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
  ...

12 files changed, 209 insertions, 93 deletions

diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca2417..892b7f168eb 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/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)
diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h
new file mode 100644
index 00000000000..b236a9d1d6e
--- /dev/null
+++ b/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 */
diff --git a/include/sound/control.h b/include/sound/control.h
index 3dc1291f52d..4721b4bba05 100644
--- a/include/sound/control.h
+++ b/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;
diff --git a/include/sound/core.h b/include/sound/core.h
index 695ee53488a..558b96284bd 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -412,13 +412,13 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 
 #endif /* CONFIG_SND_DEBUG */
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 /**
  * snd_printdd - debug printk
  * @format: format string
  *
  * Works like snd_printk() for debugging purposes.
- * Ignored when CONFIG_SND_DEBUG_DETECT is not set.
+ * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
  */
 #define snd_printdd(format, args...) snd_printk(format, ##args)
 #else
@@ -442,7 +442,7 @@ struct snd_pci_quirk {
 	unsigned short subvendor;	/* PCI subvendor ID */
 	unsigned short subdevice;	/* PCI subdevice ID */
 	int value;			/* value */
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	const char *name;		/* name of the device (optional) */
 #endif
 };
@@ -450,7 +450,7 @@ struct snd_pci_quirk {
 #define _SND_PCI_QUIRK_ID(vend,dev) \
 	.subvendor = (vend), .subdevice = (dev)
 #define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 #define SND_PCI_QUIRK(vend,dev,xname,val) \
 	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)}
 #else
diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h
index e8d1f3e31f9..92647532c45 100644
--- a/include/sound/cs4231-regs.h
+++ b/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 */
diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h
index 66055d702aa..f0785f9f4ae 100644
--- a/include/sound/cs4231.h
+++ b/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);
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 7b7b9b13b4d..10ee28eac01 100644
--- a/include/sound/emu10k1.h
+++ b/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 */
diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h
index f023c1b97f8..3d9afb6a8c9 100644
--- a/include/sound/seq_kernel.h
+++ b/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
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index a105b01e06d..3030fdc6981 100644
--- a/include/sound/soc-dapm.h
+++ b/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 */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index d3c8c033dff..1890d87c520 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -73,6 +73,15 @@
 	.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
 	.private_value = (reg_left) | ((shift) << 8)  | \
 		((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \
+{	.iface  = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		  SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p  = (tlv_array), \
+	.info   = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \
+	.put    = snd_soc_put_volsw_s8, \
+	.private_value = (reg) | (((signed char)max) << 16) | \
+			 (((signed char)min) << 24) }
 #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \
 {	.reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
 	.mask = xmask, .texts = xtexts }
@@ -91,6 +100,15 @@
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) }
+#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\
+	 xhandler_get, xhandler_put, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) }
 #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_bool_ext, \
@@ -103,6 +121,24 @@
 	.private_value = (unsigned long)&xenum }
 
 /*
+ * Bias levels
+ *
+ * @ON:      Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ *           stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ *           in progress. NOTE: The transition time between STANDBY and ON
+ *           should be as fast as possible and no longer than 10ms.
+ * @OFF:     Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_PREPARE,
+	SND_SOC_BIAS_STANDBY,
+	SND_SOC_BIAS_OFF,
+};
+
+/*
  * Digital Audio Interface (DAI) types
  */
 #define SND_SOC_DAI_AC97	0x1
@@ -185,8 +221,7 @@ struct snd_soc_pcm_stream;
 struct snd_soc_ops;
 struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
-struct snd_soc_codec_dai;
-struct snd_soc_cpu_dai;
+struct snd_soc_dai;
 struct snd_soc_codec;
 struct snd_soc_machine_config;
 struct soc_enum;
@@ -221,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	struct snd_ac97_bus_ops *ops, int num);
 void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
 
+/* Digital Audio Interface clocking API.*/
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+	unsigned int freq, int dir);
+
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+	int div_id, int div);
+
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+	int pll_id, unsigned int freq_in, unsigned int freq_out);
+
+/* Digital Audio interface formatting */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
+
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+	unsigned int mask, int slots);
+
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
+
+/* Digital Audio Interface mute */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+
 /*
  *Controls
  */
@@ -249,6 +305,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo);
+int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
 
 /* SoC PCM stream information */
 struct snd_soc_pcm_stream {
@@ -272,87 +334,45 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-/* ASoC codec DAI ops */
-struct snd_soc_codec_ops {
-	/* codec DAI clocking configuration */
-	int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai,
+/* ASoC DAI ops */
+struct snd_soc_dai_ops {
+	/* DAI clocking configuration */
+	int (*set_sysclk)(struct snd_soc_dai *dai,
 		int clk_id, unsigned int freq, int dir);
-	int (*set_pll)(struct snd_soc_codec_dai *codec_dai,
+	int (*set_pll)(struct snd_soc_dai *dai,
 		int pll_id, unsigned int freq_in, unsigned int freq_out);
-	int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai,
-		int div_id, int div);
+	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
 
