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
@@ -845,7 +848,7 @@ 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
@@ -853,6 +856,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  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
@@ -442,7 +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

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 */

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, \
@@ -102,6 +120,24 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.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
  */
@@ -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;
 };
 

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"
 
 /*

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 ""

sound/Kconfig

@@ -1,11 +1,9 @@
 # sound/Config.in
 #
 
-menu "Sound"
-	depends on HAS_IOMEM
-
-config SOUND
+menuconfig SOUND
 	tristate "Sound card support"
+	depends on HAS_IOMEM
 	help
 	  If you have a sound card in your computer, i.e. if it can say more
 	  than an occasional beep, say Y.  Be sure to have all the information
@@ -28,22 +26,22 @@ config SOUND
 	  and read <file:Documentation/sound/oss/README.modules>; the module
 	  will be called soundcore.
 
+if SOUND
+
 source "sound/oss/dmasound/Kconfig"
 
 if !M68K
 
-menu "Advanced Linux Sound Architecture"
-	depends on SOUND!=n
-
-config SND
+menuconfig SND
 	tristate "Advanced Linux Sound Architecture"
-	depends on SOUND
 	help
 	  Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture),
 	  the new base sound system.
 
 	  For more information, see <http://www.alsa-project.org/>
 
+if SND
+
 source "sound/core/Kconfig"
 
 source "sound/drivers/Kconfig"
@@ -58,9 +56,7 @@ source "sound/aoa/Kconfig"
 
 source "sound/arm/Kconfig"
 
-if SPI
 source "sound/spi/Kconfig"
-endif
 
 source "sound/mips/Kconfig"
 
@@ -80,22 +76,20 @@ source "sound/parisc/Kconfig"
 
 source "sound/soc/Kconfig"
 
-endmenu
+endif # SND
 
-menu "Open Sound System"
-	depends on SOUND!=n
-
-config SOUND_PRIME
+menuconfig SOUND_PRIME
 	tristate "Open Sound System (DEPRECATED)"
-	depends on SOUND
 	help
 	  Say 'Y' or 'M' to enable Open Sound System drivers.
 
+if SOUND_PRIME
+
 source "sound/oss/Kconfig"
 
-endmenu
+endif # SOUND_PRIME
 
-endif
+endif # !M68K
 
 config AC97_BUS
 	tristate
@@ -105,4 +99,4 @@ config AC97_BUS
 	  sound although they're sharing the AC97 bus. Concerned drivers
 	  should "select" this.
 
-endmenu
+endif # SOUND

sound/aoa/Kconfig

@@ -1,18 +1,17 @@
-menu "Apple Onboard Audio driver"
-	depends on SND!=n && PPC_PMAC
-
-config SND_AOA
+menuconfig SND_AOA
 	tristate "Apple Onboard Audio driver"
-	depends on SND
+	depends on PPC_PMAC
 	select SND_PCM
 	---help---
 	This option enables the new driver for the various
 	Apple Onboard Audio components.
 
+if SND_AOA
+
 source "sound/aoa/fabrics/Kconfig"
 
 source "sound/aoa/codecs/Kconfig"
 
 source "sound/aoa/soundbus/Kconfig"
 
-endmenu
+endif	# SND_AOA

sound/aoa/codecs/Kconfig

@@ -1,6 +1,5 @@
 config SND_AOA_ONYX
 	tristate "support Onyx chip"
-	depends on SND_AOA
 	select I2C
 	select I2C_POWERMAC
 	---help---
@@ -10,7 +9,6 @@ config SND_AOA_ONYX
 
 #config SND_AOA_TOPAZ
 #	tristate "support Topaz chips"
-#	depends on SND_AOA
 #	---help---
 #	This option enables support for the Topaz (CS84xx)
 #	codec chips found in the latest Apple machines,
@@ -19,7 +17,6 @@ config SND_AOA_ONYX
 
 config SND_AOA_TAS
 	tristate "support TAS chips"
-	depends on SND_AOA
 	select I2C
 	select I2C_POWERMAC
 	---help---
@@ -29,7 +26,6 @@ config SND_AOA_TAS
 
 config SND_AOA_TOONIE
 	tristate "support Toonie chip"
-	depends on SND_AOA
 	---help---
 	This option enables support for the toonie codec
 	found in the Mac Mini. If you have a Mac Mini and

