		}
-		return patch_alc662(codec);
-	}
-	return patch_alc882(codec);
-}
-
-static int patch_alc899(struct hda_codec *codec)
-{
-	if ((alc_read_coef_idx(codec, 0) & 0x2000) != 0x2000) {
-		kfree(codec->chip_name);
-		codec->chip_name = kstrdup("ALC898", GFP_KERNEL);
-	}
-	return patch_alc882(codec);
+ error:
+	alc_free(codec);
+	return err;
 }
 
 /*
@@ -5288,14 +5770,9 @@
 
 /*
  */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc680_quirks.c"
-#endif
-
 static int patch_alc680(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config;
 	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5306,43 +5783,11 @@
 
 	/* ALC680 has no aa-loopback mixer */
 
-	board_config = alc_board_config(codec, ALC680_MODEL_LAST,
-					alc680_models, alc680_cfg_tbl);
-
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = ALC_MODEL_AUTO;
-	}
-
-	if (board_config == ALC_MODEL_AUTO) {
-		/* automatic parse from the BIOS config */
-		err = alc680_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		}
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-			board_config = ALC680_BASE;
-		}
-#endif
-	}
-
-	if (board_config != ALC_MODEL_AUTO) {
-		setup_preset(codec, &alc680_presets[board_config]);
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-		spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
-#endif
-	}
-
-	if (!spec->no_analog && !spec->adc_nids) {
-		alc_auto_fill_adc_caps(codec);
-		alc_rebuild_imux_for_auto_mic(codec);
-		alc_remove_invalid_adc_nids(codec);
+	/* automatic parse from the BIOS config */
+	err = alc680_parse_auto_config(codec);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
 	}
 
 	if (!spec->no_analog && !spec->cap_mixer)
@@ -5351,8 +5796,7 @@
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC_MODEL_AUTO)
-		spec->init_hook = alc_auto_init_std;
+	spec->init_hook = alc_auto_init_std;
 
 	return 0;
 }
@@ -5380,6 +5824,8 @@
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
 	  .patch = patch_alc662 },
+	{ .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3",
+	  .patch = patch_alc662 },
 	{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
 	{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
 	{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
@@ -5392,13 +5838,13 @@
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
+	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
 	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
 	  .patch = patch_alc882 },
-	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
+	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
 	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
 	{ .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
-	{ .id = 0x10ec0899, .name = "ALC899", .patch = patch_alc899 },
+	{ .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b5..de4c360 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
 {
@@ -696,6 +697,7 @@
 
 	return 1;
 }
+#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
@@ -2970,8 +2972,9 @@
 static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
 	int j, conn_len;
-	hda_nid_t conn[HDA_MAX_CONNECTIONS];
+	hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac;
 	unsigned int wcaps, wtype;
 
 	conn_len = snd_hda_get_connections(codec, nid, conn,
@@ -2999,10 +3002,21 @@
 			return conn[j];
 		}
 	}
-	/* if all DACs are already assigned, connect to the primary DAC */
+
+	/* if all DACs are already assigned, connect to the primary DAC,
+	   unless we're assigning a secondary headphone */
+	fallback_dac = spec->multiout.dac_nids[0];
+	if (spec->multiout.hp_nid) {
+		for (j = 0; j < cfg->hp_outs; j++)
+			if (cfg->hp_pins[j] == nid) {
+				fallback_dac = spec->multiout.hp_nid;
+				break;
+			}
+	}
+
 	if (conn_len > 1) {
 		for (j = 0; j < conn_len; j++) {
-			if (conn[j] == spec->multiout.dac_nids[0]) {
+			if (conn[j] == fallback_dac) {
 				snd_hda_codec_write_cache(codec, nid, 0,
 						  AC_VERB_SET_CONNECT_SEL, j);
 				break;
@@ -3777,9 +3791,10 @@
 }
 
 
-static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
+static int stac92xx_parse_auto_config(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t dig_out = 0, dig_in = 0;
 	int hp_swap = 0;
 	int i, err;
 
@@ -3962,6 +3977,22 @@
 	if (spec->multiout.max_channels > 2)
 		spec->surr_switch = 1;
 
+	/* find digital out and in converters */
+	for (i = codec->start_nid; i < codec->start_nid + codec->num_nodes; i++) {
+		unsigned int wid_caps = get_wcaps(codec, i);
+		if (wid_caps & AC_WCAP_DIGITAL) {
+			switch (get_wcaps_type(wid_caps)) {
+			case AC_WID_AUD_OUT:
+				if (!dig_out)
+					dig_out = i;
+				break;
+			case AC_WID_AUD_IN:
+				if (!dig_in)
+					dig_in = i;
+				break;
+			}
+		}
+	}
 	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = dig_out;
 	if (dig_in && spec->autocfg.dig_in_pin)
@@ -4128,22 +4159,14 @@
 #ifdef CONFIG_SND_HDA_INPUT_JACK
 	int def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	int connectivity = get_defcfg_connect(def_conf);
-	char name[32];
-	int err;
 
 	if (connectivity && connectivity != AC_JACK_PORT_FIXED)
 		return 0;
 
-	snprintf(name, sizeof(name), "%s at %s %s Jack",
-		snd_hda_get_jack_type(def_conf),
-		snd_hda_get_jack_connectivity(def_conf),
-		snd_hda_get_jack_location(def_conf));
-
-	err = snd_hda_input_jack_add(codec, nid, type, name);
-	if (err < 0)
-		return err;
-#endif /* CONFIG_SND_HDA_INPUT_JACK */
+	return snd_hda_input_jack_add(codec, nid, type, NULL);
+#else
 	return 0;
+#endif /* CONFIG_SND_HDA_INPUT_JACK */
 }
 
 static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid,
@@ -5273,7 +5296,7 @@
 	spec->capvols = stac925x_capvols;
 	spec->capsws = stac925x_capsws;
 
-	err = stac92xx_parse_auto_config(codec, 0x8, 0x7);
+	err = stac92xx_parse_auto_config(codec);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -5414,7 +5437,7 @@
 	spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids);
 	spec->pwr_nids = stac92hd73xx_pwr_nids;
 
-	err = stac92xx_parse_auto_config(codec, 0x25, 0x27);
+	err = stac92xx_parse_auto_config(codec);
 
 	if (!err) {
 		if (spec->board_config < 0) {
@@ -5583,9 +5606,7 @@
 static int patch_stac92hd83xxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
-	hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
 	int err;
-	int num_dacs;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -5625,25 +5646,8 @@
 		stac92xx_set_config_regs(codec,
 				stac92hd83xxx_brd_tbl[spec->board_config]);
 
-	switch (codec->vendor_id) {
-	case 0x111d76d1:
-	case 0x111d76d9:
-	case 0x111d76e5:
-	case 0x111d7666:
-	case 0x111d7667:
-	case 0x111d7668:
-	case 0x111d7669:
-	case 0x111d76e3:
-	case 0x111d7604:
-	case 0x111d76d4:
-	case 0x111d7605:
-	case 0x111d76d5:
-	case 0x111d76e7:
-		if (spec->board_config == STAC_92HD83XXX_PWR_REF)
-			break;
+	if (spec->board_config != STAC_92HD83XXX_PWR_REF)
 		spec->num_pwrs = 0;
-		break;
-	}
 
 	codec->patch_ops = stac92xx_patch_ops;
 
@@ -5670,7 +5674,7 @@
 	}
 #endif	
 
-	err = stac92xx_parse_auto_config(codec, 0x1d, 0);
+	err = stac92xx_parse_auto_config(codec);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -5686,22 +5690,6 @@
 		return err;
 	}
 
-	/* docking output support */
-	num_dacs = snd_hda_get_connections(codec, 0xF,
-				conn, STAC92HD83_DAC_COUNT + 1) - 1;
-	/* skip non-DAC connections */
-	while (num_dacs >= 0 &&
-			(get_wcaps_type(get_wcaps(codec, conn[num_dacs]))
-					!= AC_WID_AUD_OUT))
-		num_dacs--;
-	/* set port E and F to select the last DAC */
-	if (num_dacs >= 0) {
-		snd_hda_codec_write_cache(codec, 0xE, 0,
-			AC_VERB_SET_CONNECT_SEL, num_dacs);
-		snd_hda_codec_write_cache(codec, 0xF, 0,
-			AC_VERB_SET_CONNECT_SEL, num_dacs);
-	}
-
 	codec->proc_widget_hook = stac92hd_proc_hook;
 
 	return 0;
@@ -6007,7 +5995,7 @@
 
 	spec->multiout.dac_nids = spec->dac_nids;
 
-	err = stac92xx_parse_auto_config(codec, 0x21, 0);
+	err = stac92xx_parse_auto_config(codec);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6116,7 +6104,7 @@
 
 	spec->multiout.dac_nids = spec->dac_nids;
 	
-	err = stac92xx_parse_auto_config(codec, 0x08, 0x09);
+	err = stac92xx_parse_auto_config(codec);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6241,7 +6229,7 @@
 	spec->aloopback_shift = 0;
 	spec->eapd_switch = 1;
 
-	err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
+	err = stac92xx_parse_auto_config(codec);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6366,7 +6354,7 @@
 		break;
 	}
 
-	err = stac92xx_parse_auto_config(codec, 0x1f, 0x20);
+	err = stac92xx_parse_auto_config(codec);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6471,7 +6459,7 @@
 	spec->capvols = stac9872_capvols;
 	spec->capsws = stac9872_capsws;
 
-	err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
+	err = stac92xx_parse_auto_config(codec);
 	if (err < 0) {
 		stac92xx_free(codec);
 		return -EINVAL;
@@ -6571,10 +6559,23 @@
 	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76e8, .name = "92HD66B1X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76e9, .name = "92HD66B2X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76ea, .name = "92HD66B3X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76eb, .name = "92HD66C1X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76ec, .name = "92HD66C2X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76ed, .name = "92HD66C3X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76ee, .name = "92HD66B1X3", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76ef, .name = "92HD66B2X3", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76f0, .name = "92HD66B3X3", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76f1, .name = "92HD66C1X3", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76f2, .name = "92HD66C2X3", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76f3, .name = "92HD66C3/65", .patch = patch_stac92hd83xxx},
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798..0b020a9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1506,39 +1506,49 @@
 	struct via_spec *spec = codec->spec;
 	struct hda_pcm *info = spec->pcm_rec;
 
-	codec->num_pcms = 1;
+	codec->num_pcms = 0;
 	codec->pcm_info = info;
 
-	snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog),
-		 "%s Analog", codec->chip_name);
-	info->name = spec->stream_name_analog;
+	if (spec->multiout.num_dacs || spec->num_adc_nids) {
+		snprintf(spec->stream_name_analog,
+			 sizeof(spec->stream_name_analog),
+			 "%s Analog", codec->chip_name);
+		info->name = spec->stream_name_analog;
 
-	if (!spec->stream_analog_playback)
-		spec->stream_analog_playback = &via_pcm_analog_playback;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
-		*spec->stream_analog_playback;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
-		spec->multiout.dac_nids[0];
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
-		spec->multiout.max_channels;
+		if (spec->multiout.num_dacs) {
+			if (!spec->stream_analog_playback)
+				spec->stream_analog_playback =
+					&via_pcm_analog_playback;
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+				*spec->stream_analog_playback;
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+				spec->multiout.dac_nids[0];
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+				spec->multiout.max_channels;
+		}
 
-	if (!spec->stream_analog_capture) {
-		if (spec->dyn_adc_switch)
-			spec->stream_analog_capture =
-				&via_pcm_dyn_adc_analog_capture;
-		else
-			spec->stream_analog_capture = &via_pcm_analog_capture;
-	}
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] =
-		*spec->stream_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-	if (!spec->dyn_adc_switch)
-		info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
-			spec->num_adc_nids;
-
-	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+		if (!spec->stream_analog_capture) {
+			if (spec->dyn_adc_switch)
+				spec->stream_analog_capture =
+					&via_pcm_dyn_adc_analog_capture;
+			else
+				spec->stream_analog_capture =
+					&via_pcm_analog_capture;
+		}
+		if (spec->num_adc_nids) {
+			info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+				*spec->stream_analog_capture;
+			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+				spec->adc_nids[0];
+			if (!spec->dyn_adc_switch)
+				info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
+					spec->num_adc_nids;
+		}
 		codec->num_pcms++;
 		info++;
+	}
+
+	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
 		snprintf(spec->stream_name_digital,
 			 sizeof(spec->stream_name_digital),
 			 "%s Digital", codec->chip_name);
@@ -1562,17 +1572,19 @@
 			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
 				spec->dig_in_nid;
 		}
+		codec->num_pcms++;
+		info++;
 	}
 
 	if (spec->hp_dac_nid) {
-		codec->num_pcms++;
-		info++;
 		snprintf(spec->stream_name_hp, sizeof(spec->stream_name_hp),
 			 "%s HP", codec->chip_name);
 		info->name = spec->stream_name_hp;
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = via_pcm_hp_playback;
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
 			spec->hp_dac_nid;
+		codec->num_pcms++;
+		info++;
 	}
 	return 0;
 }
@@ -2084,7 +2096,7 @@
 	struct via_spec *spec = codec->spec;
 	struct nid_path *path;
 	bool check_dac;
-	hda_nid_t pin, dac;
+	hda_nid_t pin, dac = 0;
 	int err;
 
 	pin = spec->autocfg.speaker_pins[0];
@@ -3688,13 +3700,8 @@
 static void set_widgets_power_state_vt1812(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
-	int imux_is_smixer =
-	snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
 	unsigned int parm;
 	unsigned int present;
-	/* MUX10 (1eh) = stereo mixer */
-	imux_is_smixer =
-	snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
 	/* inputs */
 	/* PW 5/6/7 (29h/2ah/2bh) */
 	parm = AC_PWRST_D3;
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 0ccc0eb..8531b98 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2748,8 +2748,9 @@
 	if (!c->no_mpu401) {
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
 			ICEREG(ice, MPU1_CTRL),
-			(c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED),
-			ice->irq, 0, &ice->rmidi[0]);
+			c->mpu401_1_info_flags |
+			MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+			-1, &ice->rmidi[0]);
 		if (err < 0) {
 			snd_card_free(card);
 			return err;
@@ -2764,8 +2765,9 @@
 			/*  2nd port used  */
 			err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712,
 				ICEREG(ice, MPU2_CTRL),
-				(c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED),
-				ice->irq, 0, &ice->rmidi[1]);
+				c->mpu401_2_info_flags |
+				MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+				-1, &ice->rmidi[1]);
 
 			if (err < 0) {
 				snd_card_free(card);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 6a5b387..45b2055 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -42,6 +42,12 @@
 #include <asm/pgtable.h>
 #include <asm/cacheflush.h>
 
+#ifdef CONFIG_KVM_GUEST
+#include <linux/kvm_para.h>
+#else
+#define kvm_para_available() (0)
+#endif
+
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
 MODULE_LICENSE("GPL");
@@ -77,6 +83,7 @@
 static int buggy_irq = -1; /* auto-check */
 static int xbox;
 static int spdif_aclink = -1;
+static int inside_vm = -1;
 
 module_param(index, int, 0444);
 MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard.");
@@ -94,6 +101,8 @@
 MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection.");
 module_param(spdif_aclink, int, 0444);
 MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link.");
+module_param(inside_vm, bool, 0444);
+MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization.");
 
 /* just for backward compatibility */
 static int enable;
@@ -400,6 +409,7 @@
 	unsigned buggy_irq: 1;		/* workaround for buggy mobos */
 	unsigned xbox: 1;		/* workaround for Xbox AC'97 detection */
 	unsigned buggy_semaphore: 1;	/* workaround for buggy codec semaphore */
+	unsigned inside_vm: 1;		/* enable VM optimization */
 
 	int spdif_idx;	/* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */
 	unsigned int sdm_saved;	/* SDM reg value */
@@ -1065,8 +1075,11 @@
 			udelay(10);
 			continue;
 		}
-		if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV) &&
-		    ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb))
+		if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV))
+			continue;
+		if (chip->inside_vm)
+			break;
+		if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb))
 			break;
 	} while (timeout--);
 	ptr = ichdev->last_pos;
@@ -2984,6 +2997,10 @@
 	if (xbox)
 		chip->xbox = 1;
 
+	chip->inside_vm = inside_vm;
+	if (inside_vm)
+		printk(KERN_INFO "intel8x0: enable KVM optimization\n");
+
 	if (pci->vendor == PCI_VENDOR_ID_INTEL &&
 	    pci->device == PCI_DEVICE_ID_INTEL_440MX)
 		chip->fix_nocache = 1; /* enable workaround */
@@ -3226,6 +3243,14 @@
 			buggy_irq = 0;
 	}
 
+	if (inside_vm < 0) {
+		/* detect KVM and Parallels virtual environments */
+		inside_vm = kvm_para_available();
+#if defined(__i386__) || defined(__x86_64__)
+		inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR);
+#endif
+	}
+
 	if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data,
 				       &chip)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 0378126..2fd4bf2 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2820,8 +2820,8 @@
 	/* TODO enable MIDI IRQ and I/O */
 	err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401,
 				  chip->iobase + MPU401_DATA_PORT,
-				  MPU401_INFO_INTEGRATED,
-				  chip->irq, 0, &chip->rmidi);
+				  MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+				  -1, &chip->rmidi);
 	if (err < 0)
 		printk(KERN_WARNING "maestro3: no MIDI support.\n");
 #endif
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 82311fc..53e5508 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -678,15 +678,15 @@
 		goto err_card;
 
 	if (chip->model.device_config & (MIDI_OUTPUT | MIDI_INPUT)) {
-		unsigned int info_flags = MPU401_INFO_INTEGRATED;
+		unsigned int info_flags =
+				MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK;
 		if (chip->model.device_config & MIDI_OUTPUT)
 			info_flags |= MPU401_INFO_OUTPUT;
 		if (chip->model.device_config & MIDI_INPUT)
 			info_flags |= MPU401_INFO_INPUT;
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
 					  chip->addr + OXYGEN_MPU401,
-					  info_flags, 0, 0,
-					  &chip->midi);
+					  info_flags, -1, &chip->midi);
 		if (err < 0)
 			goto err_card;
 	}
diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c
index 32d096c..8433aa7 100644
--- a/sound/pci/oxygen/xonar_pcm179x.c
+++ b/sound/pci/oxygen/xonar_pcm179x.c
@@ -1074,6 +1074,7 @@
 	.device_config = PLAYBACK_0_TO_I2S |
 			 PLAYBACK_1_TO_SPDIF |
 			 CAPTURE_0_FROM_I2S_2 |
+			 CAPTURE_1_FROM_SPDIF |
 			 AC97_FMIC_SWITCH,
 	.dac_channels_pcm = 2,
 	.dac_channels_mixer = 2,
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index e34ae14..88cc776 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2109,7 +2109,7 @@
 		val = mpu_port[dev];
 		pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val);
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE,
-					  val, 0, chip->irq, 0,
+					  val, MPU401_INFO_IRQ_HOOK, -1,
 					  &chip->rmidi);
 		if (err < 0)
 			snd_printk(KERN_WARNING
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 1c6d1e1..f742202 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -151,7 +151,7 @@
 #define HDSP_PROGRAM	        0x020
 #define HDSP_CONFIG_MODE_0	0x040
 #define HDSP_CONFIG_MODE_1	0x080
-#define HDSP_VERSION_BIT	0x100
+#define HDSP_VERSION_BIT	(0x100 | HDSP_S_LOAD)
 #define HDSP_BIGENDIAN_MODE     0x200
 #define HDSP_RD_MULTIPLE        0x400
 #define HDSP_9652_ENABLE_MIXER  0x800
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67c..15a6c3b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -520,16 +520,9 @@
 #define HDSPM_DMA_AREA_BYTES (HDSPM_MAX_CHANNELS * HDSPM_CHANNEL_BUFFER_BYTES)
 #define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
 
