154 files changed, 4369 insertions, 7796 deletions

diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 598c22f3b3ac..5de23c007078 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -4288,7 +4288,7 @@ struct _snd_pcm_runtime {
 <![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 @@ struct _snd_pcm_runtime {
         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 @@ struct _snd_pcm_runtime {
       </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 @@ struct _snd_pcm_runtime {
       <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 89757012c7ff..936699e4f04b 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -886,6 +886,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
                 disable)
     power_save_controller - Reset HD-audio controller in power-saving mode
                 (default = on)
+    align_buffer_size - Force rounding of buffer/period sizes to multiples
+                      of 128 bytes. This is more efficient in terms of memory
+                      access but isn't required by the HDA spec and prevents
+                      users from specifying exact period/buffer sizes.
+                      (default = on)
+    snoop       - Enable/disable snooping (default = on)
 
     This module supports multiple cards and autoprobe.
     
diff --git a/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt
index 1482035243e6..e9621e349e17 100644
--- a/Documentation/sound/alsa/HD-Audio-Controls.txt
+++ b/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -98,3 +98,19 @@ Conexant codecs
 
 * Auto-Mute Mode
   See Reatek codecs.
+
+
+Analog codecs
+--------------
+
+* Channel Mode
+  This is an enum control to change the surround-channel setup,
+  appears only when the surround channels are available.
+  It gives the number of channels to be used, "2ch", "4ch" and "6ch".
+  According to the configuration, this also controls the
+  jack-retasking of multi-I/O jacks.
+
+* Independent HP
+  When this enum control is enabled, the headphone output is routed
+  from an individual stream (the third PCM such as hw:0,2) instead of
+  the primary stream.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index d70c93bdcadf..4f3443230d89 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -29,9 +29,6 @@ ALC880
 
 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 @@ ALC260
 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 @@ ALC262
 
 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 @@ ALC662/663/272
   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 @@ ALC882/883/885/888/889
 
 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 @@ Conexant 5051
   hp-dv6736     HP dv6736
   hp-f700       HP Compaq Presario F700
   ideapad       Lenovo IdeaPad laptop
-  lenovo-x200   Lenovo X200 laptop
   toshiba       Toshiba Satellite M300
 
 Conexant 5066
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c82beb007634..03e2771ddeef 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -447,7 +447,10 @@ The file needs to have a line `[codec]`.  The next line should contain
 three numbers indicating the codec vendor-id (0x12345678 in the
 example), the codec subsystem-id (0xabcd1234) and the address (2) of
 the codec.  The rest patch entries are applied to this specified codec
-until another codec entry is given.
+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 @@ Also, the codec chip name can be rewritten via `[chip_name]` line.
 The hd-audio driver reads the file via request_firmware().  Thus,
 a patch file has to be located on the appropriate firmware path,
 typically, /lib/firmware.  For example, when you pass the option
-`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be
+`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be
 present.
 
 The patch module option is specific to each card instance, and you
@@ -524,6 +527,54 @@ power-saving.  See /sys/module/snd_hda_intel/parameters/power_save to
 check the current value.  If it's non-zero, the feature is turned on.
 
 
+Tracepoints
+~~~~~~~~~~~
+The hd-audio driver gives a few basic tracepoints.
+`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response`
+traces the response from RIRB (only when read from the codec driver).
+`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc,
+`hda:hda_unsol_event` traces the unsolicited events, and
+`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up
+via power-saving behavior.
+
+Enabling all tracepoints can be done like
+------------------------------------------------------------------------
+  # echo 1 > /sys/kernel/debug/tracing/events/hda/enable
+------------------------------------------------------------------------
+then after some commands, you can traces from
+/sys/kernel/debug/tracing/trace file.  For example, when you want to
+trace what codec command is sent, enable the tracepoint like:
+------------------------------------------------------------------------
+  # cat /sys/kernel/debug/tracing/trace
+  # tracer: nop
+  #
+  #       TASK-PID    CPU#    TIMESTAMP  FUNCTION
+  #          | |       |          |         |
+         <...>-7807  [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019
+         <...>-7807  [002] 105147.774893: hda_send_cmd: [0:0] val=e39019
+         <...>-7807  [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a
+         <...>-7807  [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a
+         <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019
+         <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019
+         <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a
+         <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a
+------------------------------------------------------------------------
+Here `[0:0]` indicates the card number and the codec address, and
+`val` shows the value sent to the codec, respectively.  The value is
+a packed value, and you can decode it via hda-decode-verb program
+included in hda-emu package below.  For example, the value e3a019 is
+to set the left output-amp value to 25.
+------------------------------------------------------------------------
+  % hda-decode-verb 0xe3a019
+  raw value = 0x00e3a019
+  cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19
+  raw value: verb = 0x3a0, parm = 0x19
+  verbname = set_amp_gain_mute
+  amp raw val = 0xa019
+  output, left, idx=0, mute=0, val=25
+------------------------------------------------------------------------
+
+
 Development Tree
 ~~~~~~~~~~~~~~~~
 The latest development codes for HD-audio are found on sound git tree:
diff --git a/MAINTAINERS b/MAINTAINERS
index bbf42cd74e2a..1dbbb278d218 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5991,7 +5991,7 @@ M:	Jaroslav Kysela <perex@perex.cz>
 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 a637e7814334..a514fb8faea3 100644
--- a/include/linux/input.h
+++ b/include/linux/input.h
@@ -814,6 +814,7 @@ struct input_keymap_entry {
 #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 0fd3fbdd8283..f30253599501 100644
--- a/include/linux/usb/ch9.h
+++ b/include/linux/usb/ch9.h
@@ -377,12 +377,6 @@ struct usb_endpoint_descriptor {
 #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 @@ struct usb_endpoint_descriptor {
 #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 5d6074faa279..a2e4ff5ba9e9 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -706,7 +706,7 @@ struct snd_timer_tread {
  *                                                                          *
  ****************************************************************************/
 
-#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 @@ struct snd_ctl_elem_info {
                         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 1daa6dff8297..f99a0d2ddfe7 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -62,7 +62,7 @@ static int snd_legacy_find_free_irq(int *irq_table)
 {
         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 c140fc7cbd3f..63c790742db4 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -42,6 +42,7 @@ enum snd_jack_types {
         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 1f1d53f8830b..20230db00ef1 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 @@ struct snd_mpu401 {
         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 @@ int snd_mpu401_uart_new(struct snd_card *card,
                         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 57e71fa33f7c..3e7fda6e8164 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -825,6 +825,8 @@ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime,
 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 @@ static inline void snd_pcm_mmap_data_close(struct vm_area_struct *area)
         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 3687a6cc9881..762af68c8996 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -1067,7 +1067,6 @@ static int onyx_i2c_probe(struct i2c_client *client,
         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 @@ static int onyx_i2c_remove(struct i2c_client *client)
 
         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 3fd1a7e24928..552b97afbca5 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
         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 d0cead38d5fb..e518d38b1c74 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -443,7 +443,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream)
         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 88eec3847df2..8ad65352bf91 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -359,7 +359,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
         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 f8c5be464510..978fe1a8e9f0 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -989,7 +989,6 @@ struct user_element {
         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 @@ static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol,
         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 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol,
         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 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
         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 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
         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 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
                 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 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
         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 426874429a5e..2bb95a7a8809 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -83,6 +83,8 @@ struct snd_ctl_elem_info32 {
                         u32 items;
                         u32 item;
                         char name[64];
+                        u64 names_ptr;
+                        u32 names_length;
                 } enumerated;
                 unsigned char reserved[128];
         } value;
@@ -372,6 +374,8 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file,
                                    &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 a70ee7f1ed98..031e215b6dde 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -272,7 +272,14 @@ static int snd_hwdep_control_ioctl(struct snd_card *card,
                         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 53b53e97c896..240a3e13470d 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -30,6 +30,7 @@ static int jack_switch_types[] = {
         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 d8359cfeca15..1b5e0c49a0ad 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -499,7 +499,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c
         
         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 86d0caf91b35..95d1e789715f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1399,6 +1399,32 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
 
 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 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
         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 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
                 }
