Merge branch 'for-linus' of git://git.alsa-project.org/alsa-kernel

* 'for-linus' of git://git.alsa-project.org/alsa-kernel: (179 commits)
  ALSA: Release v1.0.17
  ALSA: correct kcalloc usage
  ALSA: ALSA driver for SGI O2 audio board
  ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform.
  ALSA: ALSA driver for SGI HAL2 audio device
  ALSA: hda - Fix FSC V5505 model
  ALSA: hda - Fix missing init for unsol events on micsense model
  ALSA: hda - Fix internal mic vref pin setup
  ALSA: hda: 92hd71bxx PC Beep
  ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model
  ALSA: usb-audio: add some Yamaha USB MIDI quirks
  ALSA: usb-audio: fix Yamaha KX quirk
  ALSA: ASoC: Au12x0/Au1550 PSC Audio support
  ALSA: Add Yamaha KX49 (USB MIDI controller) to usbquirks.h
  ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration
  ALSA: tosa: fix compilation with new DAPM API
  ALSA: wavefront - add const
  ALSA: remove CONFIG_KMOD from sound
  ALSA: Fix a const to non-const assignment in the Digigram VXpocket sound driver
  ALSA: Fix a const pointer usage warning in the Digigram VX soundcard driver
  ...

Documentation/sound/alsa/ALSA-Configuration.txt

@@ -753,8 +753,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     [Multiple options for each card instance]
     model	- force the model name
-    position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)
+    position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
     probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
+    bdl_pos_adj	- Specifies the DMA IRQ timing delay in samples.
+		Passing -1 will make the driver to choose the appropriate
+		value based on the controller chip.
 
     [Single (global) options]
     single_cmd  - Use single immediate commands to communicate with
@@ -845,14 +848,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	ALC269
 	  basic		Basic preset
 
-	ALC662
+	ALC662/663
 	  3stack-dig	3-stack (2-channel) with SPDIF
 	  3stack-6ch	 3-stack (6-channel)
 	  3stack-6ch-dig 3-stack (6-channel) with SPDIF
 	  6stack-dig	 6-stack with SPDIF
 	  lenovo-101e	 Lenovo laptop
 	  eeepc-p701	ASUS Eeepc P701
 	  eeepc-ep20	ASUS Eeepc EP20
+	  m51va		ASUS M51VA
+	  g71v		ASUS G71V
+	  h13		ASUS H13
+	  g50v		ASUS G50V
 	  auto		auto-config reading BIOS (default)
 
 	ALC882/885
@@ -1091,7 +1098,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     This occurs when the access to non-existing or non-working codec slot
     (likely a modem one) causes a stall of the communication via HD-audio
     bus.  You can see which codec slots are probed by enabling
-    CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec
+    CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec
     proc files.  Then limit the slots to probe by probe_mask option.
     For example, probe_mask=1 means to probe only the first slot, and
     probe_mask=4 means only the third slot.
@@ -2267,6 +2274,10 @@ case above again, the first two slots are already reserved.  If any
 other driver (e.g. snd-usb-audio) is loaded before snd-interwave or
 snd-ens1371, it will be assigned to the third or later slot.
 
+When a module name is given with '!', the slot will be given for any
+modules but that name.  For example, "slots=!snd-pcsp" will reserve
+the first slot for any modules but snd-pcsp. 
+
 
 ALSA PCM devices to OSS devices mapping
 =======================================

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

@@ -6127,8 +6127,8 @@ struct _snd_pcm_runtime {
 
       <para>
         <function>snd_printdd()</function> is compiled in only when
-      <constant>CONFIG_SND_DEBUG_DETECT</constant> is set. Please note
-      that <constant>DEBUG_DETECT</constant> is not set as default
+      <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is set. Please note
+      that <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is not set as default
       even if you configure the alsa-driver with
       <option>--with-debug=full</option> option. You need to give
       explicitly <option>--with-debug=detect</option> option instead. 

