Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel…

…/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (297 commits)
  ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
  ALSA: asihpi - HPI version 4.08
  ALSA: asihpi - Add volume mute controls
  ALSA: asihpi - Control name updates
  ALSA: asihpi - Use size_t for sizeof result
  ALSA: asihpi - Explicitly include mutex.h
  ALSA: asihpi - Add new node and message defines
  ALSA: asihpi - Make local function static
  ALSA: asihpi - Fix minor typos and spelling
  ALSA: asihpi - Remove unused structures, macros and functions
  ALSA: asihpi - Remove spurious adapter index check
  ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
  ALSA: asihpi - DSP code loader API now independent of OS
  ALSA: asihpi - Remove controlex structs and associated special data transfer code
  ALSA: asihpi - Increase request and response buffer sizes
  ALSA: asihpi - Give more meaningful name to hpi request message type
  ALSA: usb-audio - Add quirk for  Roland / BOSS BR-800
  ALSA: hda - Remove a superfluous argument of via_auto_init_output()
  ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
  ALSA: hda - Add documentation for codec-specific mixer controls
  ...

Documentation/DocBook/writing-an-alsa-driver.tmpl

@@ -1164,7 +1164,7 @@
           }
           chip->port = pci_resource_start(pci, 0);
           if (request_irq(pci->irq, snd_mychip_interrupt,
-                          IRQF_SHARED, "My Chip", chip)) {
+                          IRQF_SHARED, KBUILD_MODNAME, chip)) {
                   printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
                   snd_mychip_free(chip);
                   return -EBUSY;
@@ -1197,7 +1197,7 @@
 
   /* pci_driver definition */
   static struct pci_driver driver = {
-          .name = "My Own Chip",
+          .name = KBUILD_MODNAME,
           .id_table = snd_mychip_ids,
           .probe = snd_mychip_probe,
           .remove = __devexit_p(snd_mychip_remove),
@@ -1340,7 +1340,7 @@
           <programlisting>
 <![CDATA[
   if (request_irq(pci->irq, snd_mychip_interrupt,
-                  IRQF_SHARED, "My Chip", chip)) {
+                  IRQF_SHARED, KBUILD_MODNAME, chip)) {
           printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
           snd_mychip_free(chip);
           return -EBUSY;
@@ -1616,7 +1616,7 @@
           <programlisting>
 <![CDATA[
   static struct pci_driver driver = {
-          .name = "My Own Chip",
+          .name = KBUILD_MODNAME,
           .id_table = snd_mychip_ids,
           .probe = snd_mychip_probe,
           .remove = __devexit_p(snd_mychip_remove),
@@ -5816,7 +5816,7 @@ struct _snd_pcm_runtime {
         <programlisting>
 <![CDATA[
   static struct pci_driver driver = {
-          .name = "My Chip",
+          .name = KBUILD_MODNAME,
           .id_table = snd_my_ids,
           .probe = snd_my_probe,
           .remove = __devexit_p(snd_my_remove),