sound/aoa/fabrics/Kconfig

@@ -1,6 +1,5 @@
 config SND_AOA_FABRIC_LAYOUT
 	tristate "layout-id fabric"
-	depends on SND_AOA
 	select SND_AOA_SOUNDBUS
 	select SND_AOA_SOUNDBUS_I2S
 	---help---

sound/aoa/soundbus/Kconfig

@@ -1,6 +1,5 @@
 config SND_AOA_SOUNDBUS
 	tristate "Apple Soundbus support"
-	depends on SOUND
 	select SND_PCM
 	---help---
 	This option enables the generic driver for the soundbus

sound/arm/Kconfig

@@ -1,11 +1,19 @@
 # ALSA ARM drivers
 
-menu "ALSA ARM devices"
-	depends on SND!=n && ARM
+menuconfig SND_ARM
+	bool "ARM sound devices"
+	depends on ARM
+	default y
+	help
+	  Support for sound devices specific to ARM architectures.
+	  Drivers that are implemented on ASoC can be found in
+	  "ALSA for SoC audio support" section.
+
+if SND_ARM
 
 config SND_SA11XX_UDA1341
 	tristate "SA11xx UDA1341TS driver (iPaq H3600)"
-	depends on ARCH_SA1100 && SND && L3
+	depends on ARCH_SA1100 && L3
 	select SND_PCM
 	help
 	  Say Y here if you have a Compaq iPaq H3x00 handheld computer
@@ -16,7 +24,7 @@ config SND_SA11XX_UDA1341
 
 config SND_ARMAACI
 	tristate "ARM PrimeCell PL041 AC Link support"
-	depends on SND && ARM_AMBA
+	depends on ARM_AMBA
 	select SND_PCM
 	select SND_AC97_CODEC
 
@@ -26,11 +34,12 @@ config SND_PXA2XX_PCM
 
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
-	depends on ARCH_PXA && SND
+	depends on ARCH_PXA
 	select SND_PXA2XX_PCM
 	select SND_AC97_CODEC
 	help
 	  Say Y or M if you want to support any AC97 codec attached to
 	  the PXA2xx AC97 interface.
 
-endmenu
+endif	# SND_ARM
+

sound/arm/sa11xx-uda1341.c

@@ -21,8 +21,6 @@
  *                              merged HAL layer (patches from Brian)
  */
 
-/* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */
-
 /***************************************************************************************************
 *
 * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai

sound/core/Kconfig

@@ -1,24 +1,19 @@
 # ALSA soundcard-configuration
 config SND_TIMER
 	tristate
-	depends on SND
 
 config SND_PCM
 	tristate
 	select SND_TIMER
-	depends on SND
 
 config SND_HWDEP
 	tristate
-	depends on SND
 
 config SND_RAWMIDI
 	tristate
-	depends on SND
 
 config SND_SEQUENCER
 	tristate "Sequencer support"
-	depends on SND
 	select SND_TIMER
 	help
 	  Say Y or M to enable MIDI sequencer and router support.  This
@@ -44,11 +39,9 @@ config SND_SEQ_DUMMY
 
 config SND_OSSEMUL
 	bool
-	depends on SND
 
 config SND_MIXER_OSS
 	tristate "OSS Mixer API"
-	depends on SND
 	select SND_OSSEMUL
 	help
 	  To enable OSS mixer API emulation (/dev/mixer*), say Y here
@@ -61,7 +54,6 @@ config SND_MIXER_OSS
 
 config SND_PCM_OSS
 	tristate "OSS PCM (digital audio) API"
-	depends on SND
 	select SND_OSSEMUL
 	select SND_PCM
 	help
@@ -84,7 +76,7 @@ config SND_PCM_OSS_PLUGINS
 
 config SND_SEQUENCER_OSS
 	bool "OSS Sequencer API"
-	depends on SND && SND_SEQUENCER
+	depends on SND_SEQUENCER
 	select SND_OSSEMUL
 	help
 	  Say Y here to enable OSS sequencer emulation (both
@@ -98,7 +90,7 @@ config SND_SEQUENCER_OSS
 
 config SND_RTCTIMER
 	tristate "RTC Timer support"
-	depends on SND && RTC
+	depends on RTC
 	select SND_TIMER
 	help
 	  Say Y here to enable RTC timer support for ALSA.  ALSA uses
@@ -123,7 +115,6 @@ config SND_SEQ_RTCTIMER_DEFAULT
 