-/* revisions >= 230 indicate AES32 card */
-#define HDSPM_MADI_ANCIENT_REV	204
-#define HDSPM_MADI_OLD_REV	207
-#define HDSPM_MADI_REV		210
 #define HDSPM_RAYDAT_REV	211
 #define HDSPM_AIO_REV		212
 #define HDSPM_MADIFACE_REV	213
-#define HDSPM_AES_REV		240
-#define HDSPM_AES32_REV		234
-#define HDSPM_AES32_OLD_REV	233
 
 /* speed factor modes */
 #define HDSPM_SPEED_SINGLE 0
@@ -1241,10 +1234,30 @@
 	return rate;
 }
 
+/* return latency in samples per period */
+static int hdspm_get_latency(struct hdspm *hdspm)
+{
+	int n;
+
+	n = hdspm_decode_latency(hdspm->control_register);
+
+	/* Special case for new RME cards with 32 samples period size.
+	 * The three latency bits in the control register
+	 * (HDSP_LatencyMask) encode latency values of 64 samples as
+	 * 0, 128 samples as 1 ... 4096 samples as 6. For old cards, 7
+	 * denotes 8192 samples, but on new cards like RayDAT or AIO,
+	 * it corresponds to 32 samples.
+	 */
+	if ((7 == n) && (RayDAT == hdspm->io_type || AIO == hdspm->io_type))
+		n = -1;
+
+	return 1 << (n + 6);
+}
+
 /* Latency function */
 static inline void hdspm_compute_period_size(struct hdspm *hdspm)
 {
-	hdspm->period_bytes = 1 << ((hdspm_decode_latency(hdspm->control_register) + 8));
+	hdspm->period_bytes = 4 * hdspm_get_latency(hdspm);
 }
 
 
@@ -1303,12 +1316,27 @@
 
 	spin_lock_irq(&s->lock);
 
-	frames >>= 7;
-	n = 0;
-	while (frames) {
-		n++;
-		frames >>= 1;
+	if (32 == frames) {
+		/* Special case for new RME cards like RayDAT/AIO which
+		 * support period sizes of 32 samples. Since latency is
+		 * encoded in the three bits of HDSP_LatencyMask, we can only
+		 * have values from 0 .. 7. While 0 still means 64 samples and
+		 * 6 represents 4096 samples on all cards, 7 represents 8192
+		 * on older cards and 32 samples on new cards.
+		 *
+		 * In other words, period size in samples is calculated by
+		 * 2^(n+6) with n ranging from 0 .. 7.
+		 */
+		n = 7;
+	} else {
+		frames >>= 7;
+		n = 0;
+		while (frames) {
+			n++;
+			frames >>= 1;
+		}
 	}
+
 	s->control_register &= ~HDSPM_LatencyMask;
 	s->control_register |= hdspm_encode_latency(n);
 
@@ -1339,6 +1367,10 @@
 		break;
 	case MADIface:
 		freq_const = 131072000000000ULL;
+		break;
+	default:
+		snd_BUG();
+		return 0;
 	}
 
 	return div_u64(freq_const, period);
@@ -1356,16 +1388,19 @@
 
 	switch (hdspm->io_type) {
 	case MADIface:
-	  n = 131072000000000ULL;  /* 125 MHz */
-	  break;
+		n = 131072000000000ULL;  /* 125 MHz */
+		break;
 	case MADI:
 	case AES32:
-	  n = 110069313433624ULL;  /* 105 MHz */
-	  break;
+		n = 110069313433624ULL;  /* 105 MHz */
+		break;
 	case RayDAT:
 	case AIO:
-	  n = 104857600000000ULL;  /* 100 MHz */
-	  break;
+		n = 104857600000000ULL;  /* 100 MHz */
+		break;
+	default:
+		snd_BUG();
+		return;
 	}
 
 	n = div_u64(n, rate);
@@ -4794,8 +4829,7 @@
 
 	snd_iprintf(buffer, "--- Settings ---\n");
 
-	x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
-							HDSPM_LatencyMask));
+	x = hdspm_get_latency(hdspm);
 
 	snd_iprintf(buffer,
 		"Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -4958,8 +4992,7 @@
 
 	snd_iprintf(buffer, "--- Settings ---\n");
 
-	x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
-				HDSPM_LatencyMask));
+	x = hdspm_get_latency(hdspm);
 
 	snd_iprintf(buffer,
 		    "Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -5665,19 +5698,6 @@
 	return 0;
 }
 
-static unsigned int period_sizes_old[] = {
-	64, 128, 256, 512, 1024, 2048, 4096
-};
-
-static unsigned int period_sizes_new[] = {
-	32, 64, 128, 256, 512, 1024, 2048, 4096
-};
-
-/* RayDAT and AIO always have a buffer of 16384 samples per channel */
-static unsigned int raydat_aio_buffer_sizes[] = {
-	16384
-};
-
 static struct snd_pcm_hardware snd_hdspm_playback_subinfo = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		 SNDRV_PCM_INFO_MMAP_VALID |
@@ -5696,8 +5716,8 @@
 	.channels_max = HDSPM_MAX_CHANNELS,
 	.buffer_bytes_max =
 	    HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
-	.period_bytes_min = (64 * 4),
-	.period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS,
+	.period_bytes_min = (32 * 4),
+	.period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
 	.periods_min = 2,
 	.periods_max = 512,
 	.fifo_size = 0
@@ -5721,31 +5741,13 @@
 	.channels_max = HDSPM_MAX_CHANNELS,
 	.buffer_bytes_max =
 	    HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
-	.period_bytes_min = (64 * 4),
-	.period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS,
+	.period_bytes_min = (32 * 4),
+	.period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
 	.periods_min = 2,
 	.periods_max = 512,
 	.fifo_size = 0
 };
 
-static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_old = {
-	.count = ARRAY_SIZE(period_sizes_old),
-	.list = period_sizes_old,
-	.mask = 0
-};
-
-static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_new = {
-	.count = ARRAY_SIZE(period_sizes_new),
-	.list = period_sizes_new,
-	.mask = 0
-};
-
-static struct snd_pcm_hw_constraint_list hw_constraints_raydat_io_buffer = {
-	.count = ARRAY_SIZE(raydat_aio_buffer_sizes),
-	.list = raydat_aio_buffer_sizes,
-	.mask = 0
-};
-
 static int snd_hdspm_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params,
 					   struct snd_pcm_hw_rule *rule)
 {
@@ -5946,26 +5948,29 @@
 	spin_unlock_irq(&hdspm->lock);
 
 	snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+	snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
 
 	switch (hdspm->io_type) {
 	case AIO:
 	case RayDAT:
-		snd_pcm_hw_constraint_list(runtime, 0,
-				SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-				&hw_constraints_period_sizes_new);
-		snd_pcm_hw_constraint_list(runtime, 0,
-				SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
-				&hw_constraints_raydat_io_buffer);
-
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+					     32, 4096);
+		/* RayDAT & AIO have a fixed buffer of 16384 samples per channel */
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+					     16384, 16384);
 		break;
 
 	default:
-		snd_pcm_hw_constraint_list(runtime, 0,
-				SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-				&hw_constraints_period_sizes_old);
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+					     64, 8192);
+		break;
 	}
 
 	if (AES32 == hdspm->io_type) {
+		runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
 		snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				&hdspm_hw_constraints_aes32_sample_rates);
 	} else {
@@ -6018,24 +6023,28 @@
 	spin_unlock_irq(&hdspm->lock);
 
 	snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+	snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+
 	switch (hdspm->io_type) {
 	case AIO:
 	case RayDAT:
-	  snd_pcm_hw_constraint_list(runtime, 0,
-				     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-				     &hw_constraints_period_sizes_new);
-	  snd_pcm_hw_constraint_list(runtime, 0,
-				     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
-				     &hw_constraints_raydat_io_buffer);
-	  break;
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+					     32, 4096);
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+					     16384, 16384);
+		break;
 
 	default:
-	  snd_pcm_hw_constraint_list(runtime, 0,
-				     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-				     &hw_constraints_period_sizes_old);
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+					     64, 8192);
+		break;
 	}
 
 	if (AES32 == hdspm->io_type) {
+		runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
 		snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				&hdspm_hw_constraints_aes32_sample_rates);
 	} else {
@@ -6081,7 +6090,7 @@
 }
 
 static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
-		unsigned int cmd, unsigned long __user arg)
+		unsigned int cmd, unsigned long arg)
 {
 	void __user *argp = (void __user *)arg;
 	struct hdspm *hdspm = hw->private_data;
@@ -6206,11 +6215,13 @@
 		info.line_out = hdspm_line_out(hdspm);
 		info.passthru = 0;
 		spin_unlock_irq(&hdspm->lock);
-		if (copy_to_user((void __user *) arg, &info, sizeof(info)))
+		if (copy_to_user(argp, &info, sizeof(info)))
 			return -EFAULT;
 		break;
 
 	case SNDRV_HDSPM_IOCTL_GET_STATUS:
+		memset(&status, 0, sizeof(status));
+
 		status.card_type = hdspm->io_type;
 
 		status.autosync_source = hdspm_autosync_ref(hdspm);
@@ -6235,7 +6246,7 @@
 			status.card_specific.madi.madi_input =
 				(statusregister & HDSPM_AB_int) ? 1 : 0;
 			status.card_specific.madi.channel_format =
-				(statusregister & HDSPM_TX_64ch) ? 1 : 0;
+				(statusregister & HDSPM_RX_64ch) ? 1 : 0;
 			/* TODO: Mac driver sets it when f_s>48kHz */
 			status.card_specific.madi.frame_format = 0;
 
@@ -6243,13 +6254,15 @@
 			break;
 		}
 
-		if (copy_to_user((void __user *) arg, &status, sizeof(status)))
+		if (copy_to_user(argp, &status, sizeof(status)))
 			return -EFAULT;
 
 
 		break;
 
 	case SNDRV_HDSPM_IOCTL_GET_VERSION:
+		memset(&hdspm_version, 0, sizeof(hdspm_version));
+
 		hdspm_version.card_type = hdspm->io_type;
 		strncpy(hdspm_version.cardname, hdspm->card_name,
 				sizeof(hdspm_version.cardname));
@@ -6260,13 +6273,13 @@
 		if (hdspm->tco)
 			hdspm_version.addons |= HDSPM_ADDON_TCO;
 
-		if (copy_to_user((void __user *) arg, &hdspm_version,
+		if (copy_to_user(argp, &hdspm_version,
 					sizeof(hdspm_version)))
 			return -EFAULT;
 		break;
 
 	case SNDRV_HDSPM_IOCTL_GET_MIXER:
-		if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer)))
+		if (copy_from_user(&mixer, argp, sizeof(mixer)))
 			return -EFAULT;
 		if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer,
 					sizeof(struct hdspm_mixer)))
@@ -6483,13 +6496,6 @@
 	strcpy(card->driver, "HDSPM");
 
 	switch (hdspm->firmware_rev) {
-	case HDSPM_MADI_REV:
-	case HDSPM_MADI_OLD_REV:
-	case HDSPM_MADI_ANCIENT_REV:
-		hdspm->io_type = MADI;
-		hdspm->card_name = "RME MADI";
-		hdspm->midiPorts = 3;
-		break;
 	case HDSPM_RAYDAT_REV:
 		hdspm->io_type = RayDAT;
 		hdspm->card_name = "RME RayDAT";
@@ -6505,17 +6511,25 @@
 		hdspm->card_name = "RME MADIface";
 		hdspm->midiPorts = 1;
 		break;
-	case HDSPM_AES_REV:
-	case HDSPM_AES32_REV:
-	case HDSPM_AES32_OLD_REV:
-		hdspm->io_type = AES32;
-		hdspm->card_name = "RME AES32";
-		hdspm->midiPorts = 2;
-		break;
 	default:
-		snd_printk(KERN_ERR "HDSPM: unknown firmware revision %x\n",
+		if ((hdspm->firmware_rev == 0xf0) ||
+			((hdspm->firmware_rev >= 0xe6) &&
+					(hdspm->firmware_rev <= 0xea))) {
+			hdspm->io_type = AES32;
+			hdspm->card_name = "RME AES32";
+			hdspm->midiPorts = 2;
+		} else if ((hdspm->firmware_rev == 0xd5) ||
+			((hdspm->firmware_rev >= 0xc8)  &&
+				(hdspm->firmware_rev <= 0xcf))) {
+			hdspm->io_type = MADI;
+			hdspm->card_name = "RME MADI";
+			hdspm->midiPorts = 3;
+		} else {
+			snd_printk(KERN_ERR
+				"HDSPM: unknown firmware revision %x\n",
 				hdspm->firmware_rev);
-		return -ENODEV;
+			return -ENODEV;
+		}
 	}
 
 	err = pci_enable_device(pci);
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index bcf6152..5ffb20b 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1234,7 +1234,7 @@
 		goto error;
 	}
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, sis)) {
 		printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq);
 		goto error;
@@ -1340,7 +1340,7 @@
 	if (rc)
 		goto error_out_cleanup;
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, sis)) {
 		printk(KERN_ERR "unable to allocate irq %d\n", sis->irq);
 		goto error_out_cleanup;
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 2571a67..c500816 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1493,9 +1493,10 @@
 		return err;
 	}
 	if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES,
-				       sonic->midi_port, MPU401_INFO_INTEGRATED,
-				       sonic->irq, 0,
-				       &midi_uart)) < 0) {
+				       sonic->midi_port,
+				       MPU401_INFO_INTEGRATED |
+				       MPU401_INFO_IRQ_HOOK,
+				       -1, &midi_uart)) < 0) {
 		snd_card_free(card);
 		return err;
 	}
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index d8a128f..5e707ef 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -148,8 +148,9 @@
 	if (trident->device != TRIDENT_DEVICE_ID_SI7018 &&
 	    (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
 				       trident->midi_port,
-				       MPU401_INFO_INTEGRATED,
-				       trident->irq, 0, &trident->rmidi)) < 0) {
+				       MPU401_INFO_INTEGRATED |
+				       MPU401_INFO_IRQ_HOOK,
+				       -1, &trident->rmidi)) < 0) {
 		snd_card_free(card);
 		return err;
 	}
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index f03fd62..c3656ff 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1175,6 +1175,7 @@
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 	struct via_rate_lock *ratep;
+	bool use_src = false;
 
 	runtime->hw = snd_via82xx_hw;
 	
@@ -1196,6 +1197,7 @@
 				     SNDRV_PCM_RATE_8000_48000);
 		runtime->hw.rate_min = 8000;
 		runtime->hw.rate_max = 48000;
+		use_src = true;
 	} else if (! ratep->rate) {
 		int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC;
 		runtime->hw.rates = chip->ac97->rates[idx];
@@ -1212,6 +1214,12 @@
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 		return err;
 
+	if (use_src) {
+		err = snd_pcm_hw_rule_noresample(runtime, 48000);
+		if (err < 0)
+			return err;
+	}
+
 	runtime->private_data = viadev;
 	viadev->substream = substream;
 
@@ -2068,8 +2076,9 @@
 	pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg);
 	if (chip->mpu_res) {
 		if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A,
-					mpu_port, MPU401_INFO_INTEGRATED,
-					chip->irq, 0, &chip->rmidi) < 0) {
+					mpu_port, MPU401_INFO_INTEGRATED |
+					MPU401_INFO_IRQ_HOOK, -1,
+					&chip->rmidi) < 0) {
 			printk(KERN_WARNING "unable to initialize MPU-401"
 			       " at 0x%lx, skipping\n", mpu_port);
 			legacy &= ~VIA_FUNC_ENABLE_MIDI;
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 511d576..3253b04 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -305,8 +305,9 @@
 	if (chip->mpu_res) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI,
 					       mpu_port[dev],
-					       MPU401_INFO_INTEGRATED,
-					       pci->irq, 0, &chip->rawmidi)) < 0) {
+					       MPU401_INFO_INTEGRATED |
+					       MPU401_INFO_IRQ_HOOK,
+					       -1, &chip->rawmidi)) < 0) {
 			printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]);
 			legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */
 			pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index f3260e6..66ea71b 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -897,6 +897,18 @@
 	struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_ymfpci_pcm *ypcm;
+	int err;
+
+	runtime->hw = snd_ymfpci_playback;
+	/* FIXME? True value is 256/48 = 5.33333 ms */
+	err = snd_pcm_hw_constraint_minmax(runtime,
+					   SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+					   5334, UINT_MAX);
+	if (err < 0)
+		return err;
+	err = snd_pcm_hw_rule_noresample(runtime, 48000);
+	if (err < 0)
+		return err;
 
 	ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL);
 	if (ypcm == NULL)
@@ -904,11 +916,8 @@
 	ypcm->chip = chip;
 	ypcm->type = PLAYBACK_VOICE;
 	ypcm->substream = substream;
-	runtime->hw = snd_ymfpci_playback;
 	runtime->private_data = ypcm;
 	runtime->private_free = snd_ymfpci_pcm_free_substream;
-	/* FIXME? True value is 256/48 = 5.33333 ms */
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX);
 	return 0;
 }
 
@@ -1013,6 +1022,18 @@
 	struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_ymfpci_pcm *ypcm;
+	int err;
+
+	runtime->hw = snd_ymfpci_capture;
+	/* FIXME? True value is 256/48 = 5.33333 ms */
+	err = snd_pcm_hw_constraint_minmax(runtime,
+					   SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+					   5334, UINT_MAX);
+	if (err < 0)
+		return err;
+	err = snd_pcm_hw_rule_noresample(runtime, 48000);
+	if (err < 0)
+		return err;
 
 	ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL);
 	if (ypcm == NULL)
@@ -1022,9 +1043,6 @@
 	ypcm->substream = substream;	
 	ypcm->capture_bank_number = capture_bank_number;
 	chip->capture_substream[capture_bank_number] = substream;
-	runtime->hw = snd_ymfpci_capture;
-	/* FIXME? True value is 256/48 = 5.33333 ms */
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX);
 	runtime->private_data = ypcm;
 	runtime->private_free = snd_ymfpci_pcm_free_substream;
 	snd_ymfpci_hw_start(chip);
@@ -1615,7 +1633,7 @@
 YMFPCI_DOUBLE("ADC Capture Volume", 0, YDSXGR_PRIADCLOOPVOL),
 YMFPCI_DOUBLE("ADC Playback Volume", 1, YDSXGR_SECADCOUTVOL),
 YMFPCI_DOUBLE("ADC Capture Volume", 1, YDSXGR_SECADCLOOPVOL),
-YMFPCI_DOUBLE("FM Legacy Volume", 0, YDSXGR_LEGACYOUTVOL),
+YMFPCI_DOUBLE("FM Legacy Playback Volume", 0, YDSXGR_LEGACYOUTVOL),
 YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ", PLAYBACK,VOLUME), 0, YDSXGR_ZVOUTVOL),
 YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("", CAPTURE,VOLUME), 0, YDSXGR_ZVLOOPVOL),
 YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ",PLAYBACK,VOLUME), 1, YDSXGR_SPDIFOUTVOL),
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 8f064c7..4080bec 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -82,7 +82,6 @@
 
 static int keywest_remove(struct i2c_client *client)
 {
-	i2c_set_clientdata(client, NULL);
 	if (! keywest_ctx)
 		return 0;
 	if (client == keywest_ctx->client)
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index bc823a5..775bd95 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -845,7 +845,7 @@
 		return ret;
 	}
 