                 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 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
                         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 1c6be91dfb98..77d7df22e7c8 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2058,16 +2058,12 @@ EXPORT_SYMBOL(snd_pcm_open_substream);
 
 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 @@ static int snd_pcm_open_file(struct file *file,
                 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_capture_open(struct inode *inode, struct file *file)
 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 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream)
         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 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = {
 /*
  * 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 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
         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 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file,
         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 7c1cbf0a0dc4..67ebf1c21c04 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -328,6 +328,8 @@ int snd_timer_close(struct snd_timer_instance *timeri)
                 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 @@ int snd_timer_close(struct snd_timer_instance *timeri)
                 }
                 mutex_unlock(&register_mutex);
         }
+ out:
         if (timeri->private_free)
                 timeri->private_free(timeri);
         kfree(timeri->owner);
@@ -531,6 +534,8 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
         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 a0da7755fcea..4067f1548949 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -575,7 +575,8 @@ static void loopback_runtime_free(struct snd_pcm_runtime *runtime)
 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 @@ static int loopback_hw_free(struct snd_pcm_substream *substream)
         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 @@ static struct snd_pcm_ops loopback_playback_ops = {
         .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 @@ static struct snd_pcm_ops loopback_capture_ops = {
         .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 @@ static int __devinit loopback_pcm_new(struct loopback *loopback,
         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 5cfcb908c430..2c7a7636f472 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
                    "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 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
                    "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 149d05a8202d..1c02852aceea 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -86,8 +86,7 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard)
         }
 
         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 2af09996a3d0..e91698a634b2 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 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up)
                         /* 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_trigger(struct snd_rawmidi_substream *substream, int up)
                 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 struct snd_rawmidi_ops snd_mpu401_uart_input =
 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 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
  * @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 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
                         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 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
                 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 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
                         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 5c426df87678..1eef4ccebe4b 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -589,7 +589,7 @@ static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard)
                 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 a25fb7b1f441..fc1d822802c3 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -816,7 +816,7 @@ static int __devinit snd_uart16550_create(struct snd_card *card,
 
         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 440030818db7..cd094ecaca3b 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -51,7 +51,6 @@ struct isight {
         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 3fc257da180c..cbe6bb9e53b6 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -778,9 +778,10 @@ static int __devexit fwspk_remove(struct device *dev)
 {
         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 @@ static void fwspk_bus_reset(struct fw_unit *unit)
         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 3cb75bc97699..a87a2b566e19 100644
--- a/sound/isa/ad1816a/ad1816a.c
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -204,7 +204,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard
 
         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 05aef8b97e96..177eed3271bc 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -595,7 +595,7 @@ int __devinit snd_ad1816a_create(struct snd_card *card,
                 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 20becc89f6f6..706effd6b3cd 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -256,7 +256,6 @@ static int __devinit snd_card_als100_probe(int dev,
                                         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 aac8dc15c2fe..b7bdbf307740 100644
--- a/sound/isa/azt2320.c
+++ b/sound/isa/azt2320.c
@@ -234,8 +234,7 @@ static int __devinit snd_card_azt2320_probe(int dev,
         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 fe79a169acb5..dca69f80305f 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -597,7 +597,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
         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 cb9153e75b82..409fa0ad7843 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -131,7 +131,6 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
                         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 999dc1e0fdbd..0dbde461e6c1 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -449,8 +449,7 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev)
                         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 0cde8131a575..5493e9e4bcd5 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -174,7 +174,7 @@ static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n)
                         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 07676200496a..d3eab6fb0866 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -661,7 +661,7 @@ int snd_es1688_create(struct snd_card *card,
                 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 fb4d6b34bbca..bf6ad0bf51c6 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1805,7 +1805,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card,
                 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 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev)
 
         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 ee54df082b9c..e51d3244742a 100644
--- a/sound/isa/galaxy/galaxy.c
+++ b/sound/isa/galaxy/galaxy.c
@@ -585,8 +585,7 @@ static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n)
 
         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 12eb98f2f931..3167e5ac3699 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -180,7 +180,7 @@ int snd_gus_create(struct snd_card *card,
                 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 008e8e5bfa37..c4733c08b60b 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -317,8 +317,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
 
         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 3e4a58b72913..c43faa057ff6 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -291,7 +291,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev)
                 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 c7b80e4730fc..5f869a32b48c 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -684,7 +684,7 @@ static int __devinit snd_interwave_probe(struct snd_card *card, int dev)
         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 91d6023a63e5..0961e2cf20ca 100644
--- a/sound/isa/msnd/msnd_pinnacle.c
+++ b/sound/isa/msnd/msnd_pinnacle.c
@@ -600,7 +600,7 @@ static int __devinit snd_msnd_attach(struct snd_card *card)
                                           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 9b915e27b5bd..bbafb0b543ea 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -667,7 +667,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
         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 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
         }
         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 8c24102d0d93..d94d0f35cb76 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -1377,8 +1377,7 @@ static int __devinit snd_miro_probe(struct snd_card *card)
                 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 c35dc68930dc..6dbbfa76b440 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -892,7 +892,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 #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 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
                 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 8ccbcddf08e1..54e3c2c18060 100644
--- a/sound/isa/sb/jazz16.c
+++ b/sound/isa/sb/jazz16.c
@@ -322,7 +322,6 @@ static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev)
                                         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 4d1c5a300ff8..237f8bd7fbe4 100644
--- a/sound/isa/sb/sb16.c
+++ b/sound/isa/sb/sb16.c
@@ -394,8 +394,9 @@ static int __devinit snd_sb16_probe(struct snd_card *card, int dev)
 
         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 eae6c1c0eff9..d2e19215813e 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -240,7 +240,7 @@ int snd_sbdsp_create(struct snd_card *card,
         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 9a8bbf6dd62a..207c161f100c 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -658,8 +658,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev)
                 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 e2d5d2d3ed96..f2379e102b63 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -825,8 +825,7 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum,
         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 711670e4a425..87142977335a 100644
--- a/sound/isa/wavefront/wavefront.c
+++ b/sound/isa/wavefront/wavefront.c
@@ -418,7 +418,7 @@ snd_wavefront_probe (struct snd_card *card, int dev)
                 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 @@ snd_wavefront_probe (struct snd_card *card, int dev)
         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 2a42cc377957..7277c5b7df6c 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1833,7 +1833,7 @@ int snd_wss_create(struct snd_card *card,
         }
         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 446cf9748664..7567ebd71913 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -465,13 +465,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000)
 
         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 8f7d175767a2..6f13ab4afc6b 100644
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@ static int pcm_set_speed(int arg)
 
         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 9b800ce5100e..2fc0624024b5 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -673,7 +673,8 @@ static void configure_nonsound_components(void)
 
         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 @@ static void __exit cleanup_pss(void)
                 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 48cda6c4c257..8021c85f076d 100644
--- a/sound/oss/sound_timer.c
+++ b/sound/oss/sound_timer.c
@@ -320,7 +320,7 @@ void  sound_timer_init(struct sound_lowlev_timer *t, char *name)