 config SND_DYNAMIC_MINORS
 	bool "Dynamic device file minor numbers"
-	depends on SND
 	help
 	  If you say Y here, the minor numbers of ALSA device files in
 	  /dev/snd/ are allocated dynamically.  This allows you to have
@@ -134,7 +125,6 @@ config SND_DYNAMIC_MINORS
 
 config SND_SUPPORT_OLD_API
 	bool "Support old ALSA API"
-	depends on SND
 	default y
 	help
 	  Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3
@@ -142,7 +132,7 @@ config SND_SUPPORT_OLD_API
 
 config SND_VERBOSE_PROCFS
 	bool "Verbose procfs contents"
-	depends on SND && PROC_FS
+	depends on PROC_FS
 	default y
 	help
 	  Say Y here to include code for verbose procfs contents (provides
@@ -151,7 +141,6 @@ config SND_VERBOSE_PROCFS
 
 config SND_VERBOSE_PRINTK
 	bool "Verbose printk"
-	depends on SND
 	help
 	  Say Y here to enable verbose log messages.  These messages
 	  will help to identify source file and position containing
@@ -161,16 +150,17 @@ config SND_VERBOSE_PRINTK
 
 config SND_DEBUG
 	bool "Debug"
-	depends on SND
 	help
 	  Say Y here to enable ALSA debug code.
 
-config SND_DEBUG_DETECT
-	bool "Debug detection"
+config SND_DEBUG_VERBOSE
+	bool "More verbose debug"
 	depends on SND_DEBUG
 	help
-	  Say Y here to enable extra-verbose log messages printed when
-	  detecting devices.
+	  Say Y here to enable extra-verbose debugging messages.
+	  
+	  Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages.
+	  So, say Y only if you are ready to be annoyed.
 
 config SND_PCM_XRUN_DEBUG
 	bool "Enable PCM ring buffer overrun/underrun debugging"
@@ -184,4 +174,3 @@ config SND_PCM_XRUN_DEBUG
 
 config SND_VMASTER
 	bool
-	depends on SND

sound/core/control.c

@@ -684,7 +684,8 @@ static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl,
 	return result;
 }
 
-int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control)
+static int snd_ctl_elem_read(struct snd_card *card,
+			     struct snd_ctl_elem_value *control)
 {
 	struct snd_kcontrol *kctl;
 	struct snd_kcontrol_volatile *vd;
@@ -734,8 +735,8 @@ static int snd_ctl_elem_read_user(struct snd_card *card,
 	return result;
 }
 
-int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
-		       struct snd_ctl_elem_value *control)
+static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
+			      struct snd_ctl_elem_value *control)
 {
 	struct snd_kcontrol *kctl;
 	struct snd_kcontrol_volatile *vd;

sound/core/init.c

@@ -46,17 +46,24 @@ static char *slots[SNDRV_CARDS];
 module_param_array(slots, charp, NULL, 0444);
 MODULE_PARM_DESC(slots, "Module names assigned to the slots.");
 
-/* return non-zero if the given index is already reserved for another
+/* return non-zero if the given index is reserved for the given
  * module via slots option
  */
-static int module_slot_mismatch(struct module *module, int idx)
+static int module_slot_match(struct module *module, int idx)
 {
+	int match = 1;
 #ifdef MODULE
-	char *s1, *s2;
+	const char *s1, *s2;
+
 	if (!module || !module->name || !slots[idx])
 		return 0;
-	s1 = slots[idx];
-	s2 = module->name;
+
+	s1 = module->name;
+	s2 = slots[idx];
+	if (*s2 == '!') {
+		match = 0; /* negative match */
+		s2++;
+	}
 	/* compare module name strings
 	 * hyphens are handled as equivalent with underscore
 	 */
@@ -68,12 +75,12 @@ static int module_slot_mismatch(struct module *module, int idx)
 		if (c2 == '-')
 			c2 = '_';
 		if (c1 != c2)
-			return 1;
+			return !match;
 		if (!c1)
 			break;
 	}
-#endif
-	return 0;
+#endif /* MODULE */
+	return match;
 }
 