-	ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED,
+	ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0,
 			  SND_PS3_DRIVER_NAME, &the_card);
 	if (ret) {
 		pr_info("%s: request_irq failed (%d)\n", __func__, ret);
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
index 7aa5b76..177f713 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -211,7 +211,7 @@
 	/* DMA setup */
 	name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
 	ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
-					au1000_dma_interrupt, IRQF_DISABLED,
+					au1000_dma_interrupt, 0,
 					&ctx->stream[s]);
 	set_dma_mode(ctx->stream[s].dma,
 		     get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 3f4920d..dc8a2b2 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1419,7 +1419,7 @@
 	/* Check if the IRQ number is valid and request it */
 	if (dac33->irq >= 0) {
 		ret = request_irq(dac33->irq, dac33_interrupt_handler,
-				  IRQF_TRIGGER_RISING | IRQF_DISABLED,
+				  IRQF_TRIGGER_RISING,
 				  codec->name, codec);
 		if (ret < 0) {
 			dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 4e3626b..ae8d680 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -268,7 +268,7 @@
 	nuc900_audio = nuc900_ac97_data;
 
 	if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
-			IRQF_DISABLED, "nuc900-dma", substream))
+			0, "nuc900-dma", substream))
 		return -EBUSY;
 
 	runtime->private_data = nuc900_audio;
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 65ea538..b5e922f 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -444,7 +444,7 @@
 	}
 
 	ret = request_irq(irq_res->start, s3c_ac97_irq,
-					IRQF_DISABLED, "AC97", NULL);
+					0, "AC97", NULL);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "ac97: interrupt request failed.\n");
 		goto err4;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 916b9f9..a32fd16 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1285,7 +1285,7 @@
 	pm_runtime_enable(&pdev->dev);
 	dev_set_drvdata(&pdev->dev, master);
 
-	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
+	ret = request_irq(irq, &fsi_interrupt, 0,
 			  id_entry->name, master);
 	if (ret) {
 		dev_err(&pdev->dev, "irq request err\n");
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 743d07b..a4e3f55 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -201,7 +201,7 @@
 	if (!drvdata->base)
 		return -EBUSY;
 	err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
-			       IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
+			       0, dev_name(&pdev->dev), drvdata);
 	if (err < 0)
 		return err;
 
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index ad7d4d7..f036776 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -962,7 +962,7 @@
 	amd7930_idle(amd);
 
 	if (request_irq(irq, snd_amd7930_interrupt,
-			IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) {
+			IRQF_SHARED, "amd7930", amd)) {
 		snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n",
 			   dev, irq);
 		snd_amd7930_free(amd);
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 1e3ae33..07bcfe4 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -16,6 +16,7 @@
 
 #include <linux/firmware.h>
 #include <linux/bitrev.h>
+#include <linux/kernel.h>
 
 #include "firmware.h"
 #include "chip.h"
@@ -59,21 +60,19 @@
 	unsigned int txt_offset; /* current position in txt_data */
 };
 
-static u8 usb6fire_fw_ihex_nibble(const u8 n)
-{
-	if (n >= '0' && n <= '9')
-		return n - '0';
-	else if (n >= 'A' && n <= 'F')
-		return n - ('A' - 10);
-	else if (n >= 'a' && n <= 'f')
-		return n - ('a' - 10);
-	return 0;
-}
-
 static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc)
 {
-	u8 val = (usb6fire_fw_ihex_nibble(data[0]) << 4) |
-			usb6fire_fw_ihex_nibble(data[1]);
+	u8 val = 0;
+	int hval;
+
+	hval = hex_to_bin(data[0]);
+	if (hval >= 0)
+		val |= (hval << 4);
+
+	hval = hex_to_bin(data[1]);
+	if (hval >= 0)
+		val |= hval;
+
 	*crc += val;
 	return val;
 }
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 8beb775..3efc21c 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -67,6 +67,7 @@
 	    * Native Instruments Guitar Rig mobile
 	    * Native Instruments Traktor Kontrol X1
 	    * Native Instruments Traktor Kontrol S4
+	    * Native Instruments Maschine Controller
 
 	   To compile this driver as a module, choose M here: the module
 	   will be called snd-usb-caiaq.
@@ -85,6 +86,7 @@
 	   * Native Instruments Kore Controller 2
 	   * Native Instruments Audio Kontrol 1
 	   * Native Instruments Traktor Kontrol S4
+	   * Native Instruments Maschine Controller
 
 config SND_USB_US122L
 	tristate "Tascam US-122L USB driver"
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index cf9ed66..ac256dc 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -3,16 +3,16 @@
 #
 
 snd-usb-audio-objs := 	card.o \
+			clock.o \
+			endpoint.o \
+			format.o \
+			helper.o \
 			mixer.o \
 			mixer_quirks.o \
+			pcm.o \
 			proc.o \
 			quirks.o \
-			format.o \
-			endpoint.o \
-			urb.o \
-			pcm.o \
-			helper.o \
-			clock.o
+			stream.o
 
 snd-usbmidi-lib-objs := midi.o
 
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493c..2cf87f5 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@
 
 	for (i = 0; i < N_URBS; i++) {
 		usb_kill_urb(dev->data_urbs_in[i]);
-		usb_kill_urb(dev->data_urbs_out[i]);
+
+		if (test_bit(i, &dev->outurb_active_mask))
+			usb_kill_urb(dev->data_urbs_out[i]);
 	}
+
+	dev->outurb_active_mask = 0;
 }
 
 static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -612,8 +616,9 @@
 {
 	struct snd_usb_caiaq_cb_info *info = urb->context;
 	struct snd_usb_caiaqdev *dev;
-	struct urb *out;
-	int frame, len, send_it = 0, outframe = 0;
+	struct urb *out = NULL;
+	int i, frame, len, send_it = 0, outframe = 0;
+	size_t offset = 0;
 
 	if (urb->status || !info)
 		return;
@@ -623,7 +628,17 @@
 	if (!dev->streaming)
 		return;
 
-	out = dev->data_urbs_out[info->index];
+	/* find an unused output urb that is unused */
+	for (i = 0; i < N_URBS; i++)
+		if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
+			out = dev->data_urbs_out[i];
+			break;
+		}
+
+	if (!out) {
+		log("Unable to find an output urb to use\n");
+		goto requeue;
+	}
 
 	/* read the recently received packet and send back one which has
 	 * the same layout */
@@ -634,7 +649,8 @@
 		len = urb->iso_frame_desc[outframe].actual_length;
 		out->iso_frame_desc[outframe].length = len;
 		out->iso_frame_desc[outframe].actual_length = 0;
-		out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+		out->iso_frame_desc[outframe].offset = offset;
+		offset += len;
 
 		if (len > 0) {
 			spin_lock(&dev->spinlock);
@@ -650,11 +666,15 @@
 	}
 
 	if (send_it) {
-		out->number_of_packets = FRAMES_PER_URB;
+		out->number_of_packets = outframe;
 		out->transfer_flags = URB_ISO_ASAP;
 		usb_submit_urb(out, GFP_ATOMIC);
+	} else {
+		struct snd_usb_caiaq_cb_info *oinfo = out->context;
+		clear_bit(oinfo->index, &dev->outurb_active_mask);
 	}
 
+requeue:
 	/* re-submit inbound urb */
 	for (frame = 0; frame < FRAMES_PER_URB; frame++) {
 		urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -676,6 +696,8 @@
 		dev->output_running = 1;
 		wake_up(&dev->prepare_wait_queue);
 	}
+
+	clear_bit(info->index, &dev->outurb_active_mask);
 }
 
 static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -827,6 +849,9 @@
 	if (!dev->data_cb_info)
 		return -ENOMEM;
 
+	dev->outurb_active_mask = 0;
+	BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
+
 	for (i = 0; i < N_URBS; i++) {
 		dev->data_cb_info[i].dev = dev;
 		dev->data_cb_info[i].index = i;
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 45bc4a2..3eb605b 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -50,7 +50,8 @@
 			 "{Native Instruments, Session I/O},"
 			 "{Native Instruments, GuitarRig mobile}"
 			 "{Native Instruments, Traktor Kontrol X1}"
-			 "{Native Instruments, Traktor Kontrol S4}");
+			 "{Native Instruments, Traktor Kontrol S4}"
+			 "{Native Instruments, Maschine Controller}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
@@ -146,6 +147,11 @@
 		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
 		.idProduct =    USB_PID_TRAKTORAUDIO2
 	},
+	{
+		.match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
+		.idProduct =    USB_PID_MASCHINECONTROLLER
+	},
 	{ /* terminator */ }
 };
 
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index b2b3101..562b0bf 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -18,6 +18,7 @@
 #define USB_PID_TRAKTORKONTROLX1	0x2305
 #define USB_PID_TRAKTORKONTROLS4	0xbaff
 #define USB_PID_TRAKTORAUDIO2		0x041d
+#define USB_PID_MASCHINECONTROLLER  0x0808
 
 #define EP1_BUFSIZE 64
 #define EP4_BUFSIZE 512
@@ -96,6 +97,7 @@
 	int input_panic, output_panic, warned;
 	char *audio_in_buf, *audio_out_buf;
 	unsigned int samplerates, bpp;
+	unsigned long outurb_active_mask;
 
 	struct snd_pcm_substream *sub_playback[MAX_STREAMS];
 	struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 4432ef7..26a121b 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@
 static unsigned short keycode_rk2[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
 					 KEY_5, KEY_6, KEY_7 };
 static unsigned short keycode_rk3[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
-					 KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
+					 KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
 
 static unsigned short keycode_kore[] = {
 	KEY_FN_F1,      /* "menu"               */
@@ -67,6 +67,61 @@
 	KEY_BRL_DOT5
 };
 
+#define MASCHINE_BUTTONS   (42)
+#define MASCHINE_BUTTON(X) ((X) + BTN_MISC)
+#define MASCHINE_PADS      (16)
+#define MASCHINE_PAD(X)    ((X) + ABS_PRESSURE)
+
+static unsigned short keycode_maschine[] = {
+	MASCHINE_BUTTON(40), /* mute       */
+	MASCHINE_BUTTON(39), /* solo       */
+	MASCHINE_BUTTON(38), /* select     */
+	MASCHINE_BUTTON(37), /* duplicate  */
+	MASCHINE_BUTTON(36), /* navigate   */
+	MASCHINE_BUTTON(35), /* pad mode   */
+	MASCHINE_BUTTON(34), /* pattern    */
+	MASCHINE_BUTTON(33), /* scene      */
+	KEY_RESERVED, /* spacer */
+
+	MASCHINE_BUTTON(30), /* rec        */
+	MASCHINE_BUTTON(31), /* erase      */
+	MASCHINE_BUTTON(32), /* shift      */
+	MASCHINE_BUTTON(28), /* grid       */
+	MASCHINE_BUTTON(27), /* >          */
+	MASCHINE_BUTTON(26), /* <          */
+	MASCHINE_BUTTON(25), /* restart    */
+
+	MASCHINE_BUTTON(21), /* E          */
+	MASCHINE_BUTTON(22), /* F          */
+	MASCHINE_BUTTON(23), /* G          */
+	MASCHINE_BUTTON(24), /* H          */
+	MASCHINE_BUTTON(20), /* D          */
+	MASCHINE_BUTTON(19), /* C          */
+	MASCHINE_BUTTON(18), /* B          */
+	MASCHINE_BUTTON(17), /* A          */
+
+	MASCHINE_BUTTON(0),  /* control    */
+	MASCHINE_BUTTON(2),  /* browse     */
+	MASCHINE_BUTTON(4),  /* <          */
+	MASCHINE_BUTTON(6),  /* snap       */
+	MASCHINE_BUTTON(7),  /* autowrite  */
+	MASCHINE_BUTTON(5),  /* >          */
+	MASCHINE_BUTTON(3),  /* sampling   */
+	MASCHINE_BUTTON(1),  /* step       */
+
+	MASCHINE_BUTTON(15), /* 8 softkeys */
+	MASCHINE_BUTTON(14),
+	MASCHINE_BUTTON(13),
+	MASCHINE_BUTTON(12),
+	MASCHINE_BUTTON(11),
+	MASCHINE_BUTTON(10),
+	MASCHINE_BUTTON(9),
+	MASCHINE_BUTTON(8),
+
+	MASCHINE_BUTTON(16), /* note repeat */
+	MASCHINE_BUTTON(29)  /* play        */
+};
+
 #define KONTROLX1_INPUTS	(40)
 #define KONTROLS4_BUTTONS	(12 * 8)
 #define KONTROLS4_AXIS		(46)
@@ -218,6 +273,29 @@
 		input_report_abs(input_dev, ABS_HAT3Y, i);
 		input_sync(input_dev);
 		break;
+
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
+		/* 4 under the left screen */
+		input_report_abs(input_dev, ABS_HAT0X, decode_erp(buf[21], buf[20]));
+		input_report_abs(input_dev, ABS_HAT0Y, decode_erp(buf[15], buf[14]));
+		input_report_abs(input_dev, ABS_HAT1X, decode_erp(buf[9],  buf[8]));
+		input_report_abs(input_dev, ABS_HAT1Y, decode_erp(buf[3],  buf[2]));
+
+		/* 4 under the right screen */
+		input_report_abs(input_dev, ABS_HAT2X, decode_erp(buf[19], buf[18]));
+		input_report_abs(input_dev, ABS_HAT2Y, decode_erp(buf[13], buf[12]));
+		input_report_abs(input_dev, ABS_HAT3X, decode_erp(buf[7],  buf[6]));
+		input_report_abs(input_dev, ABS_HAT3Y, decode_erp(buf[1],  buf[0]));
+
+		/* volume */
+		input_report_abs(input_dev, ABS_RX, decode_erp(buf[17], buf[16]));
+		/* tempo */
+		input_report_abs(input_dev, ABS_RY, decode_erp(buf[11], buf[10]));
+		/* swing */
+		input_report_abs(input_dev, ABS_RZ, decode_erp(buf[5],  buf[4]));
+
+		input_sync(input_dev);
+		break;
 	}
 }
 
@@ -400,6 +478,25 @@
 	input_sync(dev->input_dev);
 }
 
+#define MASCHINE_MSGBLOCK_SIZE 2
+
+static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *dev,
+					const unsigned char *buf,
+					unsigned int len)
+{
+	unsigned int i, pad_id;
+	uint16_t pressure;
+
+	for (i = 0; i < MASCHINE_PADS; i++) {
+		pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]);
+		pad_id = pressure >> 12;
+
+		input_report_abs(dev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff);
+	}
+
+	input_sync(dev->input_dev);
+}
+
 static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb)
 {
 	struct snd_usb_caiaqdev *dev = urb->context;
@@ -425,6 +522,13 @@
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4):
 		snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length);
 		break;
+
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
+		if (urb->actual_length < (MASCHINE_PADS * MASCHINE_MSGBLOCK_SIZE))
+			goto requeue;
+
+		snd_usb_caiaq_maschine_dispatch(dev, buf, urb->actual_length);
+		break;
 	}
 
 requeue:
@@ -444,6 +548,7 @@
 	switch (dev->chip.usb_id) {
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4):
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
 		if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0)
 			return -EIO;
 		break;
@@ -462,6 +567,7 @@
 	switch (dev->chip.usb_id) {
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4):
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
 		usb_kill_urb(dev->ep4_in_urb);
 		break;
 	}
@@ -652,6 +758,50 @@
 
 		break;
 
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
+		input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS);
+		input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) |
+			BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) |
+			BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) |
+			BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) |
+			BIT_MASK(ABS_RX) | BIT_MASK(ABS_RY) |
+			BIT_MASK(ABS_RZ);
+
+		BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_maschine));
+		memcpy(dev->keycode, keycode_maschine, sizeof(keycode_maschine));
+		input->keycodemax = ARRAY_SIZE(keycode_maschine);
+
+		for (i = 0; i < MASCHINE_PADS; i++) {
+			input->absbit[0] |= MASCHINE_PAD(i);
+			input_set_abs_params(input, MASCHINE_PAD(i), 0, 0xfff, 5, 10);
+		}
+
+		input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT1X, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT1Y, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT2X, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT2Y, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT3X, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_HAT3Y, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_RX, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_RY, 0, 999, 0, 10);
+		input_set_abs_params(input, ABS_RZ, 0, 999, 0, 10);
+
+		dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL);
+		if (!dev->ep4_in_urb) {
+			ret = -ENOMEM;
+			goto exit_free_idev;
+		}
+
+		usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev,
+				  usb_rcvbulkpipe(usb_dev, 0x4),
+				  dev->ep4_in_buf, EP4_BUFSIZE,
+				  snd_usb_caiaq_ep4_reply_dispatch, dev);
+
+		snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5);
+		break;
+
 	default:
 		/* no input methods supported on this device */
 		goto exit_free_idev;
@@ -664,15 +814,17 @@
 	for (i = 0; i < input->keycodemax; i++)
 		__set_bit(dev->keycode[i], input->keybit);
 
+	dev->input_dev = input;
+
 	ret = input_register_device(input);
 	if (ret < 0)
 		goto exit_free_idev;
 
-	dev->input_dev = input;
 	return 0;
 
 exit_free_idev:
 	input_free_device(input);
+	dev->input_dev = NULL;
 	return ret;
 }
 
@@ -688,4 +840,3 @@
 	input_unregister_device(dev->input_dev);
 	dev->input_dev = NULL;
 }
-
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 781d9e6..c1575ea 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -65,9 +65,9 @@
 #include "helper.h"
 #include "debug.h"
 #include "pcm.h"
-#include "urb.h"
 #include "format.h"
 #include "power.h"
+#include "stream.h"
 
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("USB Audio");
@@ -185,7 +185,7 @@
 		return -EINVAL;
 	}
 
-	if (! snd_usb_parse_audio_endpoints(chip, interface)) {
+	if (! snd_usb_parse_audio_interface(chip, interface)) {
 		usb_set_interface(dev, interface, 0); /* reset the current interface */
 		usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
 		return -EINVAL;
@@ -530,8 +530,11 @@
 	return chip;
 
  __error:
-	if (chip && !chip->num_interfaces)
-		snd_card_free(chip->card);
+	if (chip) {
+		if (!chip->num_interfaces)
+			snd_card_free(chip->card);
+		chip->probing = 0;
+	}
 	mutex_unlock(&register_mutex);
  __err_val:
 	return NULL;
diff --git a/sound/usb/card.h b/sound/usb/card.h
index ae4251d..a39edcc 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -94,6 +94,8 @@
 	spinlock_t lock;
 
 	struct snd_urb_ops ops;		/* callbacks (must be filled at init) */
+	int last_frame_number;          /* stored frame number */
+	int last_delay;                 /* stored delay */
 };
 
 struct snd_usb_stream {
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 075195e..379baad 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -91,7 +91,7 @@
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 			      UAC2_CX_CLOCK_SELECTOR << 8,
 			      snd_usb_ctrl_intf(chip) | (selector_id << 8),
-			      &buf, sizeof(buf), 1000);
+			      &buf, sizeof(buf));
 
 	if (ret < 0)
 		return ret;
@@ -118,7 +118,7 @@
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
 			      UAC2_CS_CONTROL_CLOCK_VALID << 8,
 			      snd_usb_ctrl_intf(chip) | (source_id << 8),
-			      &data, sizeof(data), 1000);
+			      &data, sizeof(data));
 
 	if (err < 0) {
 		snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n",
@@ -222,7 +222,7 @@
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
 				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-				   data, sizeof(data), 1000)) < 0) {
+				   data, sizeof(data))) < 0) {
 		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
 			   dev->devnum, iface, fmt->altsetting, rate, ep);
 		return err;
@@ -231,7 +231,7 @@
 	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
 				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-				   data, sizeof(data), 1000)) < 0) {
+				   data, sizeof(data))) < 0) {
 		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
 			   dev->devnum, iface, fmt->altsetting, ep);
 		return 0; /* some devices don't support reading */
@@ -273,7 +273,7 @@
 				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
 				   UAC2_CS_CONTROL_SAM_FREQ << 8,
 				   snd_usb_ctrl_intf(chip) | (clock << 8),
-				   data, sizeof(data), 1000)) < 0) {
+				   data, sizeof(data))) < 0) {
 		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
 			   dev->devnum, iface, fmt->altsetting, rate);
 		return err;
@@ -283,7 +283,7 @@
 				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
 				   UAC2_CS_CONTROL_SAM_FREQ << 8,
 				   snd_usb_ctrl_intf(chip) | (clock << 8),
-				   data, sizeof(data), 1000)) < 0) {
+				   data, sizeof(data))) < 0) {
 		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);
 		return err;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7c0d21e..81c6ede 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -15,436 +15,951 @@
  *
  */
 