         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 50abf5bf8e09..88168044375f 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 @@ config SND_FM801_TEA575X_BOOL
           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 200c9a1d48b7..a872d0a82976 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
         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 a9c1af33f276..04628696eb08 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -931,8 +931,9 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci,
 
         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 65b7ca13115b..bd47521b24ec 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
         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 0dc8d259d1ed..e6c6a0febb75 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -84,7 +84,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
 #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 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
         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 e4d76a270c9f..d24fe425e87f 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
         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 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
            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 9cf99fb7eb9c..da9c73211eca 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3228,8 +3228,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
                 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 457d21189b0d..2c8622617c8c 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -404,7 +404,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc,
         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 c749fa720889..e134b3a5780d 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 b23adfca4de6..e6da60eb19ce 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 622bace148e3..e22b8e2bbd88 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1146,6 +1146,11 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream)
                 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 26a5a2f25d4b..718a2643474e 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1854,8 +1854,9 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci,
                 }
         }
         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 99ea9320c6b5..407e4abc4356 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2843,8 +2843,9 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci,
         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 f9123f09e83e..136f7232bb7c 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 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 @@ static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = {
         { .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 @@ static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea)
 {
         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 @@ static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output
 {
         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 @@ static int snd_fm801_free(struct fm801 *chip)
 
       __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 @@ static int __devinit snd_fm801_create(struct snd_card *card,
             (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 @@ static int __devinit snd_fm801_create(struct snd_card *card,
                         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 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
         }
         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 87365d5ea2a9..f928d6634723 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -6,6 +6,9 @@ snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
 snd-hda-codec-$(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 21ec2cb100b0..3b5170b9700f 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 @@ static const struct hda_channel_mode alc260_modes[1] = {
 /* 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 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = {
         { } /* 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 @@ static const struct hda_verb alc260_init_verbs[] = {
         { }
 };
 
-#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 struct hda_verb alc260_test_init_verbs[] = {
  */
 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 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = {
         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 @@ static const struct alc_config_preset alc260_presets[] = {
                 .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 8d2097d77642..7894b2b5aacf 100644
--- a/sound/pci/hda/alc262_quirks.c
+++ b/sound/pci/hda/alc262_quirks.c
@@ -10,13 +10,7 @@ enum {
         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,
@@ -66,164 +60,31 @@ static const struct snd_kcontrol_new alc262_base_mixer[] = {
         { } /* end */
 };
 
-/* update HP, line and mono-out pins according to the master switch */
-#define alc262_hp_master_update         alc260_hp_master_update
+/* bind hp and internal speaker mute (with plug check) as master switch */
 
-static void alc262_hp_bpc_setup(struct hda_codec *codec)
+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;
-
-        spec->autocfg.hp_pins[0] = 0x1b;
-        spec->autocfg.speaker_pins[0] = 0x16;
-        spec->automute = 1;
-        spec->automute_mode = ALC_AUTOMUTE_PIN;
+        *ucontrol->value.integer.value = !spec->master_mute;
+        return 0;
 }
 
-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)
+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;
 
-        spec->autocfg.hp_pins[0] = 0x15;
-        spec->autocfg.speaker_pins[0] = 0x14;
-        spec->automute = 1;
-        spec->automute_mode = ALC_AUTOMUTE_PIN;
+        if (val == spec->master_mute)
+                return 0;
+        spec->master_mute = val;
+        update_outputs(codec);
+        return 1;
 }
 
-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
-
 #define ALC262_HIPPO_MASTER_SWITCH                              \
         {                                                       \
                 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,            \
@@ -239,6 +100,9 @@ static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
                              (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 @@ static void alc262_hippo_setup(struct hda_codec *codec)
 
         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 @@ static void alc262_hippo1_setup(struct hda_codec *codec)
 
         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 @@ static void alc262_tyan_setup(struct hda_codec *codec)
 
         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 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec)
         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_fujitsu_capture_source = {
         },
 };
 
-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 @@ static void alc262_fujitsu_setup(struct hda_codec *codec)
         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 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec)
         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 @@ static void alc262_ultra_automute(struct hda_codec *codec)
         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 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
         { } /* 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 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = {
         [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 char * const alc262_models[ALC262_MODEL_LAST] = {
 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 @@ static const struct alc_config_preset alc262_presets[] = {
                 .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 @@ static const struct alc_config_preset alc262_presets[] = {
                 .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 be58bf2f3aec..000000000000
--- 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 14fdcf29b154..000000000000
--- 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 e69a6ea3083a..000000000000
--- 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 0eeb227c7bc2..000000000000
--- 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 d719ec6350eb..000000000000
--- 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 8f28450f41f8..000000000000
--- 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 c844d2b59988..bea22edcfd8c 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -749,8 +749,7 @@ static void alc880_uniwill_setup(struct hda_codec *codec)
         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 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec)
 
         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 @@ static void alc880_lg_setup(struct hda_codec *codec)
 
         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 @@ static void alc880_lg_lw_setup(struct hda_codec *codec)
 
         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 @@ static void alc880_medion_rim_automute(struct hda_codec *codec)
         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 @@ static void alc880_medion_rim_setup(struct hda_codec *codec)
 
         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 617d04723b82..e251514a26a4 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -173,8 +173,7 @@ static void alc889_automute_setup(struct hda_codec *codec)
         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 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
         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 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
         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 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
         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 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
         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 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
         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 @@ static void alc885_imac24_setup(struct hda_codec *codec)
         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 @@ static void alc885_imac24_setup(struct hda_codec *codec)
 /* 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 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
 
         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 @@ static void alc885_imac91_setup(struct hda_codec *codec)
         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 @@ static void alc882_targa_automute(struct hda_codec *codec)
         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 @@ static void alc882_targa_setup(struct hda_codec *codec)
 
         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 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec)
 
         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 @@ static void alc883_mitac_setup(struct hda_codec *codec)
         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 @@ static void alc888_3st_hp_setup(struct hda_codec *codec)
         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 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec)
         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 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec)
 
         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 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec)
 
         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 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
 
         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 @@ static void alc883_haier_w66_setup(struct hda_codec *codec)
 
         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 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec)
         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 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec)
         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 @@ static void alc888_6st_dell_setup(struct hda_codec *codec)
         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 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec)