include/asm-mips/mach-au1x00/au1xxx_psc.h

@@ -204,6 +204,14 @@ typedef struct	psc_i2s {
 	u32	psc_i2sudf;
 } psc_i2s_t;
 
+#define PSC_I2SCFG_OFFSET	0x08
+#define PSC_I2SMASK_OFFSET	0x0C
+#define PSC_I2SPCR_OFFSET	0x10
+#define PSC_I2SSTAT_OFFSET	0x14
+#define PSC_I2SEVENT_OFFSET	0x18
+#define PSC_I2SRXTX_OFFSET	0x1C
+#define PSC_I2SUDF_OFFSET	0x20
+
 /* I2S Config Register. */
 #define PSC_I2SCFG_RT_MASK	(3 << 30)
 #define PSC_I2SCFG_RT_FIFO1	(0 << 30)

include/sound/ad1843.h

@@ -0,0 +1,46 @@
+/*
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License.  See the file "COPYING" in the main directory of this archive
+ * for more details.
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ * Copyright 2008 Thomas Bogendoerfer <tsbogend@franken.de>
+ */
+
+#ifndef __SOUND_AD1843_H
+#define __SOUND_AD1843_H
+
+struct snd_ad1843 {
+	void *chip;
+	int (*read)(void *chip, int reg);
+	int (*write)(void *chip, int reg, int val);
+};
+
+#define AD1843_GAIN_RECLEV 0
+#define AD1843_GAIN_LINE   1
+#define AD1843_GAIN_LINE_2 2
+#define AD1843_GAIN_MIC    3
+#define AD1843_GAIN_PCM_0  4
+#define AD1843_GAIN_PCM_1  5
+#define AD1843_GAIN_SIZE   (AD1843_GAIN_PCM_1+1)
+
+int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id);
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id);
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval);
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843);
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc);
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+		      unsigned int id,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels);
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843,
+			 unsigned int id);
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels);
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843);
+int ad1843_init(struct snd_ad1843 *ad1843);
+
+#endif /* __SOUND_AD1843_H */

include/sound/control.h

@@ -129,9 +129,6 @@ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn);
 #define snd_ctl_unregister_ioctl_compat(fcn)
 #endif
 
-int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control);
-int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, struct snd_ctl_elem_value *control);
-
 static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
 {
 	return id->numid - kctl->id.numid;

include/sound/core.h

@@ -412,13 +412,13 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 
 #endif /* CONFIG_SND_DEBUG */
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 /**
  * snd_printdd - debug printk
  * @format: format string
  *
  * Works like snd_printk() for debugging purposes.
- * Ignored when CONFIG_SND_DEBUG_DETECT is not set.
+ * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
  */
 #define snd_printdd(format, args...) snd_printk(format, ##args)
 #else
@@ -442,15 +442,15 @@ struct snd_pci_quirk {
 	unsigned short subvendor;	/* PCI subvendor ID */
 	unsigned short subdevice;	/* PCI subdevice ID */
 	int value;			/* value */
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	const char *name;		/* name of the device (optional) */
 #endif
 };
 
 #define _SND_PCI_QUIRK_ID(vend,dev) \
 	.subvendor = (vend), .subdevice = (dev)
 #define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 #define SND_PCI_QUIRK(vend,dev,xname,val) \
 	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)}
 #else

include/sound/cs4231-regs.h

@@ -177,4 +177,12 @@
 #define CS4236_RIGHT_WAVE	0x1c	/* right wavetable serial port volume */
 #define CS4236_VERSION		0x9c	/* chip version and ID */
 
+/* definitions for extended registers - OPTI93X */
+#define OPTi931_AUX_LEFT_INPUT	0x10
+#define OPTi931_AUX_RIGHT_INPUT	0x11
+#define OPTi93X_MIC_LEFT_INPUT	0x14
+#define OPTi93X_MIC_RIGHT_INPUT	0x15
+#define OPTi93X_OUT_LEFT	0x16
+#define OPTi93X_OUT_RIGHT	0x17
+
 #endif /* __SOUND_CS4231_REGS_H */

include/sound/cs4231.h

@@ -58,6 +58,7 @@
 /* compatible, but clones */
 #define CS4231_HW_INTERWAVE     0x1000	/* InterWave chip */
 #define CS4231_HW_OPL3SA2       0x1101	/* OPL3-SA2 chip, similar to cs4231 */
+#define CS4231_HW_OPTI93X 	0x1102	/* Opti 930/931/933 */
 