 #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
@@ -129,7 +136,7 @@ struct snd_card *snd_card_new(int idx, const char *xid,
 			 struct module *module, int extra_size)
 {
 	struct snd_card *card;
-	int err;
+	int err, idx2;
 
 	if (extra_size < 0)
 		extra_size = 0;
@@ -144,35 +151,41 @@ struct snd_card *snd_card_new(int idx, const char *xid,
 	err = 0;
 	mutex_lock(&snd_card_mutex);
 	if (idx < 0) {
-		int idx2;
 		for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++)
 			/* idx == -1 == 0xffff means: take any free slot */
 			if (~snd_cards_lock & idx & 1<<idx2) {
-				if (module_slot_mismatch(module, idx2))
-					continue;
-				idx = idx2;
-				if (idx >= snd_ecards_limit)
-					snd_ecards_limit = idx + 1;
-				break;
+				if (module_slot_match(module, idx2)) {
+					idx = idx2;
+					break;
+				}
+			}
+	}
+	if (idx < 0) {
+		for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++)
+			/* idx == -1 == 0xffff means: take any free slot */
+			if (~snd_cards_lock & idx & 1<<idx2) {
+				if (!slots[idx2] || !*slots[idx2]) {
+					idx = idx2;
+					break;
+				}
 			}
-	} else {
-		 if (idx < snd_ecards_limit) {
-			if (snd_cards_lock & (1 << idx))
-				err = -EBUSY;	/* invalid */
-		} else {
-			if (idx < SNDRV_CARDS)
-				snd_ecards_limit = idx + 1; /* increase the limit */
-			else
-				err = -ENODEV;
-		}
 	}
-	if (idx < 0 || err < 0) {
+	if (idx < 0)
+		err = -ENODEV;
+	else if (idx < snd_ecards_limit) {
+		if (snd_cards_lock & (1 << idx))
+			err = -EBUSY;	/* invalid */
+	} else if (idx >= SNDRV_CARDS)
+		err = -ENODEV;
+	if (err < 0) {
 		mutex_unlock(&snd_card_mutex);
 		snd_printk(KERN_ERR "cannot find the slot for index %d (range 0-%i), error: %d\n",
 			 idx, snd_ecards_limit - 1, err);
 		goto __error;
 	}
 	snd_cards_lock |= 1 << idx;		/* lock it */
+	if (idx >= snd_ecards_limit)
+		snd_ecards_limit = idx + 1; /* increase the limit */
 	mutex_unlock(&snd_card_mutex);
 	card->number = idx;
 	card->module = module;

sound/core/memalloc.c

@@ -79,68 +79,6 @@ struct snd_mem_list {
 #define snd_assert(expr, args...) /**/
 #endif
 
-/*
- *  Hacks
- */
-
-#if defined(__i386__)
-/*
- * A hack to allocate large buffers via dma_alloc_coherent()
- *
- * since dma_alloc_coherent always tries GFP_DMA when the requested
- * pci memory region is below 32bit, it happens quite often that even
- * 2 order of pages cannot be allocated.
- *
- * so in the following, we allocate at first without dma_mask, so that
- * allocation will be done without GFP_DMA.  if the area doesn't match
- * with the requested region, then realloate with the original dma_mask
- * again.
- *
- * Really, we want to move this type of thing into dma_alloc_coherent()
- * so dma_mask doesn't have to be messed with.
- */
-
-static void *snd_dma_hack_alloc_coherent(struct device *dev, size_t size,
-					 dma_addr_t *dma_handle,
-					 gfp_t flags)
-{
-	void *ret;
-	u64 dma_mask, coherent_dma_mask;
-
-	if (dev == NULL || !dev->dma_mask)
-		return dma_alloc_coherent(dev, size, dma_handle, flags);
-	dma_mask = *dev->dma_mask;
-	coherent_dma_mask = dev->coherent_dma_mask;
-	*dev->dma_mask = 0xffffffff; 	/* do without masking */
-	dev->coherent_dma_mask = 0xffffffff; 	/* do without masking */
-	ret = dma_alloc_coherent(dev, size, dma_handle, flags);
-	*dev->dma_mask = dma_mask;	/* restore */
-	dev->coherent_dma_mask = coherent_dma_mask;	/* restore */
-	if (ret) {
-		/* obtained address is out of range? */
-		if (((unsigned long)*dma_handle + size - 1) & ~dma_mask) {
-			/* reallocate with the proper mask */
-			dma_free_coherent(dev, size, ret, *dma_handle);
-			ret = dma_alloc_coherent(dev, size, dma_handle, flags);
-		}
-	} else {
-		/* wish to success now with the proper mask... */
-		if (dma_mask != 0xffffffffUL) {
-			/* allocation with GFP_ATOMIC to avoid the long stall */
-			flags &= ~GFP_KERNEL;
-			flags |= GFP_ATOMIC;
-			ret = dma_alloc_coherent(dev, size, dma_handle, flags);
-		}
-	}
-	return ret;
-}
-
-/* redefine dma_alloc_coherent for some architectures */
-#undef dma_alloc_coherent
-#define dma_alloc_coherent snd_dma_hack_alloc_coherent
-
-#endif /* arch */
-
 /*
  *
  *  Generic memory allocators