         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 @@ static void alc883_vaiott_setup(struct hda_codec *codec)
         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 @@ static void alc883_mode2_setup(struct hda_codec *codec)
         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 2be1129cf458..a18952ed4311 100644
--- a/sound/pci/hda/alc_quirks.c
+++ b/sound/pci/hda/alc_quirks.c
@@ -453,6 +453,19 @@ static void setup_preset(struct hda_codec *codec,
         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 3e7850c238c3..1715e8b24ff0 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 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
  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 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
                 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 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
         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 @@ static void really_cleanup_stream(struct hda_codec *codec,
                                   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 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
 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 @@ static void hda_power_work(struct work_struct *work)
                 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 @@ void snd_hda_power_up(struct hda_codec *codec)
         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 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
                 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 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
                 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 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
         }
 }
 
+/* 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 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
  * 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 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
         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 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
                 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 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
          * 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 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
          * 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 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
                 }
         }
 
-        /* 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 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 
         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 @@ void snd_array_free(struct snd_array *array)
 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 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid,
                 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 28ce17d09c33..1c8ddf547a2d 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
         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 snd_hdmi_get_eld(struct hdmi_eld *eld,
         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 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
                 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;
-
-        for (i = 0; i < size; i++)
-                buf[i] = hdmi_get_eld_byte(codec, nid, i);
+        /* set ELD buffer */
+        buf = eld->eld_buffer;
+
+        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 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
         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 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a,
                         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 bf3ced51e0f8..7e7d0788ddcf 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -643,14 +643,14 @@ static inline int strmatch(const char *a, const char *b)
 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 @@ static int get_line_from_fw(char *buf, int size, struct firmware *fw)
         }
         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 be6982289c0d..bd7fc99af187 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_param(power_save_controller, bool, 0644);
 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 @@ struct azx_dev {
                                          */
         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 @@ struct azx_dev {
          *  when link position is not greater than FIFO size
          */
         unsigned int insufficient :1;
+        unsigned int wc_marked:1;
 };
 
 /* CORB/RIRB */
@@ -438,6 +460,7 @@ struct azx {
         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 @@ enum {
 #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 @@ static char *driver_short_names[] __devinitdata = {
 /* 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 @@ static int azx_alloc_cmd_io(struct azx *chip)
                 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 update_pci_byte(struct pci_dev *pci, unsigned int reg,
 
 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 @@ static void azx_init_pci(struct azx *chip)
          * 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 @@ static void azx_init_pci(struct azx *chip)
 
         /* 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 void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev)
  */
 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 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
 {
         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 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
         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 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
         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 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
         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_close(struct snd_pcm_substream *substream)
 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 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 
         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_via_get_position(struct azx *chip,
 }
 
 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 @@ static unsigned int azx_get_position(struct azx *chip,
         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 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
         struct azx *chip = apcm->chip;
         struct azx_dev *azx_dev = get_azx_dev(substream);
         return bytes_to_frames(substream->runtime,
-                               azx_get_position(chip, azx_dev));
+                               azx_get_position(chip, azx_dev, false));
 }
 
 /*
@@ -1987,7 +2099,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
                 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 @@ static void azx_clear_irq_pending(struct azx *chip)
         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 @@ static struct snd_pcm_ops azx_pcm_ops = {
         .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 @@ static int azx_free(struct azx *chip)
 
         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 int azx_dev_free(struct snd_device *device)
 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 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
         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 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
                 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 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
                         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 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
                 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 void __devexit azx_remove(struct pci_dev *pci)
 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 2e7ac31afa8d..79f49e2e8cbc 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -267,11 +267,14 @@ int snd_hda_ch_mode_put(struct hda_codec *codec,
 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 @@ int snd_hda_codec_proc_new(struct hda_codec *codec);
 static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
 #endif
 
-#define SND_PRINT_RATES_ADVISED_BUFSIZE 80
-void snd_print_pcm_rates(int pcm, char *buf, int buflen);
-
 #define SND_PRINT_BITS_ADVISED_BUFSIZE  16
 void snd_print_pcm_bits(int pcm, char *buf, int buflen);
 
@@ -385,7 +385,7 @@ enum {
         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 @@ struct auto_pin_cfg {
         (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)
+
+/* 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_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);
+
+/* 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 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
                               unsigned int caps);
 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 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 #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 @@ struct cea_sad {
 };
 
 #define ELD_FIXED_BYTES 20
+#define ELD_MAX_SIZE    256
 #define ELD_MAX_MNL     16
 #define ELD_MAX_SAD     16
 
@@ -631,6 +648,7 @@ struct hdmi_eld {
         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 2be57b051aa2..2c981b55940b 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -152,12 +152,18 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
 
 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 000000000000..9884871ddb00
--- /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 8648917acffb..d8aac588f23b 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -48,6 +48,8 @@ struct ad198x_spec {
 
         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 @@ static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 }
 #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 @@ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
                                     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 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
         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 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
         return 0;
 }
 
-
 /*
  */
 static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
@@ -413,7 +517,8 @@ static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
         .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 @@ static int patch_ad1981(struct hda_codec *codec)
 enum {
         AD1988_6STACK,
         AD1988_6STACK_DIG,
-        AD1988_6STACK_DIG_FP,
         AD1988_3STACK,
         AD1988_3STACK_DIG,
         AD1988_LAPTOP,
@@ -2168,6 +2272,17 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
         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_mixers1_rev2[] = {
 };
 
 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 @@ static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
 
         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_mixers1_rev2[] = {
 };
 
 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 @@ static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
 
 /* 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 @@ static const struct hda_verb ad1988_6stack_init_verbs[] = {
         {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 @@ static const struct hda_verb ad1988_3stack_init_verbs[] = {
         {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 @@ static const struct hda_verb ad1988_laptop_init_verbs[] = {
         {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 inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx)
 {
         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 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec,
         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 int ad1988_auto_init(struct hda_codec *codec)
 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 @@ static int patch_ad1988(struct hda_codec *codec)
         }
         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 @@ static int patch_ad1988(struct hda_codec *codec)