Documentation/sound/alsa/HD-Audio-Controls.txt

@@ -0,0 +1,100 @@
+This file explains the codec-specific mixer controls.
+
+Realtek 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", "6ch",
+  and "8ch".  According to the configuration, this also controls the
+  jack-retasking of multi-I/O jacks.
+
+* Auto-Mute Mode
+  This is an enum control to change the auto-mute behavior of the
+  headphone and line-out jacks.  If built-in speakers and headphone
+  and/or line-out jacks are available on a machine, this controls
+  appears.
+  When there are only either headphones or line-out jacks, it gives
+  "Disabled" and "Enabled" state.  When enabled, the speaker is muted
+  automatically when a jack is plugged.
+
+  When both headphone and line-out jacks are present, it gives
+  "Disabled", "Speaker Only" and "Line-Out+Speaker".  When
+  speaker-only is chosen, plugging into a headphone or a line-out jack
+  mutes the speakers, but not line-outs.  When line-out+speaker is
+  selected, plugging to a headphone jack mutes both speakers and
+  line-outs.
+
+
+IDT/Sigmatel codecs
+-------------------
+
+* Analog Loopback
+  This control enables/disables the analog-loopback circuit.  This
+  appears only when "loopback" is set to true in a codec hint
+  (see HD-Audio.txt).  Note that on some codecs the analog-loopback
+  and the normal PCM playback are exclusive, i.e. when this is on, you
+  won't hear any PCM stream.
+
+* Swap Center/LFE
+  Swaps the center and LFE channel order.  Normally, the left
+  corresponds to the center and the right to the LFE.  When this is
+  ON, the left to the LFE and the right to the center.
+
+* Headphone as Line Out
+  When this control is ON, treat the headphone jacks as line-out
+  jacks.  That is, the headphone won't auto-mute the other line-outs,
+  and no HP-amp is set to the pins.
+
+* Mic Jack Mode, Line Jack Mode, etc
+  These enum controls the direction and the bias of the input jack
+  pins.  Depending on the jack type, it can set as "Mic In" and "Line 
+  In", for determining the input bias, or it can be set to "Line Out"
+  when the pin is a multi-I/O jack for surround channels.
+
+
+VIA codecs
+----------
+
+* Smart 5.1
+  An enum control to re-task the multi-I/O jacks for surround outputs.
+  When it's ON, the corresponding input jacks (usually a line-in and a
+  mic-in) are switched as the surround and the CLFE output 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.  In the case the headphone DAC is shared with a
+  side or a CLFE-channel DAC, the DAC is switched to the headphone
+  automatically.
+
+* Loopback Mixing
+  An enum control to determine whether the analog-loopback route is
+  enabled or not.  When it's enabled, the analog-loopback is mixed to
+  the front-channel.  Also, the same route is used for the headphone
+  and speaker outputs.  As a side-effect, when this mode is set, the
+  individual volume controls will be no longer available for
+  headphones and speakers because there is only one DAC connected to a
+  mixer widget.
+
+* Dynamic Power-Control
+  This control determines whether the dynamic power-control per jack
+  detection is enabled or not.  When enabled, the widgets power state
+  (D0/D3) are changed dynamically depending on the jack plugging
+  state for saving power consumptions.  However, if your system
+  doesn't provide a proper jack-detection, this won't work; in such a
+  case, turn this control OFF.
+
+* Jack Detect
+  This control is provided only for VT1708 codec which gives no proper
+  unsolicited event per jack plug.  When this is on, the driver polls
+  the jack detection so that the headphone auto-mute can work, while 
+  turning this off would reduce the power consumption.
+
+
+Conexant codecs
+---------------
+
+* Auto-Mute Mode
+  See Reatek codecs.

MAINTAINERS

@@ -534,6 +534,8 @@ L:	[email protected]
 L:	[email protected] (moderated for non-subscribers)
 W:	http://wiki.analog.com/
 S:	Supported
+F:	sound/soc/codecs/adau*
+F:	sound/soc/codecs/adav*
 F:	sound/soc/codecs/ad1*
 F:	sound/soc/codecs/ssm*
 

include/linux/pci_ids.h

@@ -1308,6 +1308,7 @@
 #define PCI_SUBDEVICE_ID_CREATIVE_SB08801	0x0041
 #define PCI_SUBDEVICE_ID_CREATIVE_SB08802	0x0042
 #define PCI_SUBDEVICE_ID_CREATIVE_SB08803	0x0043
+#define PCI_SUBDEVICE_ID_CREATIVE_SB1270	0x0062
 #define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX	0x6000
 
 #define PCI_VENDOR_ID_ECTIVA		0x1102 /* duplicate: CREATIVE */

include/sound/rawmidi.h

@@ -27,6 +27,7 @@
 #include <linux/spinlock.h>
 #include <linux/wait.h>
 #include <linux/mutex.h>
+#include <linux/workqueue.h>
 
 #if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
 #include "seq_device.h"
@@ -63,6 +64,7 @@ struct snd_rawmidi_global_ops {
 };
 
 struct snd_rawmidi_runtime {
+	struct snd_rawmidi_substream *substream;
 	unsigned int drain: 1,	/* drain stage */
 		     oss: 1;	/* OSS compatible mode */
 	/* midi stream buffer */
@@ -79,7 +81,7 @@ struct snd_rawmidi_runtime {
 	/* event handler (new bytes, input only) */
 	void (*event)(struct snd_rawmidi_substream *substream);
 	/* defers calls to event [input] or ops->trigger [output] */
-	struct tasklet_struct tasklet;
+	struct work_struct event_work;
 	/* private data */
 	void *private_data;
 	void (*private_free)(struct snd_rawmidi_substream *substream);

include/sound/soc-dai.h

@@ -209,6 +209,10 @@ struct snd_soc_dai_driver {
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
 	unsigned int symmetric_rates:1;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /*

include/sound/soc-dapm.h

@@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+			     const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@@ -429,6 +431,7 @@ struct snd_soc_dapm_path {
 	/* status */
 	u32 connect:1;	/* source and sink widgets are connected */
 	u32 walked:1;	/* path has been walked */
+	u32 weak:1;	/* path ignored for power management */
 