+#include <linux/gfp.h>
 #include <linux/init.h>
-#include <linux/slab.h>
 #include <linux/usb.h>
 #include <linux/usb/audio.h>
-#include <linux/usb/audio-v2.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 
 #include "usbaudio.h"
-#include "card.h"
-#include "proc.h"
-#include "quirks.h"
-#include "endpoint.h"
-#include "urb.h"
-#include "pcm.h"
 #include "helper.h"
-#include "format.h"
-#include "clock.h"
+#include "card.h"
+#include "endpoint.h"
+#include "pcm.h"
 
 /*
- * free a substream
+ * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
+ * this will overflow at approx 524 kHz
  */
-static void free_substream(struct snd_usb_substream *subs)
+static inline unsigned get_usb_full_speed_rate(unsigned int rate)
 {
-	struct list_head *p, *n;
-
-	if (!subs->num_formats)
-		return; /* not initialized */
-	list_for_each_safe(p, n, &subs->fmt_list) {
-		struct audioformat *fp = list_entry(p, struct audioformat, list);
-		kfree(fp->rate_table);
-		kfree(fp);
-	}
-	kfree(subs->rate_list.list);
+	return ((rate << 13) + 62) / 125;
 }
 
-
 /*
- * free a usb stream instance
+ * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
+ * this will overflow at approx 4 MHz
  */
-static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
+static inline unsigned get_usb_high_speed_rate(unsigned int rate)
 {
-	free_substream(&stream->substream[0]);
-	free_substream(&stream->substream[1]);
-	list_del(&stream->list);
-	kfree(stream);
+	return ((rate << 10) + 62) / 125;
 }
 
-static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_usb_stream *stream = pcm->private_data;
-	if (stream) {
-		stream->pcm = NULL;
-		snd_usb_audio_stream_free(stream);
-	}
-}
-
-
 /*
- * add this endpoint to the chip instance.
- * if a stream with the same endpoint already exists, append to it.
- * if not, create a new pcm stream.
+ * unlink active urbs.
  */
-int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp)
+static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
 {
-	struct list_head *p;
-	struct snd_usb_stream *as;
-	struct snd_usb_substream *subs;
-	struct snd_pcm *pcm;
-	int err;
+	struct snd_usb_audio *chip = subs->stream->chip;
+	unsigned int i;
+	int async;
 
-	list_for_each(p, &chip->pcm_list) {
-		as = list_entry(p, struct snd_usb_stream, list);
-		if (as->fmt_type != fp->fmt_type)
-			continue;
-		subs = &as->substream[stream];
-		if (!subs->endpoint)
-			continue;
-		if (subs->endpoint == fp->endpoint) {
-			list_add_tail(&fp->list, &subs->fmt_list);
-			subs->num_formats++;
-			subs->formats |= fp->formats;
-			return 0;
+	subs->running = 0;
+
+	if (!force && subs->stream->chip->shutdown) /* to be sure... */
+		return -EBADFD;
+
+	async = !can_sleep && chip->async_unlink;
+
+	if (!async && in_interrupt())
+		return 0;
+
+	for (i = 0; i < subs->nurbs; i++) {
+		if (test_bit(i, &subs->active_mask)) {
+			if (!test_and_set_bit(i, &subs->unlink_mask)) {
+				struct urb *u = subs->dataurb[i].urb;
+				if (async)
+					usb_unlink_urb(u);
+				else
+					usb_kill_urb(u);
+			}
 		}
 	}
-	/* look for an empty stream */
-	list_for_each(p, &chip->pcm_list) {
-		as = list_entry(p, struct snd_usb_stream, list);
-		if (as->fmt_type != fp->fmt_type)
-			continue;
-		subs = &as->substream[stream];
-		if (subs->endpoint)
-			continue;
-		err = snd_pcm_new_stream(as->pcm, stream, 1);
-		if (err < 0)
-			return err;
-		snd_usb_init_substream(as, stream, fp);
-		return 0;
+	if (subs->syncpipe) {
+		for (i = 0; i < SYNC_URBS; i++) {
+			if (test_bit(i+16, &subs->active_mask)) {
+				if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
+					struct urb *u = subs->syncurb[i].urb;
+					if (async)
+						usb_unlink_urb(u);
+					else
+						usb_kill_urb(u);
+				}
+			}
+		}
 	}
+	return 0;
+}
 
-	/* create a new pcm */
-	as = kzalloc(sizeof(*as), GFP_KERNEL);
-	if (!as)
-		return -ENOMEM;
-	as->pcm_index = chip->pcm_devs;
-	as->chip = chip;
-	as->fmt_type = fp->fmt_type;
-	err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs,
-			  stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0,
-			  stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1,
-			  &pcm);
-	if (err < 0) {
-		kfree(as);
-		return err;
+
+/*
+ * release a urb data
+ */
+static void release_urb_ctx(struct snd_urb_ctx *u)
+{
+	if (u->urb) {
+		if (u->buffer_size)
+			usb_free_coherent(u->subs->dev, u->buffer_size,
+					u->urb->transfer_buffer,
+					u->urb->transfer_dma);
+		usb_free_urb(u->urb);
+		u->urb = NULL;
 	}
-	as->pcm = pcm;
-	pcm->private_data = as;
-	pcm->private_free = snd_usb_audio_pcm_free;
-	pcm->info_flags = 0;
-	if (chip->pcm_devs > 0)
-		sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs);
+}
+
+/*
+ *  wait until all urbs are processed.
+ */
+static int wait_clear_urbs(struct snd_usb_substream *subs)
+{
+	unsigned long end_time = jiffies + msecs_to_jiffies(1000);
+	unsigned int i;
+	int alive;
+
+	do {
+		alive = 0;
+		for (i = 0; i < subs->nurbs; i++) {
+			if (test_bit(i, &subs->active_mask))
+				alive++;
+		}
+		if (subs->syncpipe) {
+			for (i = 0; i < SYNC_URBS; i++) {
+				if (test_bit(i + 16, &subs->active_mask))
+					alive++;
+			}
+		}
+		if (! alive)
+			break;
+		schedule_timeout_uninterruptible(1);
+	} while (time_before(jiffies, end_time));
+	if (alive)
+		snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
+	return 0;
+}
+
+/*
+ * release a substream
+ */
+void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
+{
+	int i;
+
+	/* stop urbs (to be sure) */
+	deactivate_urbs(subs, force, 1);
+	wait_clear_urbs(subs);
+
+	for (i = 0; i < MAX_URBS; i++)
+		release_urb_ctx(&subs->dataurb[i]);
+	for (i = 0; i < SYNC_URBS; i++)
+		release_urb_ctx(&subs->syncurb[i]);
+	usb_free_coherent(subs->dev, SYNC_URBS * 4,
+			subs->syncbuf, subs->sync_dma);
+	subs->syncbuf = NULL;
+	subs->nurbs = 0;
+}
+
+/*
+ * complete callback from data urb
+ */
+static void snd_complete_urb(struct urb *urb)
+{
+	struct snd_urb_ctx *ctx = urb->context;
+	struct snd_usb_substream *subs = ctx->subs;
+	struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
+	int err = 0;
+
+	if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
+	    !subs->running || /* can be stopped during retire callback */
+	    (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
+	    (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+		clear_bit(ctx->index, &subs->active_mask);
+		if (err < 0) {
+			snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
+			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		}
+	}
+}
+
+
+/*
+ * complete callback from sync urb
+ */
+static void snd_complete_sync_urb(struct urb *urb)
+{
+	struct snd_urb_ctx *ctx = urb->context;
+	struct snd_usb_substream *subs = ctx->subs;
+	struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
+	int err = 0;
+
+	if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
+	    !subs->running || /* can be stopped during retire callback */
+	    (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
+	    (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+		clear_bit(ctx->index + 16, &subs->active_mask);
+		if (err < 0) {
+			snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
+			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		}
+	}
+}
+
+
+/*
+ * initialize a substream for plaback/capture
+ */
+int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
+				unsigned int period_bytes,
+				unsigned int rate,
+				unsigned int frame_bits)
+{
+	unsigned int maxsize, i;
+	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int urb_packs, total_packs, packs_per_ms;
+	struct snd_usb_audio *chip = subs->stream->chip;
+
+	/* calculate the frequency in 16.16 format */
+	if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+		subs->freqn = get_usb_full_speed_rate(rate);
 	else
-		strcpy(pcm->name, "USB Audio");
+		subs->freqn = get_usb_high_speed_rate(rate);
+	subs->freqm = subs->freqn;
+	subs->freqshift = INT_MIN;
+	/* calculate max. frequency */
+	if (subs->maxpacksize) {
+		/* whatever fits into a max. size packet */
+		maxsize = subs->maxpacksize;
+		subs->freqmax = (maxsize / (frame_bits >> 3))
+				<< (16 - subs->datainterval);
+	} else {
+		/* no max. packet size: just take 25% higher than nominal */
+		subs->freqmax = subs->freqn + (subs->freqn >> 2);
+		maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
+				>> (16 - subs->datainterval);
+	}
+	subs->phase = 0;
 
-	snd_usb_init_substream(as, stream, fp);
+	if (subs->fill_max)
+		subs->curpacksize = subs->maxpacksize;
+	else
+		subs->curpacksize = maxsize;
 
-	list_add(&as->list, &chip->pcm_list);
-	chip->pcm_devs++;
+	if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
+		packs_per_ms = 8 >> subs->datainterval;
+	else
+		packs_per_ms = 1;
 
-	snd_usb_proc_pcm_format_add(as);
+	if (is_playback) {
+		urb_packs = max(chip->nrpacks, 1);
+		urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
+	} else
+		urb_packs = 1;
+	urb_packs *= packs_per_ms;
+	if (subs->syncpipe)
+		urb_packs = min(urb_packs, 1U << subs->syncinterval);
+
+	/* decide how many packets to be used */
+	if (is_playback) {
+		unsigned int minsize, maxpacks;
+		/* determine how small a packet can be */
+		minsize = (subs->freqn >> (16 - subs->datainterval))
+			  * (frame_bits >> 3);
+		/* with sync from device, assume it can be 12% lower */
+		if (subs->syncpipe)
+			minsize -= minsize >> 3;
+		minsize = max(minsize, 1u);
+		total_packs = (period_bytes + minsize - 1) / minsize;
+		/* we need at least two URBs for queueing */
+		if (total_packs < 2) {
+			total_packs = 2;
+		} else {
+			/* and we don't want too long a queue either */
+			maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
+			total_packs = min(total_packs, maxpacks);
+		}
+	} else {
+		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+			urb_packs >>= 1;
+		total_packs = MAX_URBS * urb_packs;
+	}
+	subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
+	if (subs->nurbs > MAX_URBS) {
+		/* too much... */
+		subs->nurbs = MAX_URBS;
+		total_packs = MAX_URBS * urb_packs;
+	} else if (subs->nurbs < 2) {
+		/* too little - we need at least two packets
+		 * to ensure contiguous playback/capture
+		 */
+		subs->nurbs = 2;
+	}
+
+	/* allocate and initialize data urbs */
+	for (i = 0; i < subs->nurbs; i++) {
+		struct snd_urb_ctx *u = &subs->dataurb[i];
+		u->index = i;
+		u->subs = subs;
+		u->packets = (i + 1) * total_packs / subs->nurbs
+			- i * total_packs / subs->nurbs;
+		u->buffer_size = maxsize * u->packets;
+		if (subs->fmt_type == UAC_FORMAT_TYPE_II)
+			u->packets++; /* for transfer delimiter */
+		u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
+		if (!u->urb)
+			goto out_of_memory;
+		u->urb->transfer_buffer =
+			usb_alloc_coherent(subs->dev, u->buffer_size,
+					   GFP_KERNEL, &u->urb->transfer_dma);
+		if (!u->urb->transfer_buffer)
+			goto out_of_memory;
+		u->urb->pipe = subs->datapipe;
+		u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
+		u->urb->interval = 1 << subs->datainterval;
+		u->urb->context = u;
+		u->urb->complete = snd_complete_urb;
+	}
+
+	if (subs->syncpipe) {
+		/* allocate and initialize sync urbs */
+		subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4,
+						 GFP_KERNEL, &subs->sync_dma);
+		if (!subs->syncbuf)
+			goto out_of_memory;
+		for (i = 0; i < SYNC_URBS; i++) {
+			struct snd_urb_ctx *u = &subs->syncurb[i];
+			u->index = i;
+			u->subs = subs;
+			u->packets = 1;
+			u->urb = usb_alloc_urb(1, GFP_KERNEL);
+			if (!u->urb)
+				goto out_of_memory;
+			u->urb->transfer_buffer = subs->syncbuf + i * 4;
+			u->urb->transfer_dma = subs->sync_dma + i * 4;
+			u->urb->transfer_buffer_length = 4;
+			u->urb->pipe = subs->syncpipe;
+			u->urb->transfer_flags = URB_ISO_ASAP |
+						 URB_NO_TRANSFER_DMA_MAP;
+			u->urb->number_of_packets = 1;
+			u->urb->interval = 1 << subs->syncinterval;
+			u->urb->context = u;
+			u->urb->complete = snd_complete_sync_urb;
+		}
+	}
+	return 0;
+
+out_of_memory:
+	snd_usb_release_substream_urbs(subs, 0);
+	return -ENOMEM;
+}
+
+/*
+ * prepare urb for full speed capture sync pipe
+ *
+ * fill the length and offset of each urb descriptor.
+ * the fixed 10.14 frequency is passed through the pipe.
+ */
+static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
+				    struct snd_pcm_runtime *runtime,
+				    struct urb *urb)
+{
+	unsigned char *cp = urb->transfer_buffer;
+	struct snd_urb_ctx *ctx = urb->context;
+
+	urb->dev = ctx->subs->dev; /* we need to set this at each time */
+	urb->iso_frame_desc[0].length = 3;
+	urb->iso_frame_desc[0].offset = 0;
+	cp[0] = subs->freqn >> 2;
+	cp[1] = subs->freqn >> 10;
+	cp[2] = subs->freqn >> 18;
+	return 0;
+}
+
+/*
+ * prepare urb for high speed capture sync pipe
+ *
+ * fill the length and offset of each urb descriptor.
+ * the fixed 12.13 frequency is passed as 16.16 through the pipe.
+ */
+static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
+				       struct snd_pcm_runtime *runtime,
+				       struct urb *urb)
+{
+	unsigned char *cp = urb->transfer_buffer;
+	struct snd_urb_ctx *ctx = urb->context;
+
+	urb->dev = ctx->subs->dev; /* we need to set this at each time */
+	urb->iso_frame_desc[0].length = 4;
+	urb->iso_frame_desc[0].offset = 0;
+	cp[0] = subs->freqn;
+	cp[1] = subs->freqn >> 8;
+	cp[2] = subs->freqn >> 16;
+	cp[3] = subs->freqn >> 24;
+	return 0;
+}
+
+/*
+ * process after capture sync complete
+ * - nothing to do
+ */
+static int retire_capture_sync_urb(struct snd_usb_substream *subs,
+				   struct snd_pcm_runtime *runtime,
+				   struct urb *urb)
+{
+	return 0;
+}
+
+/*
+ * prepare urb for capture data pipe
+ *
+ * fill the offset and length of each descriptor.
+ *
+ * we use a temporary buffer to write the captured data.
+ * since the length of written data is determined by host, we cannot
+ * write onto the pcm buffer directly...  the data is thus copied
+ * later at complete callback to the global buffer.
+ */
+static int prepare_capture_urb(struct snd_usb_substream *subs,
+			       struct snd_pcm_runtime *runtime,
+			       struct urb *urb)
+{
+	int i, offs;
+	struct snd_urb_ctx *ctx = urb->context;
+
+	offs = 0;
+	urb->dev = ctx->subs->dev; /* we need to set this at each time */
+	for (i = 0; i < ctx->packets; i++) {
+		urb->iso_frame_desc[i].offset = offs;
+		urb->iso_frame_desc[i].length = subs->curpacksize;
+		offs += subs->curpacksize;
+	}
+	urb->transfer_buffer_length = offs;
+	urb->number_of_packets = ctx->packets;
+	return 0;
+}
+
+/*
+ * process after capture complete
+ *
+ * copy the data from each desctiptor to the pcm buffer, and
+ * update the current position.
+ */
+static int retire_capture_urb(struct snd_usb_substream *subs,
+			      struct snd_pcm_runtime *runtime,
+			      struct urb *urb)
+{
+	unsigned long flags;
+	unsigned char *cp;
+	int i;
+	unsigned int stride, frames, bytes, oldptr;
+	int period_elapsed = 0;
+
+	stride = runtime->frame_bits >> 3;
+
+	for (i = 0; i < urb->number_of_packets; i++) {
+		cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
+		if (urb->iso_frame_desc[i].status) {
+			snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
+			// continue;
+		}
+		bytes = urb->iso_frame_desc[i].actual_length;
+		frames = bytes / stride;
+		if (!subs->txfr_quirk)
+			bytes = frames * stride;
+		if (bytes % (runtime->sample_bits >> 3) != 0) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+			int oldbytes = bytes;
+#endif
+			bytes = frames * stride;
+			snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
+							oldbytes, bytes);
+		}
+		/* update the current pointer */
+		spin_lock_irqsave(&subs->lock, flags);
+		oldptr = subs->hwptr_done;
+		subs->hwptr_done += bytes;
+		if (subs->hwptr_done >= runtime->buffer_size * stride)
+			subs->hwptr_done -= runtime->buffer_size * stride;
+		frames = (bytes + (oldptr % stride)) / stride;
+		subs->transfer_done += frames;
+		if (subs->transfer_done >= runtime->period_size) {
+			subs->transfer_done -= runtime->period_size;
+			period_elapsed = 1;
+		}
+		spin_unlock_irqrestore(&subs->lock, flags);
+		/* copy a data chunk */
+		if (oldptr + bytes > runtime->buffer_size * stride) {
+			unsigned int bytes1 =
+					runtime->buffer_size * stride - oldptr;
+			memcpy(runtime->dma_area + oldptr, cp, bytes1);
+			memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
+		} else {
+			memcpy(runtime->dma_area + oldptr, cp, bytes);
+		}
+	}
+	if (period_elapsed)
+		snd_pcm_period_elapsed(subs->pcm_substream);
+	return 0;
+}
+
+/*
+ * Process after capture complete when paused.  Nothing to do.
+ */
+static int retire_paused_capture_urb(struct snd_usb_substream *subs,
+				     struct snd_pcm_runtime *runtime,
+				     struct urb *urb)
+{
+	return 0;
+}
+
+
+/*
+ * prepare urb for playback sync pipe
+ *
+ * set up the offset and length to receive the current frequency.
+ */
+static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
+				     struct snd_pcm_runtime *runtime,
+				     struct urb *urb)
+{
+	struct snd_urb_ctx *ctx = urb->context;
+
+	urb->dev = ctx->subs->dev; /* we need to set this at each time */
+	urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize);
+	urb->iso_frame_desc[0].offset = 0;
+	return 0;
+}
+
+/*
+ * process after playback sync complete
+ *
+ * Full speed devices report feedback values in 10.14 format as samples per
+ * frame, high speed devices in 16.16 format as samples per microframe.
+ * Because the Audio Class 1 spec was written before USB 2.0, many high speed
+ * devices use a wrong interpretation, some others use an entirely different
+ * format.  Therefore, we cannot predict what format any particular device uses
+ * and must detect it automatically.
+ */
+static int retire_playback_sync_urb(struct snd_usb_substream *subs,
+				    struct snd_pcm_runtime *runtime,
+				    struct urb *urb)
+{
+	unsigned int f;
+	int shift;
+	unsigned long flags;
+
+	if (urb->iso_frame_desc[0].status != 0 ||
+	    urb->iso_frame_desc[0].actual_length < 3)
+		return 0;
+
+	f = le32_to_cpup(urb->transfer_buffer);
+	if (urb->iso_frame_desc[0].actual_length == 3)
+		f &= 0x00ffffff;
+	else
+		f &= 0x0fffffff;
+	if (f == 0)
+		return 0;
+
+	if (unlikely(subs->freqshift == INT_MIN)) {
+		/*
+		 * The first time we see a feedback value, determine its format
+		 * by shifting it left or right until it matches the nominal
+		 * frequency value.  This assumes that the feedback does not
+		 * differ from the nominal value more than +50% or -25%.
+		 */
+		shift = 0;
+		while (f < subs->freqn - subs->freqn / 4) {
+			f <<= 1;
+			shift++;
+		}
+		while (f > subs->freqn + subs->freqn / 2) {
+			f >>= 1;
+			shift--;
+		}
+		subs->freqshift = shift;
+	}
+	else if (subs->freqshift >= 0)
+		f <<= subs->freqshift;
+	else
+		f >>= -subs->freqshift;
+
+	if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) {
+		/*
+		 * If the frequency looks valid, set it.
+		 * This value is referred to in prepare_playback_urb().
+		 */
+		spin_lock_irqsave(&subs->lock, flags);
+		subs->freqm = f;
+		spin_unlock_irqrestore(&subs->lock, flags);
+	} else {
+		/*
+		 * Out of range; maybe the shift value is wrong.
+		 * Reset it so that we autodetect again the next time.
+		 */
+		subs->freqshift = INT_MIN;
+	}
 
 	return 0;
 }
 
-static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
-					 struct usb_host_interface *alts,
-					 int protocol, int iface_no)
+/* determine the number of frames in the next packet */
+static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
 {
-	/* parsed with a v1 header here. that's ok as we only look at the
-	 * header first which is the same for both versions */
-	struct uac_iso_endpoint_descriptor *csep;
-	struct usb_interface_descriptor *altsd = get_iface_desc(alts);
-	int attributes = 0;
+	if (subs->fill_max)
+		return subs->maxframesize;
+	else {
+		subs->phase = (subs->phase & 0xffff)
+			+ (subs->freqm << subs->datainterval);
+		return min(subs->phase >> 16, subs->maxframesize);
+	}
+}
 