                 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 @@ static int patch_ad1988(struct hda_codec *codec)
                 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 47d6ffc9b5b5..c45f3e69bcf0 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
 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 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
                 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 @@ static int add_volume(struct hda_codec *codec, const char *name,
                       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 502fc9499453..0c8b5a1993ed 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -136,6 +136,8 @@ struct conexant_spec {
         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 @@ static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
         { } /* 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 @@ enum {
         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 @@ static const char *const cxt5051_models[CXT5051_MODELS] = {
         [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 @@ static const struct snd_pci_quirk cxt5051_cfg_tbl[] = {
         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 @@ static int patch_cxt5051(struct hda_codec *codec)
                 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 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
         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 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
 
 #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 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 /* 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 @@ static void cx_auto_parse_output(struct hda_codec *codec)
         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 cx_auto_turn_eapd(struct hda_codec *codec, int num_pins,
 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 @@ static void cx_auto_update_speakers(struct hda_codec *codec)
         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 @@ static void cx_auto_parse_input(struct hda_codec *codec)
         }
         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 @@ static void cx_auto_parse_beep(struct hda_codec *codec)
 #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 @@ static void cx_auto_parse_eapd(struct hda_codec *codec)
                         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 @@ static void cx_auto_init_output(struct hda_codec *codec)
                 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 @@ static void cx_auto_init_output(struct hda_codec *codec)
                 }
         }
         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 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
                              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 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
         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 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
                         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 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
                 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 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
         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 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
                         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 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
         .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 @@ static int patch_conexant_auto(struct hda_codec *codec)
                 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 19cb72db9c38..aac3bfacda3f 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -324,6 +324,66 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid)
         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 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
         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 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 
         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 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
         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 @@ static int generic_hdmi_build_controls(struct hda_codec *codec)
 
         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 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
                                      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 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
 
         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 e125c60fe352..80d6add8a620 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -116,6 +116,8 @@ struct alc_spec {
         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 @@ struct alc_spec {
         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 @@ struct alc_spec {
         /* 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 @@ struct alc_spec {
         /* 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 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
         }
 }
 
-/* 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 @@ static void update_speakers(struct hda_codec *codec)
         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 @@ static void update_speakers(struct hda_codec *codec)
         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 @@ static void alc_line_automute(struct hda_codec *codec)
 {
         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 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
 
         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 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
 {
         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 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
 
         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 @@ static int alc_add_automute_mode_enum(struct hda_codec *codec)
  * 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 @@ static void alc_init_auto_hp(struct hda_codec *codec)
                 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 @@ static void alc_init_auto_hp(struct hda_codec *codec)
                 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 @@ static void alc_init_auto_hp(struct hda_codec *codec)
                 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->detect_hp = 1;
+        }
+
+        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_lines = spec->detect_line;
+                spec->automute_lo_possible = spec->detect_hp;
         }
 
-        if (spec->automute) {
+        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 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 /* 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 @@ do_sku:
          * 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 @@ static void alc_write_coef_idx(struct hda_codec *codec, unsigned int coef_idx,
                             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_init_digital(struct hda_codec *codec)
 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 @@ static const char * const alc_slave_vols[] = {
         "Speaker Playback Volume",
         "Mono Playback Volume",
         "Line-Out Playback Volume",
+        "PCM Playback Volume",
         NULL,
 };
 
@@ -1798,6 +1846,7 @@ static const char * const alc_slave_sws[] = {
         "Mono Playback Switch",
         "IEC958 Playback Switch",
         "Line-Out Playback Switch",
+        "PCM Playback Switch",
         NULL,
 };
 
@@ -2223,6 +2272,7 @@ static int alc_build_pcms(struct hda_codec *codec)
         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 @@ static int alc_build_pcms(struct hda_codec *codec)
         /* 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 @@ static int alc_build_pcms(struct hda_codec *codec)
                                 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 @@ static void alc_free_kctls(struct hda_codec *codec)
         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 @@ static void alc_free(struct hda_codec *codec)
         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 @@ static int alc_codec_rename(struct hda_codec *codec, const char *name)
 }
 
 /*
+ * 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 @@ enum {
         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 @@ static int add_control_with_pfx(struct alc_spec *spec, int type,
 #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 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
                         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 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
         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 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
                                 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 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
                 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 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
         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 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
  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 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
                         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 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
                                 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 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec,
                              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 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec,
                 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 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
                 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 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
         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 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
         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_multi_out(struct hda_codec *codec)
 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 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
 {
         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 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
                 }
         }
         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 @@ static const struct snd_kcontrol_new alc_auto_channel_mode_enum = {
         .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 @@ static int alc_auto_add_multi_channel_mode(struct hda_codec *codec,
                 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 @@ static int alc_parse_auto_config(struct hda_codec *codec,
                                  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 @@ static int patch_alc880(struct hda_codec *codec)
         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 @@ static int patch_alc880(struct hda_codec *codec)
 
         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 @@ static int patch_alc880(struct hda_codec *codec)
 #endif
 
         return 0;
+
+ error:
+        alc_free(codec);
+        return err;
 }
 
 
@@ -3790,10 +4072,8 @@ static int patch_alc260(struct hda_codec *codec)
         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 @@ static int patch_alc260(struct hda_codec *codec)
 
         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 @@ static int patch_alc260(struct hda_codec *codec)
 #endif
 
         return 0;
+
+ error:
+        alc_free(codec);
+        return err;
 }
 
 
@@ -3865,6 +4147,7 @@ enum {
         PINFIX_LENOVO_Y530,
         PINFIX_PB_M5210,
         PINFIX_ACER_ASPIRE_7736,
+        PINFIX_ASUS_W90V,
 };
 
 static const struct alc_fixup alc882_fixups[] = {
@@ -3896,10 +4179,18 @@ static const struct alc_fixup alc882_fixups[] = {
                 .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 @@ static int patch_alc882(struct hda_codec *codec)