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink);
@@ -444,6 +447,7 @@ struct snd_soc_dapm_widget {
 	char *name;		/* widget name */
 	char *sname;	/* stream name */
 	struct snd_soc_codec *codec;
+	struct snd_soc_platform *platform;
 	struct list_head list;
 	struct snd_soc_dapm_context *dapm;
 
@@ -507,10 +511,11 @@ struct snd_soc_dapm_context {
 
 	struct device *dev; /* from parent - for debug */
 	struct snd_soc_codec *codec; /* parent codec */
+	struct snd_soc_platform *platform; /* parent platform */
 	struct snd_soc_card *card; /* parent card */
 
 	/* used during DAPM updates */
-	int dev_power;
+	enum snd_soc_bias_level target_bias_level;
 	struct list_head list;
 
 #ifdef CONFIG_DEBUG_FS

include/sound/soc.h

@@ -202,6 +202,16 @@
 #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
 	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
 
+/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST		-2
+#define SND_SOC_COMP_ORDER_EARLY		-1
+#define SND_SOC_COMP_ORDER_NORMAL		0
+#define SND_SOC_COMP_ORDER_LATE		1
+#define SND_SOC_COMP_ORDER_LAST		2
+
 /*
  * Bias levels
  *
@@ -214,10 +224,10 @@
  * @OFF:     Power Off. No restrictions on transition times.
  */
 enum snd_soc_bias_level {
-	SND_SOC_BIAS_OFF,
-	SND_SOC_BIAS_STANDBY,
-	SND_SOC_BIAS_PREPARE,
-	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_OFF = 0,
+	SND_SOC_BIAS_STANDBY = 1,
+	SND_SOC_BIAS_PREPARE = 2,
+	SND_SOC_BIAS_ON = 3,
 };
 
 struct snd_jack;
@@ -258,6 +268,11 @@ enum snd_soc_compress_type {
 	SND_SOC_RBTREE_COMPRESSION
 };
 
+enum snd_soc_pcm_subclass {
+	SND_SOC_PCM_CLASS_PCM	= 0,
+	SND_SOC_PCM_CLASS_BE	= 1,
+};
+
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			     unsigned int freq, int dir);
 int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -297,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
 int snd_soc_default_writable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+					unsigned int reg);
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+					unsigned int reg, unsigned int val);
 
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@@ -349,6 +368,8 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 				  const char *prefix);
 int snd_soc_add_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -612,6 +633,10 @@ struct snd_soc_codec_driver {
 
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /* SoC platform interface */
@@ -623,10 +648,17 @@ struct snd_soc_platform_driver {
 	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
-		struct snd_pcm *);
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
+	/* Default control and setup, added after probe() is run */
+	const struct snd_kcontrol_new *controls;
+	int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	int num_dapm_routes;
+
 	/*
 	 * For platform caused delay reporting.
 	 * Optional.
@@ -636,6 +668,14 @@ struct snd_soc_platform_driver {
 
 	/* platform stream ops */
 	struct snd_pcm_ops *ops;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
+
+	/* platform IO - used for platform DAPM */
+	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
+	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
 };
 
 struct snd_soc_platform {
@@ -650,6 +690,8 @@ struct snd_soc_platform {
 	struct snd_soc_card *card;
 	struct list_head list;
 	struct list_head card_list;
+
+	struct snd_soc_dapm_context dapm;
 };
 
 struct snd_soc_dai_link {
@@ -725,8 +767,10 @@ struct snd_soc_card {
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_card *,
+			      struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 	int (*set_bias_level_post)(struct snd_soc_card *,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
 	long pmdown_time;
@@ -789,6 +833,9 @@ struct snd_soc_pcm_runtime  {
 	struct device dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
+	struct mutex pcm_mutex;
+	enum snd_soc_pcm_subclass pcm_subclass;
+	struct snd_pcm_ops ops;
 
 	unsigned int complete:1;
 	unsigned int dev_registered:1;