-	csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+/*
+ * Prepare urb for streaming before playback starts or when paused.
+ *
+ * We don't have any data, so we send silence.
+ */
+static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
+				       struct snd_pcm_runtime *runtime,
+				       struct urb *urb)
+{
+	unsigned int i, offs, counts;
+	struct snd_urb_ctx *ctx = urb->context;
+	int stride = runtime->frame_bits >> 3;
 
-	/* Creamware Noah has this descriptor after the 2nd endpoint */
-	if (!csep && altsd->bNumEndpoints >= 2)
-		csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
+	offs = 0;
+	urb->dev = ctx->subs->dev;
+	for (i = 0; i < ctx->packets; ++i) {
+		counts = snd_usb_audio_next_packet_size(subs);
+		urb->iso_frame_desc[i].offset = offs * stride;
+		urb->iso_frame_desc[i].length = counts * stride;
+		offs += counts;
+	}
+	urb->number_of_packets = ctx->packets;
+	urb->transfer_buffer_length = offs * stride;
+	memset(urb->transfer_buffer,
+	       runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
+	       offs * stride);
+	return 0;
+}
 
-	if (!csep || csep->bLength < 7 ||
-	    csep->bDescriptorSubtype != UAC_EP_GENERAL) {
-		snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
-			   " class specific endpoint descriptor\n",
-			   chip->dev->devnum, iface_no,
-			   altsd->bAlternateSetting);
+/*
+ * prepare urb for playback data pipe
+ *
+ * Since a URB can handle only a single linear buffer, we must use double
+ * buffering when the data to be transferred overflows the buffer boundary.
+ * To avoid inconsistencies when updating hwptr_done, we use double buffering
+ * for all URBs.
+ */
+static int prepare_playback_urb(struct snd_usb_substream *subs,
+				struct snd_pcm_runtime *runtime,
+				struct urb *urb)
+{
+	int i, stride;
+	unsigned int counts, frames, bytes;
+	unsigned long flags;
+	int period_elapsed = 0;
+	struct snd_urb_ctx *ctx = urb->context;
+
+	stride = runtime->frame_bits >> 3;
+
+	frames = 0;
+	urb->dev = ctx->subs->dev; /* we need to set this at each time */
+	urb->number_of_packets = 0;
+	spin_lock_irqsave(&subs->lock, flags);
+	for (i = 0; i < ctx->packets; i++) {
+		counts = snd_usb_audio_next_packet_size(subs);
+		/* set up descriptor */
+		urb->iso_frame_desc[i].offset = frames * stride;
+		urb->iso_frame_desc[i].length = counts * stride;
+		frames += counts;
+		urb->number_of_packets++;
+		subs->transfer_done += counts;
+		if (subs->transfer_done >= runtime->period_size) {
+			subs->transfer_done -= runtime->period_size;
+			period_elapsed = 1;
+			if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
+				if (subs->transfer_done > 0) {
+					/* FIXME: fill-max mode is not
+					 * supported yet */
+					frames -= subs->transfer_done;
+					counts -= subs->transfer_done;
+					urb->iso_frame_desc[i].length =
+						counts * stride;
+					subs->transfer_done = 0;
+				}
+				i++;
+				if (i < ctx->packets) {
+					/* add a transfer delimiter */
+					urb->iso_frame_desc[i].offset =
+						frames * stride;
+					urb->iso_frame_desc[i].length = 0;
+					urb->number_of_packets++;
+				}
+				break;
+			}
+		}
+		if (period_elapsed) /* finish at the period boundary */
+			break;
+	}
+	bytes = frames * stride;
+	if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
+		/* err, the transferred area goes over buffer boundary. */
+		unsigned int bytes1 =
+			runtime->buffer_size * stride - subs->hwptr_done;
+		memcpy(urb->transfer_buffer,
+		       runtime->dma_area + subs->hwptr_done, bytes1);
+		memcpy(urb->transfer_buffer + bytes1,
+		       runtime->dma_area, bytes - bytes1);
+	} else {
+		memcpy(urb->transfer_buffer,
+		       runtime->dma_area + subs->hwptr_done, bytes);
+	}
+	subs->hwptr_done += bytes;
+	if (subs->hwptr_done >= runtime->buffer_size * stride)
+		subs->hwptr_done -= runtime->buffer_size * stride;
+
+	/* update delay with exact number of samples queued */
+	runtime->delay = subs->last_delay;
+	runtime->delay += frames;
+	subs->last_delay = runtime->delay;
+
+	/* realign last_frame_number */
+	subs->last_frame_number = usb_get_current_frame_number(subs->dev);
+	subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
+
+	spin_unlock_irqrestore(&subs->lock, flags);
+	urb->transfer_buffer_length = bytes;
+	if (period_elapsed)
+		snd_pcm_period_elapsed(subs->pcm_substream);
+	return 0;
+}
+
+/*
+ * process after playback data complete
+ * - decrease the delay count again
+ */
+static int retire_playback_urb(struct snd_usb_substream *subs,
+			       struct snd_pcm_runtime *runtime,
+			       struct urb *urb)
+{
+	unsigned long flags;
+	int stride = runtime->frame_bits >> 3;
+	int processed = urb->transfer_buffer_length / stride;
+	int est_delay;
+
+	spin_lock_irqsave(&subs->lock, flags);
+
+	est_delay = snd_usb_pcm_delay(subs, runtime->rate);
+	/* update delay with exact number of samples played */
+	if (processed > subs->last_delay)
+		subs->last_delay = 0;
+	else
+		subs->last_delay -= processed;
+	runtime->delay = subs->last_delay;
+
+	/*
+	 * Report when delay estimate is off by more than 2ms.
+	 * The error should be lower than 2ms since the estimate relies
+	 * on two reads of a counter updated every ms.
+	 */
+	if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+		snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
+			est_delay, subs->last_delay);
+
+	spin_unlock_irqrestore(&subs->lock, flags);
+	return 0;
+}
+
+static const char *usb_error_string(int err)
+{
+	switch (err) {
+	case -ENODEV:
+		return "no device";
+	case -ENOENT:
+		return "endpoint not enabled";
+	case -EPIPE:
+		return "endpoint stalled";
+	case -ENOSPC:
+		return "not enough bandwidth";
+	case -ESHUTDOWN:
+		return "device disabled";
+	case -EHOSTUNREACH:
+		return "device suspended";
+	case -EINVAL:
+	case -EAGAIN:
+	case -EFBIG:
+	case -EMSGSIZE:
+		return "internal error";
+	default:
+		return "unknown error";
+	}
+}
+
+/*
+ * set up and start data/sync urbs
+ */
+static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
+{
+	unsigned int i;
+	int err;
+
+	if (subs->stream->chip->shutdown)
+		return -EBADFD;
+
+	for (i = 0; i < subs->nurbs; i++) {
+		if (snd_BUG_ON(!subs->dataurb[i].urb))
+			return -EINVAL;
+		if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
+			snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
+			goto __error;
+		}
+	}
+	if (subs->syncpipe) {
+		for (i = 0; i < SYNC_URBS; i++) {
+			if (snd_BUG_ON(!subs->syncurb[i].urb))
+				return -EINVAL;
+			if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
+				snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
+				goto __error;
+			}
+		}
+	}
+
+	subs->active_mask = 0;
+	subs->unlink_mask = 0;
+	subs->running = 1;
+	for (i = 0; i < subs->nurbs; i++) {
+		err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
+		if (err < 0) {
+			snd_printk(KERN_ERR "cannot submit datapipe "
+				   "for urb %d, error %d: %s\n",
+				   i, err, usb_error_string(err));
+			goto __error;
+		}
+		set_bit(i, &subs->active_mask);
+	}
+	if (subs->syncpipe) {
+		for (i = 0; i < SYNC_URBS; i++) {
+			err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
+			if (err < 0) {
+				snd_printk(KERN_ERR "cannot submit syncpipe "
+					   "for urb %d, error %d: %s\n",
+					   i, err, usb_error_string(err));
+				goto __error;
+			}
+			set_bit(i + 16, &subs->active_mask);
+		}
+	}
+	return 0;
+
+ __error:
+	// snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
+	deactivate_urbs(subs, 0, 0);
+	return -EPIPE;
+}
+
+
+/*
+ */
+static struct snd_urb_ops audio_urb_ops[2] = {
+	{
+		.prepare =	prepare_nodata_playback_urb,
+		.retire =	retire_playback_urb,
+		.prepare_sync =	prepare_playback_sync_urb,
+		.retire_sync =	retire_playback_sync_urb,
+	},
+	{
+		.prepare =	prepare_capture_urb,
+		.retire =	retire_capture_urb,
+		.prepare_sync =	prepare_capture_sync_urb,
+		.retire_sync =	retire_capture_sync_urb,
+	},
+};
+
+/*
+ * initialize the substream instance.
+ */
+
+void snd_usb_init_substream(struct snd_usb_stream *as,
+			    int stream, struct audioformat *fp)
+{
+	struct snd_usb_substream *subs = &as->substream[stream];
+
+	INIT_LIST_HEAD(&subs->fmt_list);
+	spin_lock_init(&subs->lock);
+
+	subs->stream = as;
+	subs->direction = stream;
+	subs->dev = as->chip->dev;
+	subs->txfr_quirk = as->chip->txfr_quirk;
+	subs->ops = audio_urb_ops[stream];
+	if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH)
+		subs->ops.prepare_sync = prepare_capture_sync_urb_hs;
+
+	snd_usb_set_pcm_ops(as->pcm, stream);
+
+	list_add_tail(&fp->list, &subs->fmt_list);
+	subs->formats |= fp->formats;
+	subs->endpoint = fp->endpoint;
+	subs->num_formats++;
+	subs->fmt_type = fp->fmt_type;
+}
+
+int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_usb_substream *subs = substream->runtime->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		subs->ops.prepare = prepare_playback_urb;
+		return 0;
+	case SNDRV_PCM_TRIGGER_STOP:
+		return deactivate_urbs(subs, 0, 0);
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		subs->ops.prepare = prepare_nodata_playback_urb;
 		return 0;
 	}
 
-	if (protocol == UAC_VERSION_1) {
-		attributes = csep->bmAttributes;
-	} else {
-		struct uac2_iso_endpoint_descriptor *csep2 =
-			(struct uac2_iso_endpoint_descriptor *) csep;
-
-		attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
-
-		/* emulate the endpoint attributes of a v1 device */
-		if (csep2->bmControls & UAC2_CONTROL_PITCH)
-			attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
-	}
-
-	return attributes;
+	return -EINVAL;
 }
 
-static struct uac2_input_terminal_descriptor *
-	snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-					       int terminal_id)
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 {
-	struct uac2_input_terminal_descriptor *term = NULL;
+	struct snd_usb_substream *subs = substream->runtime->private_data;
 
-	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
-					       ctrl_iface->extralen,
-					       term, UAC_INPUT_TERMINAL))) {
-		if (term->bTerminalID == terminal_id)
-			return term;
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		subs->ops.retire = retire_capture_urb;
+		return start_urbs(subs, substream->runtime);
+	case SNDRV_PCM_TRIGGER_STOP:
+		return deactivate_urbs(subs, 0, 0);
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		subs->ops.retire = retire_paused_capture_urb;
+		return 0;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		subs->ops.retire = retire_capture_urb;
+		return 0;
 	}
 
-	return NULL;
+	return -EINVAL;
 }
 
-static struct uac2_output_terminal_descriptor *
-	snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-						int terminal_id)
+int snd_usb_substream_prepare(struct snd_usb_substream *subs,
+			      struct snd_pcm_runtime *runtime)
 {
-	struct uac2_output_terminal_descriptor *term = NULL;
+	/* clear urbs (to be sure) */
+	deactivate_urbs(subs, 0, 1);
+	wait_clear_urbs(subs);
 
-	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
-					       ctrl_iface->extralen,
-					       term, UAC_OUTPUT_TERMINAL))) {
-		if (term->bTerminalID == terminal_id)
-			return term;
+	/* for playback, submit the URBs now; otherwise, the first hwptr_done
+	 * updates for all URBs would happen at the same time when starting */
+	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+		subs->ops.prepare = prepare_nodata_playback_urb;
+		return start_urbs(subs, runtime);
 	}
 
-	return NULL;
-}
-
-int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
-{
-	struct usb_device *dev;
-	struct usb_interface *iface;
-	struct usb_host_interface *alts;
-	struct usb_interface_descriptor *altsd;
-	int i, altno, err, stream;
-	int format = 0, num_channels = 0;
-	struct audioformat *fp = NULL;
-	int num, protocol, clock = 0;
-	struct uac_format_type_i_continuous_descriptor *fmt;
-
-	dev = chip->dev;
-
-	/* parse the interface's altsettings */
-	iface = usb_ifnum_to_if(dev, iface_no);
-
-	num = iface->num_altsetting;
-
-	/*
-	 * Dallas DS4201 workaround: It presents 5 altsettings, but the last
-	 * one misses syncpipe, and does not produce any sound.
-	 */
-	if (chip->usb_id == USB_ID(0x04fa, 0x4201))
-		num = 4;
-
-	for (i = 0; i < num; i++) {
-		alts = &iface->altsetting[i];
-		altsd = get_iface_desc(alts);
-		protocol = altsd->bInterfaceProtocol;
-		/* skip invalid one */
-		if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
-		     altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
-		    (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING &&
-		     altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) ||
-		    altsd->bNumEndpoints < 1 ||
-		    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0)
-			continue;
-		/* must be isochronous */
-		if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) !=
-		    USB_ENDPOINT_XFER_ISOC)
-			continue;
-		/* check direction */
-		stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
-			SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
-		altno = altsd->bAlternateSetting;
-
-		if (snd_usb_apply_interface_quirk(chip, iface_no, altno))
-			continue;
-
-		/* get audio formats */
-		switch (protocol) {
-		default:
-			snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
-				    dev->devnum, iface_no, altno, protocol);
-			protocol = UAC_VERSION_1;
-			/* fall through */
-
-		case UAC_VERSION_1: {
-			struct uac1_as_header_descriptor *as =
-				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-
-			if (!as) {
-				snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
-					   dev->devnum, iface_no, altno);
-				continue;
-			}
-
-			if (as->bLength < sizeof(*as)) {
-				snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
-					   dev->devnum, iface_no, altno);
-				continue;
-			}
-
-			format = le16_to_cpu(as->wFormatTag); /* remember the format value */
-			break;
-		}
-
-		case UAC_VERSION_2: {
-			struct uac2_input_terminal_descriptor *input_term;
-			struct uac2_output_terminal_descriptor *output_term;
-			struct uac2_as_header_descriptor *as =
-				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-
-			if (!as) {
-				snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
-					   dev->devnum, iface_no, altno);
-				continue;
-			}
-
-			if (as->bLength < sizeof(*as)) {
-				snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
-					   dev->devnum, iface_no, altno);
-				continue;
-			}
-
-			num_channels = as->bNrChannels;
-			format = le32_to_cpu(as->bmFormats);
-
-			/* lookup the terminal associated to this interface
-			 * to extract the clock */
-			input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
-									    as->bTerminalLink);
-			if (input_term) {
-				clock = input_term->bCSourceID;
-				break;
-			}
-
-			output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-									      as->bTerminalLink);
-			if (output_term) {
-				clock = output_term->bCSourceID;
-				break;
-			}
-
-			snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
-				   dev->devnum, iface_no, altno, as->bTerminalLink);
-			continue;
-		}
-		}
-
-		/* get format type */
-		fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE);
-		if (!fmt) {
-			snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n",
-				   dev->devnum, iface_no, altno);
-			continue;
-		}
-		if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
-		    ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
-			snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
-				   dev->devnum, iface_no, altno);
-			continue;
-		}
-
-		/*
-		 * Blue Microphones workaround: The last altsetting is identical
-		 * with the previous one, except for a larger packet size, but
-		 * is actually a mislabeled two-channel setting; ignore it.
-		 */
-		if (fmt->bNrChannels == 1 &&
-		    fmt->bSubframeSize == 2 &&
-		    altno == 2 && num == 3 &&
-		    fp && fp->altsetting == 1 && fp->channels == 1 &&
-		    fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
-		    protocol == UAC_VERSION_1 &&
-		    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
-							fp->maxpacksize * 2)
-			continue;
-
-		fp = kzalloc(sizeof(*fp), GFP_KERNEL);
-		if (! fp) {
-			snd_printk(KERN_ERR "cannot malloc\n");
-			return -ENOMEM;
-		}
-
-		fp->iface = iface_no;
-		fp->altsetting = altno;
-		fp->altset_idx = i;
-		fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
-		fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
-		fp->datainterval = snd_usb_parse_datainterval(chip, alts);
-		fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
-		/* num_channels is only set for v2 interfaces */
-		fp->channels = num_channels;
-		if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
-			fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
-					* (fp->maxpacksize & 0x7ff);
-		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
-		fp->clock = clock;
-
-		/* some quirks for attributes here */
-
-		switch (chip->usb_id) {
-		case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */
-			/* Optoplay sets the sample rate attribute although
-			 * it seems not supporting it in fact.
-			 */
-			fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE;
-			break;
-		case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
-		case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
-			/* doesn't set the sample rate attribute, but supports it */
-			fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE;
-			break;
-		case USB_ID(0x0763, 0x2001):  /* M-Audio Quattro USB */
-		case USB_ID(0x0763, 0x2012):  /* M-Audio Fast Track Pro USB */
-		case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
-		case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
-						an older model 77d:223) */
-		/*
-		 * plantronics headset and Griffin iMic have set adaptive-in
-		 * although it's really not...
-		 */
-			fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
-			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-				fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE;
-			else
-				fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
-			break;
-		}
-
-		/* ok, let's parse further... */
-		if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
-			kfree(fp->rate_table);
-			kfree(fp);
-			fp = NULL;
-			continue;
-		}
-
-		snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint);
-		err = snd_usb_add_audio_endpoint(chip, stream, fp);
-		if (err < 0) {
-			kfree(fp->rate_table);
-			kfree(fp);
-			return err;
-		}
-		/* try to set the interface... */
-		usb_set_interface(chip->dev, iface_no, altno);
-		snd_usb_init_pitch(chip, iface_no, alts, fp);
-		snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max);
-	}
 	return 0;
 }
 