sound/core/seq/seq_clientmgr.c

@@ -148,7 +148,7 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
 		return NULL;
 	}
 	spin_unlock_irqrestore(&clients_lock, flags);
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 	if (!in_interrupt()) {
 		static char client_requested[SNDRV_SEQ_GLOBAL_CLIENTS];
 		static char card_requested[SNDRV_CARDS];

sound/core/seq/seq_device.c

@@ -124,7 +124,7 @@ static void snd_seq_device_info(struct snd_info_entry *entry,
  * load all registered drivers (called from seq_clientmgr.c)
  */
 
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 /* avoid auto-loading during module_init() */
 static int snd_seq_in_init;
 void snd_seq_autoload_lock(void)
@@ -140,7 +140,7 @@ void snd_seq_autoload_unlock(void)
 
 void snd_seq_device_load_drivers(void)
 {
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 	struct ops_list *ops;
 
 	/* Calling request_module during module_init()
@@ -566,7 +566,5 @@ EXPORT_SYMBOL(snd_seq_device_load_drivers);
 EXPORT_SYMBOL(snd_seq_device_new);
 EXPORT_SYMBOL(snd_seq_device_register_driver);
 EXPORT_SYMBOL(snd_seq_device_unregister_driver);
-#ifdef CONFIG_KMOD
 EXPORT_SYMBOL(snd_seq_autoload_lock);
 EXPORT_SYMBOL(snd_seq_autoload_unlock);
-#endif

sound/core/sound.c

@@ -60,14 +60,14 @@ EXPORT_SYMBOL(snd_ecards_limit);
 static struct snd_minor *snd_minors[SNDRV_OS_MINORS];
 static DEFINE_MUTEX(sound_mutex);
 
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 
 /**
  * snd_request_card - try to load the card module
  * @card: the card number
  *
  * Tries to load the module "snd-card-X" for the given card number
- * via KMOD.  Returns immediately if already loaded.
+ * via request_module.  Returns immediately if already loaded.
  */
 void snd_request_card(int card)
 {
@@ -92,7 +92,7 @@ static void snd_request_other(int minor)
 	request_module(str);
 }
 
-#endif				/* request_module support */
+#endif	/* modular kernel */
 
 /**
  * snd_lookup_minor_data - get user data of a registered device
@@ -132,7 +132,7 @@ static int snd_open(struct inode *inode, struct file *file)
 		return -ENODEV;
 	mptr = snd_minors[minor];
 	if (mptr == NULL) {
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 		int dev = SNDRV_MINOR_DEVICE(minor);
 		if (dev == SNDRV_MINOR_CONTROL) {
 			/* /dev/aloadC? */

sound/core/timer.c

@@ -146,7 +146,7 @@ static struct snd_timer *snd_timer_find(struct snd_timer_id *tid)
 	return NULL;
 }
 
-#ifdef CONFIG_KMOD
+#ifdef CONFIG_MODULES
 
 static void snd_timer_request(struct snd_timer_id *tid)
 {
@@ -259,8 +259,8 @@ int snd_timer_open(struct snd_timer_instance **ti,
 	/* open a master instance */
 	mutex_lock(&register_mutex);
 	timer = snd_timer_find(tid);
-#ifdef CONFIG_KMOD
-	if (timer == NULL) {
+#ifdef CONFIG_MODULES
+	if (!timer) {
 		mutex_unlock(&register_mutex);
 		snd_timer_request(tid);
 		mutex_lock(&register_mutex);

sound/drivers/Kconfig

@@ -1,15 +1,41 @@
-# ALSA generic drivers
+config SND_MPU401_UART
+        tristate
+        select SND_RAWMIDI
 