-	/* CPU DAI format configuration */
-	int (*set_fmt)(struct snd_soc_codec_dai *codec_dai,
-		unsigned int fmt);
-	int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai,
+	/* DAI format configuration */
+	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+	int (*set_tdm_slot)(struct snd_soc_dai *dai,
 		unsigned int mask, int slots);
-	int (*set_tristate)(struct snd_soc_codec_dai *, int tristate);
+	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 
 	/* digital mute */
-	int (*digital_mute)(struct snd_soc_codec_dai *, int mute);
-};
-
-/* ASoC cpu DAI ops */
-struct snd_soc_cpu_ops {
-	/* CPU DAI clocking configuration */
-	int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai,
-		int clk_id, unsigned int freq, int dir);
-	int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai,
-		int div_id, int div);
-	int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out);
-
-	/* CPU DAI format configuration */
-	int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai,
-		unsigned int fmt);
-	int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai,
-		unsigned int mask, int slots);
-	int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate);
-};
-
-/* SoC Codec DAI */
-struct snd_soc_codec_dai {
-	char *name;
-	int id;
-	unsigned char type;
-
-	/* DAI capabilities */
-	struct snd_soc_pcm_stream playback;
-	struct snd_soc_pcm_stream capture;
-
-	/* DAI runtime info */
-	struct snd_soc_codec *codec;
-	unsigned int active;
-	unsigned char pop_wait:1;
-
-	/* ops */
-	struct snd_soc_ops ops;
-	struct snd_soc_codec_ops dai_ops;
-
-	/* DAI private data */
-	void *private_data;
+	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
 };
 
-/* SoC CPU DAI */
-struct snd_soc_cpu_dai {
-
+/* SoC  DAI (Digital Audio Interface) */
+struct snd_soc_dai {
 	/* DAI description */
 	char *name;
 	unsigned int id;
 	unsigned char type;
 
 	/* DAI callbacks */
-	int (*probe)(struct platform_device *pdev);
-	void (*remove)(struct platform_device *pdev);
+	int (*probe)(struct platform_device *pdev,
+		     struct snd_soc_dai *dai);
+	void (*remove)(struct platform_device *pdev,
+		       struct snd_soc_dai *dai);
 	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_cpu_dai *cpu_dai);
+		struct snd_soc_dai *dai);
 	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_cpu_dai *cpu_dai);
+		struct snd_soc_dai *dai);
 
 	/* ops */
 	struct snd_soc_ops ops;
-	struct snd_soc_cpu_ops dai_ops;
+	struct snd_soc_dai_ops dai_ops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
@@ -360,7 +380,9 @@ struct snd_soc_cpu_dai {
 
 	/* DAI runtime info */
 	struct snd_pcm_runtime *runtime;
-	unsigned char active:1;
+	struct snd_soc_codec *codec;
+	unsigned int active;
+	unsigned char pop_wait:1;
 	void *dma_data;
 
 	/* DAI private data */
@@ -374,7 +396,8 @@ struct snd_soc_codec {
 	struct mutex mutex;
 
 	/* callbacks */
-	int (*dapm_event)(struct snd_soc_codec *codec, int event);
+	int (*set_bias_level)(struct snd_soc_codec *,
+			      enum snd_soc_bias_level level);
 
 	/* runtime */
 	struct snd_card *card;
@@ -396,12 +419,12 @@ struct snd_soc_codec {
 	/* dapm */
 	struct list_head dapm_widgets;
 	struct list_head dapm_paths;
-	unsigned int dapm_state;
-	unsigned int suspend_dapm_state;
+	enum snd_soc_bias_level bias_level;
+	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
 
 	/* codec DAI's */
-	struct snd_soc_codec_dai *dai;
+	struct snd_soc_dai *dai;
 	unsigned int num_dai;
 };
 
@@ -420,12 +443,12 @@ struct snd_soc_platform {
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
 	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_cpu_dai *cpu_dai);
+		struct snd_soc_dai *dai);
 	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_cpu_dai *cpu_dai);
+		struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *,
+	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
 		struct snd_pcm *);
 	void (*pcm_free)(struct snd_pcm *);
 
@@ -439,8 +462,8 @@ struct snd_soc_dai_link  {
 	char *stream_name;		/* Stream name */
 
 	/* DAI */
-	struct snd_soc_codec_dai *codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai;
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *cpu_dai;
 
 	/* machine stream operations */
 	struct snd_soc_ops *ops;
@@ -467,7 +490,8 @@ struct snd_soc_machine {
 	int (*resume_post)(struct platform_device *pdev);
 
 	/* callbacks */
-	int (*dapm_event)(struct snd_soc_machine *, int event);
+	int (*set_bias_level)(struct snd_soc_machine *,
+			      enum snd_soc_bias_level level);
 
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;
@@ -482,6 +506,7 @@ struct snd_soc_device {
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
 	struct delayed_work delayed_work;
+	struct work_struct deferred_resume_work;
 	void *codec_data;
 };
 
diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h
index 2e564bfb37f..110d5dc3a2b 100644
--- a/include/sound/uda1341.h
+++ b/include/sound/uda1341.h
@@ -15,8 +15,6 @@
  *                           features support
  */
 
-/* $Id: uda1341.h,v 1.8 2005/11/17 14:17:21 tiwai Exp $ */
-
 #define UDA1341_ALSA_NAME "snd-uda1341"
 
 /*
diff --git a/include/sound/version.h b/include/sound/version.h
index ed6fb2eb1ea..6b78aff273a 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
-/* include/version.h.  Generated by alsa/ksync script.  */
-#define CONFIG_SND_VERSION "1.0.16"
+/* include/version.h */
+#define CONFIG_SND_VERSION "1.0.17"
 #define CONFIG_SND_DATE ""