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 64dd0db..88eb63a 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -1,11 +1,21 @@
 #ifndef __USBAUDIO_ENDPOINT_H
 #define __USBAUDIO_ENDPOINT_H
 
-int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip,
-				  int iface_no);
+void snd_usb_init_substream(struct snd_usb_stream *as,
+			    int stream,
+			    struct audioformat *fp);
 
-int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip,
-			       int stream,
-			       struct audioformat *fp);
+int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
+				unsigned int period_bytes,
+				unsigned int rate,
+				unsigned int frame_bits);
+
+void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force);
+
+int snd_usb_substream_prepare(struct snd_usb_substream *subs,
+			      struct snd_pcm_runtime *runtime);
+
+int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd);
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd);
 
 #endif /* __USBAUDIO_ENDPOINT_H */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 8d042dc..89421d1 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -286,7 +286,7 @@
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
 			      UAC2_CS_CONTROL_SAM_FREQ << 8,
 			      snd_usb_ctrl_intf(chip) | (clock << 8),
-			      tmp, sizeof(tmp), 1000);
+			      tmp, sizeof(tmp));
 
 	if (ret < 0) {
 		snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n",
@@ -307,7 +307,7 @@
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
 			      UAC2_CS_CONTROL_SAM_FREQ << 8,
 			      snd_usb_ctrl_intf(chip) | (clock << 8),
-			      data, data_size, 1000);
+			      data, data_size);
 
 	if (ret < 0) {
 		snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n",
diff --git a/sound/usb/helper.c b/sound/usb/helper.c
index f280c19..9eed8f4 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -81,7 +81,7 @@
  */
 int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request,
 		    __u8 requesttype, __u16 value, __u16 index, void *data,
-		    __u16 size, int timeout)
+		    __u16 size)
 {
 	int err;
 	void *buf = NULL;
@@ -92,7 +92,7 @@
 			return -ENOMEM;
 	}
 	err = usb_control_msg(dev, pipe, request, requesttype,
-			      value, index, buf, size, timeout);
+			      value, index, buf, size, 1000);
 	if (size > 0) {
 		memcpy(data, buf, size);
 		kfree(buf);
diff --git a/sound/usb/helper.h b/sound/usb/helper.h
index 09bd943..805c300 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -8,7 +8,7 @@
 
 int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe,
 		    __u8 request, __u8 requesttype, __u16 value, __u16 index,
-		    void *data, __u16 size, int timeout);
+		    void *data, __u16 size);
 
 unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
 					 struct usb_host_interface *alts);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index f928910..e21f026 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -816,6 +816,22 @@
 	.output = snd_usbmidi_raw_output,
 };
 
+/*
+ * FTDI protocol: raw MIDI bytes, but input packets have two modem status bytes.
+ */
+
+static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint* ep,
+				   uint8_t* buffer, int buffer_length)
+{
+	if (buffer_length > 2)
+		snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2);
+}
+
+static struct usb_protocol_ops snd_usbmidi_ftdi_ops = {
+	.input = snd_usbmidi_ftdi_input,
+	.output = snd_usbmidi_raw_output,
+};
+
 static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep,
 				     uint8_t *buffer, int buffer_length)
 {
@@ -2163,6 +2179,17 @@
 		/* endpoint 1 is input-only */
 		endpoints[1].out_cables = 0;
 		break;
+	case QUIRK_MIDI_FTDI:
+		umidi->usb_protocol_ops = &snd_usbmidi_ftdi_ops;
+
+		/* set baud rate to 31250 (48 MHz / 16 / 96) */
+		err = usb_control_msg(umidi->dev, usb_sndctrlpipe(umidi->dev, 0),
+				      3, 0x40, 0x60, 0, NULL, 0, 1000);
+		if (err < 0)
+			break;
+
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		break;
 	default:
 		snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
 		err = -ENXIO;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c22fa76..60f65ac 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@
 	if (p && p->dB) {
 		cval->dBmin = p->dB->min;
 		cval->dBmax = p->dB->max;
+		cval->initialized = 1;
 	}
 }
 
@@ -295,7 +296,7 @@
 		if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request,
 				    USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 				    validx, snd_usb_ctrl_intf(chip) | (cval->id << 8),
-				    buf, val_len, 100) >= val_len) {
+				    buf, val_len) >= val_len) {
 			*value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len));
 			snd_usb_autosuspend(cval->mixer->chip);
 			return 0;
@@ -332,7 +333,7 @@
 	ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest,
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 			      validx, snd_usb_ctrl_intf(chip) | (cval->id << 8),
-			      buf, size, 1000);
+			      buf, size);
 	snd_usb_autosuspend(chip);
 
 	if (ret < 0) {
@@ -444,7 +445,7 @@
 				    usb_sndctrlpipe(chip->dev, 0), request,
 				    USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
 				    validx, snd_usb_ctrl_intf(chip) | (cval->id << 8),
-				    buf, val_len, 100) >= 0) {
+				    buf, val_len) >= 0) {
 			snd_usb_autosuspend(chip);
 			return 0;
 		}
@@ -880,8 +881,17 @@
 		uinfo->value.integer.min = 0;
 		uinfo->value.integer.max = 1;
 	} else {
-		if (! cval->initialized)
-			get_min_max(cval,  0);
+		if (!cval->initialized) {
+			get_min_max(cval, 0);
+			if (cval->initialized && cval->dBmin >= cval->dBmax) {
+				kcontrol->vd[0].access &= 
+					~(SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+					  SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK);
+				snd_ctl_notify(cval->mixer->chip->card,
+					       SNDRV_CTL_EVENT_MASK_INFO,
+					       &kcontrol->id);
+			}
+		}
 		uinfo->value.integer.min = 0;
 		uinfo->value.integer.max =
 			(cval->max - cval->min + cval->res - 1) / cval->res;
@@ -1092,7 +1102,7 @@
 				" Switch" : " Volume");
 		if (control == UAC_FU_VOLUME) {
 			check_mapped_dB(map, cval);
-			if (cval->dBmin < cval->dBmax) {
+			if (cval->dBmin < cval->dBmax || !cval->initialized) {
 				kctl->tlv.c = mixer_vol_tlv;
 				kctl->vd[0].access |= 
 					SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1191,6 +1201,11 @@
 
 	if (state->mixer->protocol == UAC_VERSION_1) {
 		csize = hdr->bControlSize;
+		if (!csize) {
+			snd_printdd(KERN_ERR "usbaudio: unit %u: "
+				    "invalid bControlSize == 0\n", unitid);
+			return -EINVAL;
+		}
 		channels = (hdr->bLength - 7) / csize - 1;
 		bmaControls = hdr->bmaControls;
 	} else {
@@ -1244,7 +1259,7 @@
 				build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0);
 		}
 	} else { /* UAC_VERSION_2 */
-		for (i = 0; i < 30/2; i++) {
+		for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) {
 			unsigned int ch_bits = 0;
 			unsigned int ch_read_only = 0;
 
@@ -1934,15 +1949,13 @@
 	struct mixer_build state;
 	int err;
 	const struct usbmix_ctl_map *map;
-	struct usb_host_interface *hostif;
 	void *p;
 
-	hostif = mixer->chip->ctrl_intf;
 	memset(&state, 0, sizeof(state));
 	state.chip = mixer->chip;
 	state.mixer = mixer;
-	state.buffer = hostif->extra;
-	state.buflen = hostif->extralen;
+	state.buffer = mixer->hostif->extra;
+	state.buflen = mixer->hostif->extralen;
 
 	/* check the mapping table */
 	for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1955,7 +1968,8 @@
 	}
 
 	p = NULL;
-	while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
+	while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
+					    p, UAC_OUTPUT_TERMINAL)) != NULL) {
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
@@ -2162,17 +2176,15 @@
 /* create the handler for the optional status interrupt endpoint */
 static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 {
-	struct usb_host_interface *hostif;
 	struct usb_endpoint_descriptor *ep;
 	void *transfer_buffer;
 	int buffer_length;
 	unsigned int epnum;
 
-	hostif = mixer->chip->ctrl_intf;
 	/* we need one interrupt input endpoint */
-	if (get_iface_desc(hostif)->bNumEndpoints < 1)
+	if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
 		return 0;
-	ep = get_endpoint(hostif, 0);
+	ep = get_endpoint(mixer->hostif, 0);
 	if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
 		return 0;
 
@@ -2202,7 +2214,6 @@
 	};
 	struct usb_mixer_interface *mixer;
 	struct snd_info_entry *entry;
-	struct usb_host_interface *host_iface;
 	int err;
 
 	strcpy(chip->card->mixername, "USB Mixer");
@@ -2219,8 +2230,8 @@
 		return -ENOMEM;
 	}
 
-	host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
-	switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+	mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+	switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
 	case UAC_VERSION_1:
 	default:
 		mixer->protocol = UAC_VERSION_1;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ae1a14d..81b2d8a 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,6 +3,7 @@
 
 struct usb_mixer_interface {
 	struct snd_usb_audio *chip;
+	struct usb_host_interface *hostif;
 	struct list_head list;
 	unsigned int ignore_ctl_error;
 	struct urb *urb;
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 3d0f487..ab125ee 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -190,18 +190,18 @@
 		err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
-			      !value, 0, NULL, 0, 100);
+			      !value, 0, NULL, 0);
 	/* USB X-Fi S51 Pro */
 	if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df))
 		err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
-			      !value, 0, NULL, 0, 100);
+			      !value, 0, NULL, 0);
 	else
 		err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
-			      value, index + 2, NULL, 0, 100);
+			      value, index + 2, NULL, 0);
 	if (err < 0)
 		return err;
 	mixer->audigy2nx_leds[index] = value;
@@ -299,7 +299,7 @@
 				      usb_rcvctrlpipe(mixer->chip->dev, 0),
 				      UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS |
 				      USB_RECIP_INTERFACE, 0,
-				      jacks[i].unitid << 8, buf, 3, 100);
+				      jacks[i].unitid << 8, buf, 3);
 		if (err == 3 && (buf[0] == 3 || buf[0] == 6))
 			snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]);
 		else
@@ -332,7 +332,7 @@
 	err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x08,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
-			      50, 0, &new_status, 1, 100);
+			      50, 0, &new_status, 1);
 	if (err < 0)
 		return err;
 	mixer->xonar_u1_status = new_status;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index b8dcbf4..0220b0f 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -28,12 +28,36 @@
 #include "card.h"
 #include "quirks.h"
 #include "debug.h"
-#include "urb.h"
+#include "endpoint.h"
 #include "helper.h"
 #include "pcm.h"
 #include "clock.h"
 #include "power.h"
 
+/* return the estimated delay based on USB frame counters */
+snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
+				    unsigned int rate)
+{
+	int current_frame_number;
+	int frame_diff;
+	int est_delay;
+
+	current_frame_number = usb_get_current_frame_number(subs->dev);
+	/*
+	 * HCD implementations use different widths, use lower 8 bits.
+	 * The delay will be managed up to 256ms, which is more than
+	 * enough
+	 */
+	frame_diff = (current_frame_number - subs->last_frame_number) & 0xff;
+
+	/* Approximation based on number of samples per USB frame (ms),
+	   some truncation for 44.1 but the estimate is good enough */
+	est_delay =  subs->last_delay - (frame_diff * rate / 1000);
+	if (est_delay < 0)
+		est_delay = 0;
+	return est_delay;
+}
+
 /*
  * return the current pcm pointer.  just based on the hwptr_done value.
  */
@@ -45,6 +69,8 @@
 	subs = (struct snd_usb_substream *)substream->runtime->private_data;
 	spin_lock(&subs->lock);
 	hwptr_done = subs->hwptr_done;
+	substream->runtime->delay = snd_usb_pcm_delay(subs,
+						substream->runtime->rate);
 	spin_unlock(&subs->lock);
 	return hwptr_done / (substream->runtime->frame_bits >> 3);
 }
@@ -126,7 +152,7 @@
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
 				   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
 				   UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep,
-				   data, sizeof(data), 1000)) < 0) {
+				   data, sizeof(data))) < 0) {
 		snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n",
 			   dev->devnum, iface, ep);
 		return err;
@@ -150,7 +176,7 @@
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
 				   UAC2_EP_CS_PITCH << 8, 0,
-				   data, sizeof(data), 1000)) < 0) {
+				   data, sizeof(data))) < 0) {
 		snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);
 		return err;
@@ -417,6 +443,8 @@
 	subs->hwptr_done = 0;
 	subs->transfer_done = 0;
 	subs->phase = 0;
+	subs->last_delay = 0;
+	subs->last_frame_number = 0;
 	runtime->delay = 0;
 
 	return snd_usb_substream_prepare(subs, runtime);
diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h
index ed3e283..df7a003 100644
--- a/sound/usb/pcm.h
+++ b/sound/usb/pcm.h
@@ -1,6 +1,9 @@
 #ifndef __USBAUDIO_PCM_H
 #define __USBAUDIO_PCM_H
 
+snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
+				    unsigned int rate);
+
 void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream);
 
 int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index dba0b7f..b61945f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -39,6 +39,17 @@
 	.idProduct = prod, \
 	.bInterfaceClass = USB_CLASS_VENDOR_SPEC
 
+/* FTDI devices */
+{
+	USB_DEVICE(0x0403, 0xb8d8),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "STARR LABS", */
+		/* .product_name = "Starr Labs MIDI USB device", */
+		.ifnum = 0,
+		.type = QUIRK_MIDI_FTDI
+	}
+},
+
 /* Creative/Toshiba Multimedia Center SB-0500 */
 {
 	USB_DEVICE(0x041e, 0x3048),
@@ -1678,6 +1689,20 @@
 	}
 },
 {
+	/* Added support for Roland UM-ONE which differs from UM-1 */
+	USB_DEVICE(0x0582, 0x012a),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "ROLAND", */
+		/* .product_name = "UM-ONE", */
+		.ifnum = 0,
+		.type = QUIRK_MIDI_FIXED_ENDPOINT,
+		.data = & (const struct snd_usb_midi_endpoint_info) {
+			.out_cables = 0x0001,
+			.in_cables  = 0x0003
+		}
+	}
+},
+{
 	USB_DEVICE(0x0582, 0x011e),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "BOSS", */
@@ -1707,6 +1732,40 @@
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0130),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "BOSS", */
+		/* .product_name = "MICRO BR-80", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables  = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Guillemot devices */
 {
@@ -2417,6 +2476,12 @@
 	.idProduct = 0x1020,
 },
 
+/* KeithMcMillen Stringport */
+{
+	USB_DEVICE(0x1f38, 0x0001),
+	.bInterfaceClass = USB_CLASS_AUDIO,
+},
+
 /* Miditech devices */
 {
 	USB_DEVICE(0x4752, 0x0011),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 77762c9..2e5bc73 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -34,6 +34,7 @@
 #include "endpoint.h"
 #include "pcm.h"
 #include "clock.h"
+#include "stream.h"
 
 /*
  * handle the quirks for the contained interfaces
@@ -106,7 +107,7 @@
 
 	alts = &iface->altsetting[0];
 	altsd = get_iface_desc(alts);
-	err = snd_usb_parse_audio_endpoints(chip, altsd->bInterfaceNumber);
+	err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot setup if %d: error %d\n",
 			   altsd->bInterfaceNumber, err);
@@ -147,7 +148,7 @@
 
 	stream = (fp->endpoint & USB_DIR_IN)
 		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
-	err = snd_usb_add_audio_endpoint(chip, stream, fp);
+	err = snd_usb_add_audio_stream(chip, stream, fp);
 	if (err < 0) {
 		kfree(fp);
 		kfree(rate_table);
@@ -254,7 +255,7 @@
 
 	stream = (fp->endpoint & USB_DIR_IN)
 		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
-	err = snd_usb_add_audio_endpoint(chip, stream, fp);
+	err = snd_usb_add_audio_stream(chip, stream, fp);
 	if (err < 0) {
 		kfree(fp);
 		return err;
@@ -306,6 +307,7 @@
 		[QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
 		[QUIRK_MIDI_CME] = create_any_midi_quirk,
 		[QUIRK_MIDI_AKAI] = create_any_midi_quirk,
+		[QUIRK_MIDI_FTDI] = create_any_midi_quirk,
 		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
 		[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
 		[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
@@ -338,7 +340,7 @@
 		snd_printdd("sending Extigy boot sequence...\n");
 		/* Send message to force it to reconnect with full interface. */
 		err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0),
-				      0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000);
+				      0x10, 0x43, 0x0001, 0x000a, NULL, 0);
 		if (err < 0) snd_printdd("error sending boot message: %d\n", err);
 		err = usb_get_descriptor(dev, USB_DT_DEVICE, 0,
 				&dev->descriptor, sizeof(dev->descriptor));
@@ -359,11 +361,11 @@
 
 	snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a,
 			USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER,
-			0, 0, &buf, 1, 1000);
+			0, 0, &buf, 1);
 	if (buf == 0) {
 		snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29,
 				USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
-				1, 2000, NULL, 0, 1000);
+				1, 2000, NULL, 0);
 		return -ENODEV;
 	}
 	return 0;
@@ -406,7 +408,7 @@
 	buf[3] = reg;
 	return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION,
 			       USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT,
-			       0, 0, &buf, 4, 1000);
+			       0, 0, &buf, 4);
 }
 
 static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
@@ -426,7 +428,7 @@
  */
 static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
 {
-	int err, reg;
+	int err  = 0, reg;
 	int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
 