-menu "Generic devices"
-	depends on SND!=n
+config SND_OPL3_LIB
+	tristate
+	select SND_TIMER
+	select SND_HWDEP
 
+config SND_OPL4_LIB
+	tristate
+	select SND_TIMER
+	select SND_HWDEP
+
+config SND_VX_LIB
+	tristate
+	select SND_HWDEP
+	select SND_PCM
+
+config SND_AC97_CODEC
+	tristate
+	select SND_PCM
+	select AC97_BUS
+	select SND_VMASTER
+
+menuconfig SND_DRIVERS
+	bool "Generic sound devices"
+	default y
+	help
+	  Support for generic sound devices.
+  
+if SND_DRIVERS
 
 config SND_PCSP
 	tristate "PC-Speaker support (READ HELP!)"
 	depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS
 	depends on INPUT
 	depends on EXPERIMENTAL
-	depends on SND
 	select SND_PCM
 	help
 	  If you don't have a sound card in your computer, you can include a
@@ -35,33 +61,8 @@ config SND_PCSP
 	  Say M if you don't.
 	  Say Y only if you really know what you do.
 
-config SND_MPU401_UART
-        tristate
-        select SND_RAWMIDI
-
-config SND_OPL3_LIB
-	tristate
-	select SND_TIMER
-	select SND_HWDEP
-
-config SND_OPL4_LIB
-	tristate
-	select SND_TIMER
-	select SND_HWDEP
-
-config SND_VX_LIB
-	tristate
-	select SND_HWDEP
-	select SND_PCM
-
-config SND_AC97_CODEC
-	tristate
-	select SND_PCM
-	select AC97_BUS
-
 config SND_DUMMY
 	tristate "Dummy (/dev/null) soundcard"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include the dummy driver.  This driver does
@@ -90,7 +91,6 @@ config SND_VIRMIDI
 
 config SND_MTPAV
 	tristate "MOTU MidiTimePiece AV multiport MIDI"
-	depends on SND
 	select SND_RAWMIDI
 	help
 	  To use a MOTU MidiTimePiece AV multiport MIDI adapter
@@ -102,7 +102,7 @@ config SND_MTPAV
 
 config SND_MTS64
 	tristate "ESI Miditerminal 4140 driver"
-	depends on SND && PARPORT
+	depends on PARPORT
 	select SND_RAWMIDI
 	help
 	  The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with 
@@ -115,7 +115,6 @@ config SND_MTS64
 
 config SND_SERIAL_U16550
 	tristate "UART16550 serial MIDI driver"
-	depends on SND
 	select SND_RAWMIDI
 	help
 	  To include support for MIDI serial port interfaces, say Y here
@@ -131,7 +130,6 @@ config SND_SERIAL_U16550
 
 config SND_MPU401
 	tristate "Generic MPU-401 UART driver"
-	depends on SND
 	select SND_MPU401_UART
 	help
 	  Say Y here to include support for MIDI ports compatible with
@@ -142,7 +140,7 @@ config SND_MPU401
 
 config SND_PORTMAN2X4
 	tristate "Portman 2x4 driver"
-	depends on SND && PARPORT
+	depends on PARPORT
 	select SND_RAWMIDI
 	help
 	  Say Y here to include support for Midiman Portman 2x4 parallel
@@ -153,7 +151,7 @@ config SND_PORTMAN2X4
 
 config SND_ML403_AC97CR
 	tristate "Xilinx ML403 AC97 Controller Reference"
-	depends on SND && XILINX_VIRTEX
+	depends on XILINX_VIRTEX
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the
@@ -163,4 +161,25 @@ config SND_ML403_AC97CR
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-ml403_ac97cr.
 
-endmenu
+config SND_AC97_POWER_SAVE
+	bool "AC97 Power-Saving Mode"
+	depends on SND_AC97_CODEC && EXPERIMENTAL
+	default n
+	help
+	  Say Y here to enable the aggressive power-saving support of
+	  AC97 codecs.  In this mode, the power-mode is dynamically
+	  controlled at each open/close.
+
+	  The mode is activated by passing power_save=1 option to
+	  snd-ac97-codec driver.  You can toggle it dynamically over
+	  sysfs, too.
+
+config SND_AC97_POWER_SAVE_DEFAULT
+	int "Default time-out for AC97 power-save mode"
+	depends on SND_AC97_POWER_SAVE
+	default 0
+	help
+	  The default time-out value in seconds for AC97 automatic
+	  power-save mode.  0 means to disable the power-save mode.
+
+endif	# SND_DRIVERS