                 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 @@ static int patch_alc882(struct hda_codec *codec)
         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 @@ static int patch_alc882(struct hda_codec *codec)
 
         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 @@ static int patch_alc882(struct hda_codec *codec)
 #endif
 
         return 0;
+
+ error:
+        alc_free(codec);
+        return err;
 }
 
 
@@ -4123,10 +4418,8 @@ static int patch_alc262(struct hda_codec *codec)
         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 @@ static int patch_alc262(struct hda_codec *codec)
 
         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 @@ static int patch_alc262(struct hda_codec *codec)
 #endif
 
         return 0;
+
+ error:
+        alc_free(codec);
+        return err;
 }
 
 /*
@@ -4222,14 +4517,9 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 
 /*
  */
-#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 @@ static int patch_alc268(struct hda_codec *codec)
 
         /* 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 @@ static int patch_alc268(struct hda_codec *codec)
 
         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 @@ static int patch_alc268(struct hda_codec *codec)
         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_toggle_power_output(struct hda_codec *codec, int power_up)
 
 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 @@ static void alc269_shutup(struct hda_codec *codec)
 #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 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
         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 @@ enum {
         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 @@ static const struct alc_fixup alc269_fixups[] = {
                 .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 @@ static int alc269_fill_coef(struct hda_codec *codec)
 {
         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 @@ static int alc269_fill_coef(struct hda_codec *codec)
 
 /*
  */
-#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 @@ static int patch_alc269(struct hda_codec *codec)
 
         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);
-
-        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);
-        }
+        alc_pick_fixup(codec, alc269_fixup_models,
+                       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 @@ static int patch_alc269(struct hda_codec *codec)
 
         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 @@ static int patch_alc269(struct hda_codec *codec)
 #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 @@ static int patch_alc269(struct hda_codec *codec)
 #endif
 
         return 0;
+
+ error:
+        alc_free(codec);
+        return err;
 }
 
 /*
@@ -4794,14 +5190,9 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = {
 
 /*
  */
-#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 @@ static int patch_alc861(struct hda_codec *codec)
 
         spec->mixer_nid = 0x15;
 
-        board_config = alc_board_config(codec, ALC861_MODEL_LAST,
-                                        alc861_models, alc861_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) {
-                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
-        }
+        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)
-                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 @@ static int patch_alc861(struct hda_codec *codec)
 
         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 @@ static int patch_alc861(struct hda_codec *codec)
         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;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-                spec->power_hook = alc_power_eapd;
-#endif
-        }
+        spec->init_hook = alc_auto_init_std;
 #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 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 }
 
 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 @@ static const struct hda_verb alc660vd_eapd_verbs[] = {
 
 /*
  */
-#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 @@ static int patch_alc861vd(struct hda_codec *codec)
 
         spec->mixer_nid = 0x0b;
 
-        board_config = alc_board_config(codec, ALC861VD_MODEL_LAST,
-                                        alc861vd_models, alc861vd_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) {
-                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
-        }
+        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)
-                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 @@ static int patch_alc861vd(struct hda_codec *codec)
 
         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 @@ static int patch_alc861vd(struct hda_codec *codec)
 
         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 @@ static int patch_alc861vd(struct hda_codec *codec)
 #endif
 
         return 0;
+
+ error:
+        alc_free(codec);
+        return err;
 }
 
 /*
@@ -5077,6 +5428,14 @@ enum {
         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 @@ static const struct alc_fixup alc662_fixups[] = {
                 .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 @@ static int patch_alc662(struct hda_codec *codec)
 
         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");
-
-        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;
-        }
+        err = alc_codec_rename_from_preset(codec);
+        if (err < 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 ((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)
-                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 @@ static int patch_alc662(struct hda_codec *codec)
 
         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 @@ static int patch_alc662(struct hda_codec *codec)
         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 @@ static int patch_alc662(struct hda_codec *codec)
 #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 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
 
 /*
  */
-#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 @@ static int patch_alc680(struct hda_codec *codec)
 
         /* 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 @@ static int patch_alc680(struct hda_codec *codec)
         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 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
           .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 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
         { .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 aa376b59c006..de4c36027cbe 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
         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 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
         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 int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
 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 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
                         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 is_dual_headphones(struct hda_codec *codec)
 }
 
 
-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 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
         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 @@ static int stac92xx_add_jack(struct hda_codec *codec,
 #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 @@ static int patch_stac925x(struct hda_codec *codec)
         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 @@ again:
         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 void stac92hd8x_fill_auto_spec(struct hda_codec *codec)
 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 @@ again:
                 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 @@ again:
         }
 #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 @@ again:
                 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 @@ again:
 
         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 @@ static int patch_stac922x(struct hda_codec *codec)
 
         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 @@ static int patch_stac927x(struct hda_codec *codec)
         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 @@ static int patch_stac9205(struct hda_codec *codec)
                 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 @@ static int patch_stac9872(struct hda_codec *codec)
         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 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
         { .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 84d8798bf33a..0b020a93a8ed 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1506,39 +1506,49 @@ static int via_build_pcms(struct hda_codec *codec)
         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;
+                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++;
         }
-        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) {
-                codec->num_pcms++;
-                info++;
                 snprintf(spec->stream_name_digital,
                          sizeof(spec->stream_name_digital),
                          "%s Digital", codec->chip_name);
@@ -1562,17 +1572,19 @@ static int via_build_pcms(struct hda_codec *codec)
                         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 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
         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 const struct hda_verb vt1812_init_verbs[] = {
 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 0ccc0eb75775..8531b983f3af 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2748,8 +2748,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
         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 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
                         /*  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 6a5b387b97fd..45b2055f5a76 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_semaphore;
 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_param(xbox, bool, 0444);
 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 @@ struct intel8x0 {
         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 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs
                         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 @@ static int __devinit snd_intel8x0_create(struct snd_card *card,
         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 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci,
                         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 0378126e6272..2fd4bf2d6653 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2820,8 +2820,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
         /* 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 82311fcb86f6..53e5508abcbf 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -678,15 +678,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
                 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 32d096c98f5b..8433aa7c3d75 100644
--- a/sound/pci/oxygen/xonar_pcm179x.c
+++ b/sound/pci/oxygen/xonar_pcm179x.c
@@ -1074,6 +1074,7 @@ static const struct oxygen_model model_xonar_st = {
         .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 e34ae14908b3..88cc776aa38b 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2109,7 +2109,7 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
                 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 1c6d1e1c27c1..f74220292254 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -151,7 +151,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #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 6edc67ced905..15a6c3b9bc9a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -520,16 +520,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #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 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
         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 @@ static int hdspm_set_interrupt_interval(struct hdspm *s, unsigned int frames)
 