 	for (reg = 0; reg < ARRAY_SIZE(val); reg++) {
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
new file mode 100644
index 0000000..5ff8010
--- /dev/null
+++ b/sound/usb/stream.c
@@ -0,0 +1,452 @@
+/*
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/usb.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#include "usbaudio.h"
+#include "card.h"
+#include "proc.h"
+#include "quirks.h"
+#include "endpoint.h"
+#include "pcm.h"
+#include "helper.h"
+#include "format.h"
+#include "clock.h"
+#include "stream.h"
+
+/*
+ * free a substream
+ */
+static void free_substream(struct snd_usb_substream *subs)
+{
+	struct list_head *p, *n;
+
+	if (!subs->num_formats)
+		return; /* not initialized */
+	list_for_each_safe(p, n, &subs->fmt_list) {
+		struct audioformat *fp = list_entry(p, struct audioformat, list);
+		kfree(fp->rate_table);
+		kfree(fp);
+	}
+	kfree(subs->rate_list.list);
+}
+
+
+/*
+ * free a usb stream instance
+ */
+static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
+{
+	free_substream(&stream->substream[0]);
+	free_substream(&stream->substream[1]);
+	list_del(&stream->list);
+	kfree(stream);
+}
+
+static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_usb_stream *stream = pcm->private_data;
+	if (stream) {
+		stream->pcm = NULL;
+		snd_usb_audio_stream_free(stream);
+	}
+}
+
+
+/*
+ * add this endpoint to the chip instance.
+ * if a stream with the same endpoint already exists, append to it.
+ * if not, create a new pcm stream.
+ */
+int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
+			     int stream,
+			     struct audioformat *fp)
+{
+	struct list_head *p;
+	struct snd_usb_stream *as;
+	struct snd_usb_substream *subs;
+	struct snd_pcm *pcm;
+	int err;
+
+	list_for_each(p, &chip->pcm_list) {
+		as = list_entry(p, struct snd_usb_stream, list);
+		if (as->fmt_type != fp->fmt_type)
+			continue;
+		subs = &as->substream[stream];
+		if (!subs->endpoint)
+			continue;
+		if (subs->endpoint == fp->endpoint) {
+			list_add_tail(&fp->list, &subs->fmt_list);
+			subs->num_formats++;
+			subs->formats |= fp->formats;
+			return 0;
+		}
+	}
+	/* look for an empty stream */
+	list_for_each(p, &chip->pcm_list) {
+		as = list_entry(p, struct snd_usb_stream, list);
+		if (as->fmt_type != fp->fmt_type)
+			continue;
+		subs = &as->substream[stream];
+		if (subs->endpoint)
+			continue;
+		err = snd_pcm_new_stream(as->pcm, stream, 1);
+		if (err < 0)
+			return err;
+		snd_usb_init_substream(as, stream, fp);
+		return 0;
+	}
+
+	/* create a new pcm */
+	as = kzalloc(sizeof(*as), GFP_KERNEL);
+	if (!as)
+		return -ENOMEM;
+	as->pcm_index = chip->pcm_devs;
+	as->chip = chip;
+	as->fmt_type = fp->fmt_type;
+	err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs,
+			  stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0,
+			  stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1,
+			  &pcm);
+	if (err < 0) {
+		kfree(as);
+		return err;
+	}
+	as->pcm = pcm;
+	pcm->private_data = as;
+	pcm->private_free = snd_usb_audio_pcm_free;
+	pcm->info_flags = 0;
+	if (chip->pcm_devs > 0)
+		sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs);
+	else
+		strcpy(pcm->name, "USB Audio");
+
+	snd_usb_init_substream(as, stream, fp);
+
+	list_add(&as->list, &chip->pcm_list);
+	chip->pcm_devs++;
+
+	snd_usb_proc_pcm_format_add(as);
+
+	return 0;
+}
+
+static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
+					 struct usb_host_interface *alts,
+					 int protocol, int iface_no)
+{
+	/* parsed with a v1 header here. that's ok as we only look at the
+	 * header first which is the same for both versions */
+	struct uac_iso_endpoint_descriptor *csep;
+	struct usb_interface_descriptor *altsd = get_iface_desc(alts);
+	int attributes = 0;
+
+	csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+	/* Creamware Noah has this descriptor after the 2nd endpoint */
+	if (!csep && altsd->bNumEndpoints >= 2)
+		csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+	if (!csep || csep->bLength < 7 ||
+	    csep->bDescriptorSubtype != UAC_EP_GENERAL) {
+		snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
+			   " class specific endpoint descriptor\n",
+			   chip->dev->devnum, iface_no,
+			   altsd->bAlternateSetting);
+		return 0;
+	}
+
+	if (protocol == UAC_VERSION_1) {
+		attributes = csep->bmAttributes;
+	} else {
+		struct uac2_iso_endpoint_descriptor *csep2 =
+			(struct uac2_iso_endpoint_descriptor *) csep;
+
+		attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
+
+		/* emulate the endpoint attributes of a v1 device */
+		if (csep2->bmControls & UAC2_CONTROL_PITCH)
+			attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
+	}
+
+	return attributes;
+}
+
+static struct uac2_input_terminal_descriptor *
+	snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+					       int terminal_id)
+{
+	struct uac2_input_terminal_descriptor *term = NULL;
+
+	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+					       ctrl_iface->extralen,
+					       term, UAC_INPUT_TERMINAL))) {
+		if (term->bTerminalID == terminal_id)
+			return term;
+	}
+
+	return NULL;
+}
+
+static struct uac2_output_terminal_descriptor *
+	snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+						int terminal_id)
+{
+	struct uac2_output_terminal_descriptor *term = NULL;
+
+	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+					       ctrl_iface->extralen,
+					       term, UAC_OUTPUT_TERMINAL))) {
+		if (term->bTerminalID == terminal_id)
+			return term;
+	}
+
+	return NULL;
+}
+
+int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
+{
+	struct usb_device *dev;
+	struct usb_interface *iface;
+	struct usb_host_interface *alts;
+	struct usb_interface_descriptor *altsd;
+	int i, altno, err, stream;
+	int format = 0, num_channels = 0;
+	struct audioformat *fp = NULL;
+	int num, protocol, clock = 0;
+	struct uac_format_type_i_continuous_descriptor *fmt;
+
+	dev = chip->dev;
+
+	/* parse the interface's altsettings */
+	iface = usb_ifnum_to_if(dev, iface_no);
+
+	num = iface->num_altsetting;
+
+	/*
+	 * Dallas DS4201 workaround: It presents 5 altsettings, but the last
+	 * one misses syncpipe, and does not produce any sound.
+	 */
+	if (chip->usb_id == USB_ID(0x04fa, 0x4201))
+		num = 4;
+
+	for (i = 0; i < num; i++) {
+		alts = &iface->altsetting[i];
+		altsd = get_iface_desc(alts);
+		protocol = altsd->bInterfaceProtocol;
+		/* skip invalid one */
+		if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
+		     altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
+		    (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING &&
+		     altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) ||
+		    altsd->bNumEndpoints < 1 ||
+		    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0)
+			continue;
+		/* must be isochronous */
+		if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) !=
+		    USB_ENDPOINT_XFER_ISOC)
+			continue;
+		/* check direction */
+		stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
+			SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+		altno = altsd->bAlternateSetting;
+
+		if (snd_usb_apply_interface_quirk(chip, iface_no, altno))
+			continue;
+
+		/* get audio formats */
+		switch (protocol) {
+		default:
+			snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
+				    dev->devnum, iface_no, altno, protocol);
+			protocol = UAC_VERSION_1;
+			/* fall through */
+
+		case UAC_VERSION_1: {
+			struct uac1_as_header_descriptor *as =
+				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
+
+			if (!as) {
+				snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
+					   dev->devnum, iface_no, altno);
+				continue;
+			}
+
+			if (as->bLength < sizeof(*as)) {
+				snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
+					   dev->devnum, iface_no, altno);
+				continue;
+			}
+
+			format = le16_to_cpu(as->wFormatTag); /* remember the format value */
+			break;
+		}
+
+		case UAC_VERSION_2: {
+			struct uac2_input_terminal_descriptor *input_term;
+			struct uac2_output_terminal_descriptor *output_term;
+			struct uac2_as_header_descriptor *as =
+				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
+
+			if (!as) {
+				snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
+					   dev->devnum, iface_no, altno);
+				continue;
+			}
+
+			if (as->bLength < sizeof(*as)) {
+				snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
+					   dev->devnum, iface_no, altno);
+				continue;
+			}
+
+			num_channels = as->bNrChannels;
+			format = le32_to_cpu(as->bmFormats);
+
+			/* lookup the terminal associated to this interface
+			 * to extract the clock */
+			input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+									    as->bTerminalLink);
+			if (input_term) {
+				clock = input_term->bCSourceID;
+				break;
+			}
+
+			output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+									      as->bTerminalLink);
+			if (output_term) {
+				clock = output_term->bCSourceID;
+				break;
+			}
+
+			snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
+				   dev->devnum, iface_no, altno, as->bTerminalLink);
+			continue;
+		}
+		}
+
+		/* get format type */
+		fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE);
+		if (!fmt) {
+			snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n",
+				   dev->devnum, iface_no, altno);
+			continue;
+		}
+		if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
+		    ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
+			snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
+				   dev->devnum, iface_no, altno);
+			continue;
+		}
+
+		/*
+		 * Blue Microphones workaround: The last altsetting is identical
+		 * with the previous one, except for a larger packet size, but
+		 * is actually a mislabeled two-channel setting; ignore it.
+		 */
+		if (fmt->bNrChannels == 1 &&
+		    fmt->bSubframeSize == 2 &&
+		    altno == 2 && num == 3 &&
+		    fp && fp->altsetting == 1 && fp->channels == 1 &&
+		    fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
+		    protocol == UAC_VERSION_1 &&
+		    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
+							fp->maxpacksize * 2)
+			continue;
+
+		fp = kzalloc(sizeof(*fp), GFP_KERNEL);
+		if (! fp) {
+			snd_printk(KERN_ERR "cannot malloc\n");
+			return -ENOMEM;
+		}
+
+		fp->iface = iface_no;
+		fp->altsetting = altno;
+		fp->altset_idx = i;
+		fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+		fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+		fp->datainterval = snd_usb_parse_datainterval(chip, alts);
+		fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+		/* num_channels is only set for v2 interfaces */
+		fp->channels = num_channels;
+		if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
+			fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
+					* (fp->maxpacksize & 0x7ff);
+		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
+		fp->clock = clock;
+
+		/* some quirks for attributes here */
+
+		switch (chip->usb_id) {
+		case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */
+			/* Optoplay sets the sample rate attribute although
+			 * it seems not supporting it in fact.
+			 */
+			fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE;
+			break;
+		case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
+		case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+			/* doesn't set the sample rate attribute, but supports it */
+			fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE;
+			break;
+		case USB_ID(0x0763, 0x2001):  /* M-Audio Quattro USB */
+		case USB_ID(0x0763, 0x2012):  /* M-Audio Fast Track Pro USB */
+		case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
+		case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
+						an older model 77d:223) */
+		/*
+		 * plantronics headset and Griffin iMic have set adaptive-in
+		 * although it's really not...
+		 */
+			fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
+			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+				fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE;
+			else
+				fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
+			break;
+		}
+
+		/* ok, let's parse further... */
+		if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
+			kfree(fp->rate_table);
+			kfree(fp);
+			fp = NULL;
+			continue;
+		}
+
+		snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint);
+		err = snd_usb_add_audio_stream(chip, stream, fp);
+		if (err < 0) {
+			kfree(fp->rate_table);
+			kfree(fp);
+			return err;
+		}
+		/* try to set the interface... */
+		usb_set_interface(chip->dev, iface_no, altno);
+		snd_usb_init_pitch(chip, iface_no, alts, fp);
+		snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max);
+	}
+	return 0;
+}
+
diff --git a/sound/usb/stream.h b/sound/usb/stream.h
new file mode 100644
index 0000000..c97f679
--- /dev/null
+++ b/sound/usb/stream.h
@@ -0,0 +1,12 @@
+#ifndef __USBAUDIO_STREAM_H
+#define __USBAUDIO_STREAM_H
+
+int snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
+				  int iface_no);
+
+int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
+			     int stream,
+			     struct audioformat *fp);
+
+#endif /* __USBAUDIO_STREAM_H */
+
diff --git a/sound/usb/urb.c b/sound/usb/urb.c
deleted file mode 100644
index e184349..0000000
--- a/sound/usb/urb.c
+++ /dev/null
@@ -1,941 +0,0 @@
-/*
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License as published by
- *   the Free Software Foundation; either version 2 of the License, or
- *   (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
- */
-
-#include <linux/gfp.h>
-#include <linux/init.h>
-#include <linux/usb.h>
-#include <linux/usb/audio.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-
-#include "usbaudio.h"
-#include "helper.h"
-#include "card.h"
-#include "urb.h"
-#include "pcm.h"
-
-/*
- * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
- * this will overflow at approx 524 kHz
- */
-static inline unsigned get_usb_full_speed_rate(unsigned int rate)
-{
-	return ((rate << 13) + 62) / 125;
-}
-
-/*
- * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
- * this will overflow at approx 4 MHz
- */
-static inline unsigned get_usb_high_speed_rate(unsigned int rate)
-{
-	return ((rate << 10) + 62) / 125;
-}
-
-/*
- * unlink active urbs.
- */
-static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
-{
-	struct snd_usb_audio *chip = subs->stream->chip;
-	unsigned int i;
-	int async;
-
-	subs->running = 0;
-
-	if (!force && subs->stream->chip->shutdown) /* to be sure... */
-		return -EBADFD;
-
-	async = !can_sleep && chip->async_unlink;
-
-	if (!async && in_interrupt())
-		return 0;
-
-	for (i = 0; i < subs->nurbs; i++) {
-		if (test_bit(i, &subs->active_mask)) {
-			if (!test_and_set_bit(i, &subs->unlink_mask)) {
-				struct urb *u = subs->dataurb[i].urb;
-				if (async)
-					usb_unlink_urb(u);
-				else
-					usb_kill_urb(u);
-			}
-		}
-	}
-	if (subs->syncpipe) {
-		for (i = 0; i < SYNC_URBS; i++) {
-			if (test_bit(i+16, &subs->active_mask)) {
-				if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
-					struct urb *u = subs->syncurb[i].urb;
-					if (async)
-						usb_unlink_urb(u);
-					else
-						usb_kill_urb(u);
-				}
-			}
-		}
-	}
-	return 0;
-}
-
-
-/*
- * release a urb data
- */
-static void release_urb_ctx(struct snd_urb_ctx *u)
-{
-	if (u->urb) {
-		if (u->buffer_size)
-			usb_free_coherent(u->subs->dev, u->buffer_size,
-					u->urb->transfer_buffer,
-					u->urb->transfer_dma);
-		usb_free_urb(u->urb);
-		u->urb = NULL;
-	}
-}
-
-/*
- *  wait until all urbs are processed.
- */
-static int wait_clear_urbs(struct snd_usb_substream *subs)
-{
-	unsigned long end_time = jiffies + msecs_to_jiffies(1000);
-	unsigned int i;
-	int alive;
-
-	do {
-		alive = 0;
-		for (i = 0; i < subs->nurbs; i++) {
-			if (test_bit(i, &subs->active_mask))
-				alive++;
-		}
-		if (subs->syncpipe) {
-			for (i = 0; i < SYNC_URBS; i++) {
-				if (test_bit(i + 16, &subs->active_mask))
-					alive++;
-			}
-		}
-		if (! alive)
-			break;
-		schedule_timeout_uninterruptible(1);
-	} while (time_before(jiffies, end_time));
-	if (alive)
-		snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
-	return 0;
-}
-
-/*
- * release a substream
- */
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
-{
-	int i;
-
-	/* stop urbs (to be sure) */
-	deactivate_urbs(subs, force, 1);
-	wait_clear_urbs(subs);
-
-	for (i = 0; i < MAX_URBS; i++)
-		release_urb_ctx(&subs->dataurb[i]);
-	for (i = 0; i < SYNC_URBS; i++)
-		release_urb_ctx(&subs->syncurb[i]);
-	usb_free_coherent(subs->dev, SYNC_URBS * 4,
-			subs->syncbuf, subs->sync_dma);
-	subs->syncbuf = NULL;
-	subs->nurbs = 0;
-}
-
-/*
- * complete callback from data urb
- */
-static void snd_complete_urb(struct urb *urb)
-{
-	struct snd_urb_ctx *ctx = urb->context;
-	struct snd_usb_substream *subs = ctx->subs;
-	struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
-	int err = 0;
-
-	if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
-	    !subs->running || /* can be stopped during retire callback */
-	    (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
-	    (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
-		clear_bit(ctx->index, &subs->active_mask);
-		if (err < 0) {
-			snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
-			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		}
-	}
-}
-
-
-/*
- * complete callback from sync urb
- */
-static void snd_complete_sync_urb(struct urb *urb)
-{
-	struct snd_urb_ctx *ctx = urb->context;
-	struct snd_usb_substream *subs = ctx->subs;
-	struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
-	int err = 0;
-
-	if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
-	    !subs->running || /* can be stopped during retire callback */
-	    (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
-	    (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
-		clear_bit(ctx->index + 16, &subs->active_mask);
-		if (err < 0) {
-			snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
-			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		}
-	}
-}
-
-
-/*
- * initialize a substream for plaback/capture
- */
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
-				unsigned int period_bytes,
-				unsigned int rate,
-				unsigned int frame_bits)
-{
-	unsigned int maxsize, i;
-	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	unsigned int urb_packs, total_packs, packs_per_ms;
-	struct snd_usb_audio *chip = subs->stream->chip;
-
-	/* calculate the frequency in 16.16 format */
-	if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
-		subs->freqn = get_usb_full_speed_rate(rate);
-	else
-		subs->freqn = get_usb_high_speed_rate(rate);
-	subs->freqm = subs->freqn;
-	subs->freqshift = INT_MIN;
-	/* calculate max. frequency */
-	if (subs->maxpacksize) {
-		/* whatever fits into a max. size packet */
-		maxsize = subs->maxpacksize;
-		subs->freqmax = (maxsize / (frame_bits >> 3))
-				<< (16 - subs->datainterval);
-	} else {
-		/* no max. packet size: just take 25% higher than nominal */
-		subs->freqmax = subs->freqn + (subs->freqn >> 2);
-		maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
-				>> (16 - subs->datainterval);
-	}
-	subs->phase = 0;
-
-	if (subs->fill_max)
-		subs->curpacksize = subs->maxpacksize;
-	else
-		subs->curpacksize = maxsize;
-
-	if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
-		packs_per_ms = 8 >> subs->datainterval;
-	else
-		packs_per_ms = 1;
-
-	if (is_playback) {
-		urb_packs = max(chip->nrpacks, 1);
-		urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
-	} else
-		urb_packs = 1;
-	urb_packs *= packs_per_ms;
-	if (subs->syncpipe)
-		urb_packs = min(urb_packs, 1U << subs->syncinterval);
-
-	/* decide how many packets to be used */
-	if (is_playback) {
-		unsigned int minsize, maxpacks;
-		/* determine how small a packet can be */
-		minsize = (subs->freqn >> (16 - subs->datainterval))
-			  * (frame_bits >> 3);