 /* defines for codec.hwshare */
 #define CS4231_HWSHARE_IRQ	(1<<0)
@@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg);
 void snd_cs4231_mce_up(struct snd_cs4231 *chip);
 void snd_cs4231_mce_down(struct snd_cs4231 *chip);
 
+void snd_cs4231_overrange(struct snd_cs4231 *chip);
+
 irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id);
 
 const char *snd_cs4231_chip_id(struct snd_cs4231 *chip);

include/sound/emu10k1.h

@@ -1670,6 +1670,7 @@ struct snd_emu_chip_details {
 	unsigned char spi_dac;      /* SPI interface for DAC */
 	unsigned char i2c_adc;      /* I2C interface for ADC */
 	unsigned char adc_1361t;    /* Use Philips 1361T ADC */
+	unsigned char invert_shared_spdif; /* analog/digital switch inverted */
 	const char *driver;
 	const char *name;
 	const char *id;		/* for backward compatibility - can be NULL if not needed */

include/sound/seq_kernel.h

@@ -105,7 +105,7 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp,
 			      int cap, int type, int midi_channels, int midi_voices, char *portname);
 int snd_seq_event_port_detach(int client, int port);
 
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 void snd_seq_autoload_lock(void);
 void snd_seq_autoload_unlock(void);
 #else

include/sound/soc-dapm.h

@@ -130,6 +130,13 @@
 {	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
 	.shift = wshift, .invert = winvert}
 
+/* generic register modifier widget */
+#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
+{	.id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \
+	.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
+	.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
+	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
+
 /* dapm kcontrol types */
 #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -193,6 +200,7 @@ struct snd_soc_dapm_widget;
 enum snd_soc_dapm_type;
 struct snd_soc_dapm_path;
 struct snd_soc_dapm_pin;
+struct snd_soc_dapm_route;
 
 /* dapm controls */
 int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@@ -205,25 +213,32 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
 	const struct snd_soc_dapm_widget *widget);
+int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+	const struct snd_soc_dapm_widget *widget,
+	int num);
 
 /* dapm path setup */
-int snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
+int  __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
 	const char *sink_name, const char *control_name, const char *src_name);
 int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
 void snd_soc_dapm_free(struct snd_soc_device *socdev);
+int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+			    const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
 	int event);
-int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event);
+int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+	enum snd_soc_bias_level level);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
 
-/* dapm audio endpoint control */
-int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
-	char *pin, int status);
-int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
+/* dapm audio pin control and status */
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_sync(struct snd_soc_codec *codec);
 
 /* dapm widget types */
 enum snd_soc_dapm_type {
@@ -245,6 +260,18 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_post,			/* machine specific post widget - exec last */
 };
 
+/*
+ * DAPM audio route definition.
+ *
+ * Defines an audio route originating at source via control and finishing
+ * at sink.
+ */
+struct snd_soc_dapm_route {
+	const char *sink;
+	const char *control;
+	const char *source;
+};
+
 /* dapm audio path between two widgets */
 struct snd_soc_dapm_path {
 	char *name;
@@ -277,6 +304,9 @@ struct snd_soc_dapm_widget {
 	unsigned char shift;			/* bits to shift */
 	unsigned int saved_value;		/* widget saved value */
 	unsigned int value;				/* widget current value */
+	unsigned int mask;			/* non-shifted mask */
+	unsigned int on_val;			/* on state value */
+	unsigned int off_val;			/* off state value */
 	unsigned char power:1;			/* block power status */
 	unsigned char invert:1;			/* invert the power bit */
 	unsigned char active:1;			/* active stream on DAC, ADC's */