         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 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
                 break;
         case MADIface:
                 freq_const = 131072000000000ULL;
+                break;
+        default:
+                snd_BUG();
+                return 0;
         }
 
         return div_u64(freq_const, period);
@@ -1356,16 +1388,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
 
         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_hdspm_proc_read_madi(struct snd_info_entry * entry,
 
         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_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 
         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 @@ static int snd_hdspm_prepare(struct snd_pcm_substream *substream)
         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 @@ static struct snd_pcm_hardware snd_hdspm_playback_subinfo = {
         .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 @@ static struct snd_pcm_hardware snd_hdspm_capture_subinfo = {
         .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 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
         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 @@ static int snd_hdspm_capture_open(struct snd_pcm_substream *substream)
         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 inline int copy_u32_le(void __user *dest, void __iomem *src)
 }
 
 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 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
                 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 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
                         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 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
                         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 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
                 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 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
         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 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
                 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 bcf61524a13f..5ffb20b18786 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1234,7 +1234,7 @@ static int sis_resume(struct pci_dev *pci)
                 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 @@ static int __devinit sis_chip_create(struct snd_card *card,
         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 2571a67b389a..c5008166cf1f 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1493,9 +1493,10 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci,
                 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 d8a128f6fc02..5e707effdc7c 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -148,8 +148,9 @@ static int __devinit snd_trident_probe(struct pci_dev *pci,
         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 f03fd620a2a0..c3656fffdb50 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1175,6 +1175,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
         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 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
                                      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 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
         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 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip)
         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 511d57653124..3253b04da184 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -305,8 +305,9 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
         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 f3260e658b8a..66ea71b2a70d 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -897,6 +897,18 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream)
         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 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream)
         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 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream,
         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 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream,
         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 Playback Volume", 0, YDSXGR_PRIADCOUTVOL),
 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 8f064c7ce745..4080becf4cef 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -82,7 +82,6 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
 
 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 bc823a547550..775bd95d4be6 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -845,7 +845,7 @@ static int __devinit snd_ps3_allocate_irq(void)
                 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 7aa5b7606777..177f7137a9c8 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -211,7 +211,7 @@ static int alchemy_pcm_open(struct snd_pcm_substream *substream)
         /* 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 3f4920d5456d..dc8a2b2bdc1c 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1419,7 +1419,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
         /* 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 4e3626b9d8f9..ae8d6806966b 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -268,7 +268,7 @@ static int nuc900_dma_open(struct snd_pcm_substream *substream)
         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 65ea53884806..b5e922f469d5 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -444,7 +444,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev)
         }
 
         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 916b9f99b7e7..a32fd16ad668 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1285,7 +1285,7 @@ static int fsi_probe(struct platform_device *pdev)
         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 743d07b82c06..a4e3f5501847 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -201,7 +201,7 @@ static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev)
         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 ad7d4d7d9237..f036776380b5 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -962,7 +962,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
         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 1e3ae3327dd3..07bcfe4d18a7 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 @@ struct ihex_record {
         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 8beb77563da2..3efc21c3d67c 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -67,6 +67,7 @@ config SND_USB_CAIAQ
             * 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 @@ config SND_USB_CAIAQ_INPUT
            * 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 cf9ed66445fa..ac256dc4c6be 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 d0d493ca28ae..2cf87f5afed4 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
 
         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 @@ static void read_completed(struct urb *urb)
 {
         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 @@ static void read_completed(struct urb *urb)
         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 @@ static void read_completed(struct urb *urb)
                 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 @@ static void read_completed(struct urb *urb)
         }
 
         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 @@ static void write_completed(struct urb *urb)
                 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 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
         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 45bc4a2dc6f0..3eb605bd9503 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -50,7 +50,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
                          "{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 @@ static struct usb_device_id snd_usb_id_table[] = {
                 .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 b2b310194ffa..562b0bff9c41 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 @@ struct snd_usb_caiaqdev {
         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 4432ef7a70a9..26a121b42c3c 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] =  { KEY_C, KEY_B, KEY_A };
 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 @@ static unsigned short keycode_kore[] = {
         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 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev,
                 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 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev,
         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 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb)
         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 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev)
         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 @@ static void snd_usb_caiaq_input_close(struct input_dev *idev)
         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 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev)
 
                 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 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev)
         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 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev)
         input_unregister_device(dev->input_dev);
         dev->input_dev = NULL;
 }
-
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 781d9e61adfb..c1575eafff12 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 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int
                 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 @@ snd_usb_audio_probe(struct usb_device *dev,
         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 ae4251d5abf7..a39edcc32a93 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -94,6 +94,8 @@ struct snd_usb_substream {
         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 075195e8661a..379baad3d5ad 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -91,7 +91,7 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i
                               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 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
                               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 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
         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 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
         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 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
                                    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 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
                                    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 7c0d21ecd821..81c6edecd862 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 "helper.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"
 
 /*
- * 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;
 }
 
+/*
+ * 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;
+}
 
 /*
- * free a usb stream instance
+ * unlink active urbs.
  */
-static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
+static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
 {
-        free_substream(&stream->substream[0]);
-        free_substream(&stream->substream[1]);
-        list_del(&stream->list);
-        kfree(stream);
+        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;
 }
 
-static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
+
+/*
+ * release a urb data
+ */
+static void release_urb_ctx(struct snd_urb_ctx *u)
 {
-        struct snd_usb_stream *stream = pcm->private_data;
-        if (stream) {
-                stream->pcm = NULL;
-                snd_usb_audio_stream_free(stream);
+        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;
+}
 
 /*
- * 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.