-		/* with sync from device, assume it can be 12% lower */
-		if (subs->syncpipe)
-			minsize -= minsize >> 3;
-		minsize = max(minsize, 1u);
-		total_packs = (period_bytes + minsize - 1) / minsize;
-		/* we need at least two URBs for queueing */
-		if (total_packs < 2) {
-			total_packs = 2;
-		} else {
-			/* and we don't want too long a queue either */
-			maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
-			total_packs = min(total_packs, maxpacks);
-		}
-	} else {
-		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
-			urb_packs >>= 1;
-		total_packs = MAX_URBS * urb_packs;
-	}
-	subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
-	if (subs->nurbs > MAX_URBS) {
-		/* too much... */
-		subs->nurbs = MAX_URBS;
-		total_packs = MAX_URBS * urb_packs;
-	} else if (subs->nurbs < 2) {
-		/* too little - we need at least two packets
-		 * to ensure contiguous playback/capture
-		 */
-		subs->nurbs = 2;
-	}
-
-	/* allocate and initialize data urbs */
-	for (i = 0; i < subs->nurbs; i++) {
-		struct snd_urb_ctx *u = &subs->dataurb[i];
-		u->index = i;
-		u->subs = subs;
-		u->packets = (i + 1) * total_packs / subs->nurbs
-			- i * total_packs / subs->nurbs;
-		u->buffer_size = maxsize * u->packets;
-		if (subs->fmt_type == UAC_FORMAT_TYPE_II)
-			u->packets++; /* for transfer delimiter */
-		u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
-		if (!u->urb)
-			goto out_of_memory;
-		u->urb->transfer_buffer =
-			usb_alloc_coherent(subs->dev, u->buffer_size,
-					   GFP_KERNEL, &u->urb->transfer_dma);
-		if (!u->urb->transfer_buffer)
-			goto out_of_memory;
-		u->urb->pipe = subs->datapipe;
-		u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
-		u->urb->interval = 1 << subs->datainterval;
-		u->urb->context = u;
-		u->urb->complete = snd_complete_urb;
-	}
-
-	if (subs->syncpipe) {
-		/* allocate and initialize sync urbs */
-		subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4,
-						 GFP_KERNEL, &subs->sync_dma);
-		if (!subs->syncbuf)
-			goto out_of_memory;
-		for (i = 0; i < SYNC_URBS; i++) {
-			struct snd_urb_ctx *u = &subs->syncurb[i];
-			u->index = i;
-			u->subs = subs;
-			u->packets = 1;
-			u->urb = usb_alloc_urb(1, GFP_KERNEL);
-			if (!u->urb)
-				goto out_of_memory;
-			u->urb->transfer_buffer = subs->syncbuf + i * 4;
-			u->urb->transfer_dma = subs->sync_dma + i * 4;
-			u->urb->transfer_buffer_length = 4;
-			u->urb->pipe = subs->syncpipe;
-			u->urb->transfer_flags = URB_ISO_ASAP |
-						 URB_NO_TRANSFER_DMA_MAP;
-			u->urb->number_of_packets = 1;
-			u->urb->interval = 1 << subs->syncinterval;
-			u->urb->context = u;
-			u->urb->complete = snd_complete_sync_urb;
-		}
-	}
-	return 0;
-
-out_of_memory:
-	snd_usb_release_substream_urbs(subs, 0);
-	return -ENOMEM;
-}
-
-/*
- * prepare urb for full speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 10.14 frequency is passed through the pipe.
- */
-static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
-				    struct snd_pcm_runtime *runtime,
-				    struct urb *urb)
-{
-	unsigned char *cp = urb->transfer_buffer;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->iso_frame_desc[0].length = 3;
-	urb->iso_frame_desc[0].offset = 0;
-	cp[0] = subs->freqn >> 2;
-	cp[1] = subs->freqn >> 10;
-	cp[2] = subs->freqn >> 18;
-	return 0;
-}
-
-/*
- * prepare urb for high speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 12.13 frequency is passed as 16.16 through the pipe.
- */
-static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
-				       struct snd_pcm_runtime *runtime,
-				       struct urb *urb)
-{
-	unsigned char *cp = urb->transfer_buffer;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->iso_frame_desc[0].length = 4;
-	urb->iso_frame_desc[0].offset = 0;
-	cp[0] = subs->freqn;
-	cp[1] = subs->freqn >> 8;
-	cp[2] = subs->freqn >> 16;
-	cp[3] = subs->freqn >> 24;
-	return 0;
-}
-
-/*
- * process after capture sync complete
- * - nothing to do
- */
-static int retire_capture_sync_urb(struct snd_usb_substream *subs,
-				   struct snd_pcm_runtime *runtime,
-				   struct urb *urb)
-{
-	return 0;
-}
-
-/*
- * prepare urb for capture data pipe
- *
- * fill the offset and length of each descriptor.
- *
- * we use a temporary buffer to write the captured data.
- * since the length of written data is determined by host, we cannot
- * write onto the pcm buffer directly...  the data is thus copied
- * later at complete callback to the global buffer.
- */
-static int prepare_capture_urb(struct snd_usb_substream *subs,
-			       struct snd_pcm_runtime *runtime,
-			       struct urb *urb)
-{
-	int i, offs;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	offs = 0;
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	for (i = 0; i < ctx->packets; i++) {
-		urb->iso_frame_desc[i].offset = offs;
-		urb->iso_frame_desc[i].length = subs->curpacksize;
-		offs += subs->curpacksize;
-	}
-	urb->transfer_buffer_length = offs;
-	urb->number_of_packets = ctx->packets;
-	return 0;
-}
-
-/*
- * process after capture complete
- *
- * copy the data from each desctiptor to the pcm buffer, and
- * update the current position.
- */
-static int retire_capture_urb(struct snd_usb_substream *subs,
-			      struct snd_pcm_runtime *runtime,
-			      struct urb *urb)
-{
-	unsigned long flags;
-	unsigned char *cp;
-	int i;
-	unsigned int stride, frames, bytes, oldptr;
-	int period_elapsed = 0;
-
-	stride = runtime->frame_bits >> 3;
-
-	for (i = 0; i < urb->number_of_packets; i++) {
-		cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
-		if (urb->iso_frame_desc[i].status) {
-			snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
-			// continue;
-		}
-		bytes = urb->iso_frame_desc[i].actual_length;
-		frames = bytes / stride;
-		if (!subs->txfr_quirk)
-			bytes = frames * stride;
-		if (bytes % (runtime->sample_bits >> 3) != 0) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-			int oldbytes = bytes;
-#endif
-			bytes = frames * stride;
-			snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
-							oldbytes, bytes);
-		}
-		/* update the current pointer */
-		spin_lock_irqsave(&subs->lock, flags);
-		oldptr = subs->hwptr_done;
-		subs->hwptr_done += bytes;
-		if (subs->hwptr_done >= runtime->buffer_size * stride)
-			subs->hwptr_done -= runtime->buffer_size * stride;
-		frames = (bytes + (oldptr % stride)) / stride;
-		subs->transfer_done += frames;
-		if (subs->transfer_done >= runtime->period_size) {
-			subs->transfer_done -= runtime->period_size;
-			period_elapsed = 1;
-		}
-		spin_unlock_irqrestore(&subs->lock, flags);
-		/* copy a data chunk */
-		if (oldptr + bytes > runtime->buffer_size * stride) {
-			unsigned int bytes1 =
-					runtime->buffer_size * stride - oldptr;
-			memcpy(runtime->dma_area + oldptr, cp, bytes1);
-			memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
-		} else {
-			memcpy(runtime->dma_area + oldptr, cp, bytes);
-		}
-	}
-	if (period_elapsed)
-		snd_pcm_period_elapsed(subs->pcm_substream);
-	return 0;
-}
-
-/*
- * Process after capture complete when paused.  Nothing to do.
- */
-static int retire_paused_capture_urb(struct snd_usb_substream *subs,
-				     struct snd_pcm_runtime *runtime,
-				     struct urb *urb)
-{
-	return 0;
-}
-
-
-/*
- * prepare urb for playback sync pipe
- *
- * set up the offset and length to receive the current frequency.
- */
-static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
-				     struct snd_pcm_runtime *runtime,
-				     struct urb *urb)
-{
-	struct snd_urb_ctx *ctx = urb->context;
-
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize);
-	urb->iso_frame_desc[0].offset = 0;
-	return 0;
-}
-
-/*
- * process after playback sync complete
- *
- * Full speed devices report feedback values in 10.14 format as samples per
- * frame, high speed devices in 16.16 format as samples per microframe.
- * Because the Audio Class 1 spec was written before USB 2.0, many high speed
- * devices use a wrong interpretation, some others use an entirely different
- * format.  Therefore, we cannot predict what format any particular device uses
- * and must detect it automatically.
- */
-static int retire_playback_sync_urb(struct snd_usb_substream *subs,
-				    struct snd_pcm_runtime *runtime,
-				    struct urb *urb)
-{
-	unsigned int f;
-	int shift;
-	unsigned long flags;
-
-	if (urb->iso_frame_desc[0].status != 0 ||
-	    urb->iso_frame_desc[0].actual_length < 3)
-		return 0;
-
-	f = le32_to_cpup(urb->transfer_buffer);
-	if (urb->iso_frame_desc[0].actual_length == 3)
-		f &= 0x00ffffff;
-	else
-		f &= 0x0fffffff;
-	if (f == 0)
-		return 0;
-
-	if (unlikely(subs->freqshift == INT_MIN)) {
-		/*
-		 * The first time we see a feedback value, determine its format
-		 * by shifting it left or right until it matches the nominal
-		 * frequency value.  This assumes that the feedback does not
-		 * differ from the nominal value more than +50% or -25%.
-		 */
-		shift = 0;
-		while (f < subs->freqn - subs->freqn / 4) {
-			f <<= 1;
-			shift++;
-		}
-		while (f > subs->freqn + subs->freqn / 2) {
-			f >>= 1;
-			shift--;
-		}
-		subs->freqshift = shift;
-	}
-	else if (subs->freqshift >= 0)
-		f <<= subs->freqshift;
-	else
-		f >>= -subs->freqshift;
-
-	if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) {
-		/*
-		 * If the frequency looks valid, set it.
-		 * This value is referred to in prepare_playback_urb().
-		 */
-		spin_lock_irqsave(&subs->lock, flags);
-		subs->freqm = f;
-		spin_unlock_irqrestore(&subs->lock, flags);
-	} else {
-		/*
-		 * Out of range; maybe the shift value is wrong.
-		 * Reset it so that we autodetect again the next time.
-		 */
-		subs->freqshift = INT_MIN;
-	}
-
-	return 0;
-}
-
-/* determine the number of frames in the next packet */
-static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
-{
-	if (subs->fill_max)
-		return subs->maxframesize;
-	else {
-		subs->phase = (subs->phase & 0xffff)
-			+ (subs->freqm << subs->datainterval);
-		return min(subs->phase >> 16, subs->maxframesize);
-	}
-}
-
-/*
- * Prepare urb for streaming before playback starts or when paused.
- *
- * We don't have any data, so we send silence.
- */
-static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
-				       struct snd_pcm_runtime *runtime,
-				       struct urb *urb)
-{
-	unsigned int i, offs, counts;
-	struct snd_urb_ctx *ctx = urb->context;
-	int stride = runtime->frame_bits >> 3;
-
-	offs = 0;
-	urb->dev = ctx->subs->dev;
-	for (i = 0; i < ctx->packets; ++i) {
-		counts = snd_usb_audio_next_packet_size(subs);
-		urb->iso_frame_desc[i].offset = offs * stride;
-		urb->iso_frame_desc[i].length = counts * stride;
-		offs += counts;
-	}
-	urb->number_of_packets = ctx->packets;
-	urb->transfer_buffer_length = offs * stride;
-	memset(urb->transfer_buffer,
-	       runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
-	       offs * stride);
-	return 0;
-}
-
-/*
- * prepare urb for playback data pipe
- *
- * Since a URB can handle only a single linear buffer, we must use double
- * buffering when the data to be transferred overflows the buffer boundary.
- * To avoid inconsistencies when updating hwptr_done, we use double buffering
- * for all URBs.
- */
-static int prepare_playback_urb(struct snd_usb_substream *subs,
-				struct snd_pcm_runtime *runtime,
-				struct urb *urb)
-{
-	int i, stride;
-	unsigned int counts, frames, bytes;
-	unsigned long flags;
-	int period_elapsed = 0;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	stride = runtime->frame_bits >> 3;
-
-	frames = 0;
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->number_of_packets = 0;
-	spin_lock_irqsave(&subs->lock, flags);
-	for (i = 0; i < ctx->packets; i++) {
-		counts = snd_usb_audio_next_packet_size(subs);
-		/* set up descriptor */
-		urb->iso_frame_desc[i].offset = frames * stride;
-		urb->iso_frame_desc[i].length = counts * stride;
-		frames += counts;
-		urb->number_of_packets++;
-		subs->transfer_done += counts;
-		if (subs->transfer_done >= runtime->period_size) {
-			subs->transfer_done -= runtime->period_size;
-			period_elapsed = 1;
-			if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
-				if (subs->transfer_done > 0) {
-					/* FIXME: fill-max mode is not
-					 * supported yet */
-					frames -= subs->transfer_done;
-					counts -= subs->transfer_done;
-					urb->iso_frame_desc[i].length =
-						counts * stride;
-					subs->transfer_done = 0;
-				}
-				i++;
-				if (i < ctx->packets) {
-					/* add a transfer delimiter */
-					urb->iso_frame_desc[i].offset =
-						frames * stride;
-					urb->iso_frame_desc[i].length = 0;
-					urb->number_of_packets++;
-				}
-				break;
-			}
-		}
-		if (period_elapsed) /* finish at the period boundary */
-			break;
-	}
-	bytes = frames * stride;
-	if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
-		/* err, the transferred area goes over buffer boundary. */
-		unsigned int bytes1 =
-			runtime->buffer_size * stride - subs->hwptr_done;
-		memcpy(urb->transfer_buffer,
-		       runtime->dma_area + subs->hwptr_done, bytes1);
-		memcpy(urb->transfer_buffer + bytes1,
-		       runtime->dma_area, bytes - bytes1);
-	} else {
-		memcpy(urb->transfer_buffer,
-		       runtime->dma_area + subs->hwptr_done, bytes);
-	}
-	subs->hwptr_done += bytes;
-	if (subs->hwptr_done >= runtime->buffer_size * stride)
-		subs->hwptr_done -= runtime->buffer_size * stride;
-	runtime->delay += frames;
-	spin_unlock_irqrestore(&subs->lock, flags);
-	urb->transfer_buffer_length = bytes;
-	if (period_elapsed)
-		snd_pcm_period_elapsed(subs->pcm_substream);
-	return 0;
-}
-
-/*
- * process after playback data complete
- * - decrease the delay count again
- */
-static int retire_playback_urb(struct snd_usb_substream *subs,
-			       struct snd_pcm_runtime *runtime,
-			       struct urb *urb)
-{
-	unsigned long flags;
-	int stride = runtime->frame_bits >> 3;
-	int processed = urb->transfer_buffer_length / stride;
-
-	spin_lock_irqsave(&subs->lock, flags);
-	if (processed > runtime->delay)
-		runtime->delay = 0;
-	else
-		runtime->delay -= processed;
-	spin_unlock_irqrestore(&subs->lock, flags);
-	return 0;
-}
-
-static const char *usb_error_string(int err)
-{
-	switch (err) {
-	case -ENODEV:
-		return "no device";
-	case -ENOENT:
-		return "endpoint not enabled";
-	case -EPIPE:
-		return "endpoint stalled";
-	case -ENOSPC:
-		return "not enough bandwidth";
-	case -ESHUTDOWN:
-		return "device disabled";
-	case -EHOSTUNREACH:
-		return "device suspended";
-	case -EINVAL:
-	case -EAGAIN:
-	case -EFBIG:
-	case -EMSGSIZE:
-		return "internal error";
-	default:
-		return "unknown error";
-	}
-}
-
-/*
- * set up and start data/sync urbs
- */
-static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
-{
-	unsigned int i;
-	int err;
-
-	if (subs->stream->chip->shutdown)
-		return -EBADFD;
-
-	for (i = 0; i < subs->nurbs; i++) {
-		if (snd_BUG_ON(!subs->dataurb[i].urb))
-			return -EINVAL;
-		if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
-			snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
-			goto __error;
-		}
-	}
-	if (subs->syncpipe) {
-		for (i = 0; i < SYNC_URBS; i++) {
-			if (snd_BUG_ON(!subs->syncurb[i].urb))
-				return -EINVAL;
-			if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
-				snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
-				goto __error;
-			}
-		}
-	}
-
-	subs->active_mask = 0;
-	subs->unlink_mask = 0;
-	subs->running = 1;
-	for (i = 0; i < subs->nurbs; i++) {
-		err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
-		if (err < 0) {
-			snd_printk(KERN_ERR "cannot submit datapipe "
-				   "for urb %d, error %d: %s\n",
-				   i, err, usb_error_string(err));
-			goto __error;
-		}
-		set_bit(i, &subs->active_mask);
-	}
-	if (subs->syncpipe) {
-		for (i = 0; i < SYNC_URBS; i++) {
-			err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
-			if (err < 0) {
-				snd_printk(KERN_ERR "cannot submit syncpipe "
-					   "for urb %d, error %d: %s\n",
-					   i, err, usb_error_string(err));
-				goto __error;
-			}
-			set_bit(i + 16, &subs->active_mask);
-		}
-	}
-	return 0;
-
- __error:
-	// snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
-	deactivate_urbs(subs, 0, 0);
-	return -EPIPE;
-}
-
-
-/*
- */
-static struct snd_urb_ops audio_urb_ops[2] = {
-	{
-		.prepare =	prepare_nodata_playback_urb,
-		.retire =	retire_playback_urb,
-		.prepare_sync =	prepare_playback_sync_urb,
-		.retire_sync =	retire_playback_sync_urb,
-	},
-	{
-		.prepare =	prepare_capture_urb,
-		.retire =	retire_capture_urb,
-		.prepare_sync =	prepare_capture_sync_urb,
-		.retire_sync =	retire_capture_sync_urb,
-	},
-};
-
-/*
- * initialize the substream instance.
- */
-
-void snd_usb_init_substream(struct snd_usb_stream *as,
-			    int stream, struct audioformat *fp)
-{
-	struct snd_usb_substream *subs = &as->substream[stream];
-
-	INIT_LIST_HEAD(&subs->fmt_list);
-	spin_lock_init(&subs->lock);
-
-	subs->stream = as;
-	subs->direction = stream;
-	subs->dev = as->chip->dev;
-	subs->txfr_quirk = as->chip->txfr_quirk;
-	subs->ops = audio_urb_ops[stream];
-	if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH)
-		subs->ops.prepare_sync = prepare_capture_sync_urb_hs;
-
-	snd_usb_set_pcm_ops(as->pcm, stream);
-
-	list_add_tail(&fp->list, &subs->fmt_list);
-	subs->formats |= fp->formats;
-	subs->endpoint = fp->endpoint;
-	subs->num_formats++;
-	subs->fmt_type = fp->fmt_type;
-}
-
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_usb_substream *subs = substream->runtime->private_data;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		subs->ops.prepare = prepare_playback_urb;
-		return 0;
-	case SNDRV_PCM_TRIGGER_STOP:
-		return deactivate_urbs(subs, 0, 0);
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		subs->ops.prepare = prepare_nodata_playback_urb;
-		return 0;
-	}
-
-	return -EINVAL;
-}
-
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_usb_substream *subs = substream->runtime->private_data;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		subs->ops.retire = retire_capture_urb;
-		return start_urbs(subs, substream->runtime);
-	case SNDRV_PCM_TRIGGER_STOP:
-		return deactivate_urbs(subs, 0, 0);
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		subs->ops.retire = retire_paused_capture_urb;
-		return 0;
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		subs->ops.retire = retire_capture_urb;
-		return 0;
-	}
-
-	return -EINVAL;
-}
-
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
-			      struct snd_pcm_runtime *runtime)
-{
-	/* clear urbs (to be sure) */
-	deactivate_urbs(subs, 0, 1);
-	wait_clear_urbs(subs);
-
-	/* for playback, submit the URBs now; otherwise, the first hwptr_done
-	 * updates for all URBs would happen at the same time when starting */
-	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
-		subs->ops.prepare = prepare_nodata_playback_urb;
-		return start_urbs(subs, runtime);
-	}
-
-	return 0;
-}
-
diff --git a/sound/usb/urb.h b/sound/usb/urb.h
deleted file mode 100644
index 888da38..0000000
--- a/sound/usb/urb.h
+++ /dev/null
@@ -1,21 +0,0 @@
-#ifndef __USBAUDIO_URB_H
-#define __USBAUDIO_URB_H
-
-void snd_usb_init_substream(struct snd_usb_stream *as,
-			    int stream,
-			    struct audioformat *fp);
-
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
-				unsigned int period_bytes,
-				unsigned int rate,
-				unsigned int frame_bits);
-
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force);
-
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
-			      struct snd_pcm_runtime *runtime);
-
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd);
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd);
-
-#endif /* __USBAUDIO_URB_H */
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 1e79986..3e2b035 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -80,6 +80,7 @@
 	QUIRK_MIDI_CME,
 	QUIRK_MIDI_AKAI,
 	QUIRK_MIDI_US122L,
+	QUIRK_MIDI_FTDI,
 	QUIRK_AUDIO_STANDARD_INTERFACE,
 	QUIRK_AUDIO_FIXED_ENDPOINT,
 	QUIRK_AUDIO_EDIROL_UAXX,