+ * release a substream
  */
-int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp)
+void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
 {
-        struct list_head *p;
-        struct snd_usb_stream *as;
-        struct snd_usb_substream *subs;
-        struct snd_pcm *pcm;
-        int err;
+        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;
+}
 
-        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;
+/*
+ * 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);
                 }
         }
-        /* 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;
+}
+
+
+/*
+ * 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);
+                }
         }
+}
+
 
-        /* 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;
+/*
+ * 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);
         }
-        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);
+        subs->phase = 0;
+
+        if (subs->fill_max)
+                subs->curpacksize = subs->maxpacksize;
         else
-                strcpy(pcm->name, "USB Audio");
+                subs->curpacksize = maxsize;
 
-        snd_usb_init_substream(as, stream, fp);
+        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;
+        }
 
-        list_add(&as->list, &chip->pcm_list);
-        chip->pcm_devs++;
+        /* 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;
 
-        snd_usb_proc_pcm_format_add(as);
+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;
 }
 
-static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
-                                         struct usb_host_interface *alts,
-                                         int protocol, int iface_no)
+/*
+ * 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)
 {
-        /* 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;
-        }
+        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;
+}
 
-        if (protocol == UAC_VERSION_1) {
-                attributes = csep->bmAttributes;
-        } else {
-                struct uac2_iso_endpoint_descriptor *csep2 =
-                        (struct uac2_iso_endpoint_descriptor *) csep;
+/*
+ * 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;
+}
 
-                attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
+/*
+ * 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;
+}
 
-                /* emulate the endpoint attributes of a v1 device */
-                if (csep2->bmControls & UAC2_CONTROL_PITCH)
-                        attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
+/*
+ * 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;
+}
 
-        return attributes;
+/*
+ * 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;
 }
 
-static struct uac2_input_terminal_descriptor *
-        snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-                                               int terminal_id)
+
+/*
+ * 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 uac2_input_terminal_descriptor *term = NULL;
+        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;
+}
 
-        while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
-                                               ctrl_iface->extralen,
-                                               term, UAC_INPUT_TERMINAL))) {
-                if (term->bTerminalID == terminal_id)
-                        return term;
+/*
+ * 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 NULL;
+        return 0;
 }
 
-static struct uac2_output_terminal_descriptor *
-        snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-                                                int terminal_id)
+/* determine the number of frames in the next packet */
+static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
 {
-        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;
+        if (subs->fill_max)
+                return subs->maxframesize;
+        else {
+                subs->phase = (subs->phase & 0xffff)
+                        + (subs->freqm << subs->datainterval);
+                return min(subs->phase >> 16, subs->maxframesize);
         }
+}
 
-        return NULL;
+/*
+ * 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;
 }
 
-int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
+/*
+ * 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)
 {
-        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;
+        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;
+}
 
-        dev = chip->dev;
+/*
+ * 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;
 
-        /* parse the interface's altsettings */
-        iface = usb_ifnum_to_if(dev, iface_no);
+        spin_lock_irqsave(&subs->lock, flags);
 
-        num = iface->num_altsetting;
+        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;
 
         /*
-         * Dallas DS4201 workaround: It presents 5 altsettings, but the last
-         * one misses syncpipe, and does not produce any sound.
+         * 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 (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 (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);
 
-                        if (as->bLength < sizeof(*as)) {
-                                snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
-                                           dev->devnum, iface_no, altno);
-                                continue;
-                        }
+        spin_unlock_irqrestore(&subs->lock, flags);
+        return 0;
+}
 
-                        format = le16_to_cpu(as->wFormatTag); /* remember the format value */
-                        break;
-                }
+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";
+        }
+}
 
-                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);
+/*
+ * 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 (!as) {
-                                snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
-                                           dev->devnum, iface_no, altno);
-                                continue;
+        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;
                         }
+                }
+        }
 
-                        if (as->bLength < sizeof(*as)) {
-                                snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
-                                           dev->devnum, iface_no, altno);
-                                continue;
+        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;
 
-                        num_channels = as->bNrChannels;
-                        format = le32_to_cpu(as->bmFormats);
+ __error:
+        // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
+        deactivate_urbs(subs, 0, 0);
+        return -EPIPE;
+}
 
-                        /* 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;
-                        }
+/*
+ */
+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,
+        },
+};
 
-                        snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
-                                   dev->devnum, iface_no, altno, as->bTerminalLink);
-                        continue;
-                }
-                }
+/*
+ * initialize the substream instance.
+ */
 
-                /* 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;
-                }
+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;
+}
 
-                /*
-                 * 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;
-                }
+int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct snd_usb_substream *subs = substream->runtime->private_data;
 
-                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;
-                }
+        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;
+        }
 
-                /* 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;
-                }
+        return -EINVAL;
+}
 
-                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);
+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/endpoint.h b/sound/usb/endpoint.h
index 64dd0db023b2..88eb63a636eb 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 8d042dce0d16..89421d176570 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -286,7 +286,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
                               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 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
                               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 f280c1903c25..9eed8f40b179 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -81,7 +81,7 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype
  */
 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 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request,
                         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 09bd943c43bf..805c300dd004 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -8,7 +8,7 @@ void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsub
 
 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 f9289102886a..e21f026d9577 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -816,6 +816,22 @@ static struct usb_protocol_ops snd_usbmidi_raw_ops = {
         .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 @@ int snd_usbmidi_create(struct snd_card *card,
                 /* 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 c22fa76e363a..60f65ace7474 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
         if (p && p->dB) {
                 cval->dBmin = p->dB->min;
                 cval->dBmax = p->dB->max;
+                cval->initialized = 1;
         }
 }
 
@@ -295,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v
                 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 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v
         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 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
                                     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 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_
                 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 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
                                 " 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 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 
         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 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
                                 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 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
         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 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
         }
 
         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 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
 /* 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 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
         };
         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 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
                 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 ae1a14dcfe82..81b2d8a32fb0 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 3d0f4873112b..ab125ee0b0f0 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
                 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 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry,
                                       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 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol,
         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 b8dcbf407bbb..0220b0f335b9 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 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream
         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 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
         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 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface,
         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 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
         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 ed3e283f618d..df7a003682ad 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 dba0b7f11c54..b61945f3af9e 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 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         }
 },
 {
+        /* 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 @@ YAMAHA_DEVICE(0x7010, "UB99"),
                 }
         }
 },
+{
+        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 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         .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 77762c99afbe..2e5bc7344026 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 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip,
 
         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 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 
         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 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 
         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 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
                 [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 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac
                 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 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
 
         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 @@ static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 valu
         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_cm106_boot_quirk(struct usb_device *dev)
  */
 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 000000000000..5ff8010b2d6f
--- /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 000000000000..c97f679fc84f
--- /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 e184349aee83..000000000000
--- 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 888da38079cf..000000000000
--- 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 1e79986b5777..3e2b03577936 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -80,6 +80,7 @@ enum quirk_type {
         QUIRK_MIDI_CME,
         QUIRK_MIDI_AKAI,
         QUIRK_MIDI_US122L,
+        QUIRK_MIDI_FTDI,
         QUIRK_AUDIO_STANDARD_INTERFACE,
         QUIRK_AUDIO_FIXED_ENDPOINT,
         QUIRK_AUDIO_EDIROL_UAXX,