sound/drivers/vx/vx_hwdep.c

@@ -183,7 +183,7 @@ static int vx_hwdep_dsp_load(struct snd_hwdep *hw,
 		kfree(fw);
 		return -ENOMEM;
 	}
-	if (copy_from_user(fw->data, dsp->image, dsp->length)) {
+	if (copy_from_user((void *)fw->data, dsp->image, dsp->length)) {
 		free_fw(fw);
 		return -EFAULT;
 	}

sound/i2c/cs8427.c

@@ -23,6 +23,7 @@
 #include <linux/slab.h>
 #include <linux/delay.h>
 #include <linux/init.h>
+#include <asm/unaligned.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/pcm.h>
@@ -264,10 +265,7 @@ int snd_cs8427_create(struct snd_i2c_bus *bus,
 		goto __fail;
 	}
 	/* write default channel status bytes */
-	buf[0] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 0));
-	buf[1] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 8));
-	buf[2] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 16));
-	buf[3] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 24));
+	put_unaligned_le32(SNDRV_PCM_DEFAULT_CON_SPDIF, buf);
 	memset(buf + 4, 0, 24 - 4);
 	if (snd_cs8427_send_corudata(device, 0, buf, 24) < 0)
 		goto __fail;

sound/i2c/l3/uda1341.c

@@ -17,8 +17,6 @@
  * 2002-05-12   Tomas Kasparek  another code cleanup
  */
 
-/* $Id: uda1341.c,v 1.18 2005/11/17 14:17:21 tiwai Exp $ */
-
 #include <linux/module.h>
 #include <linux/init.h>
 #include <linux/types.h>

sound/isa/Kconfig

@@ -21,12 +21,17 @@ config SND_SB16_DSP
         select SND_PCM
         select SND_SB_COMMON
 
-menu "ISA devices"
-	depends on SND!=n && ISA && ISA_DMA_API
+menuconfig SND_ISA
+	bool "ISA sound devices"
+	depends on ISA && ISA_DMA_API
+	default y
+	help
+	  Support for sound devices connected via the ISA bus.
+
+if SND_ISA
 
 config SND_ADLIB
 	tristate "AdLib FM card"
-	depends on SND
 	select SND_OPL3_LIB
 	help
 	  Say Y here to include support for AdLib FM cards.
@@ -36,7 +41,7 @@ config SND_ADLIB
 
 config SND_AD1816A
 	tristate "Analog Devices SoundPort AD1816A"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -50,7 +55,6 @@ config SND_AD1816A
 
 config SND_AD1848
 	tristate "Generic AD1848/CS4248 driver"
-	depends on SND
 	select SND_AD1848_LIB
 	help
 	  Say Y here to include support for AD1848 (Analog Devices) or
@@ -64,7 +68,7 @@ config SND_AD1848
 
 config SND_ALS100
 	tristate "Avance Logic ALS100/ALS120"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -78,7 +82,7 @@ config SND_ALS100
 
 config SND_AZT2320
 	tristate "Aztech Systems AZT2320"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -92,7 +96,6 @@ config SND_AZT2320
 
 config SND_CMI8330
 	tristate "C-Media CMI8330"
-	depends on SND
 	select SND_AD1848_LIB
 	select SND_SB16_DSP
 	help
@@ -104,7 +107,6 @@ config SND_CMI8330
 
 config SND_CS4231
 	tristate "Generic Cirrus Logic CS4231 driver"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
 	help
@@ -116,7 +118,6 @@ config SND_CS4231
 
 config SND_CS4232
 	tristate "Generic Cirrus Logic CS4232 driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -129,7 +130,6 @@ config SND_CS4232
 
 config SND_CS4236
 	tristate "Generic Cirrus Logic CS4236+ driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -142,7 +142,7 @@ config SND_CS4236
 
 config SND_DT019X
 	tristate "Diamond Technologies DT-019X, Avance Logic ALS-007"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -156,7 +156,7 @@ config SND_DT019X
 
 config SND_ES968
 	tristate "Generic ESS ES968 driver"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select ISAPNP
 	select SND_MPU401_UART
 	select SND_SB8_DSP
@@ -168,7 +168,6 @@ config SND_ES968
 
 config SND_ES1688
 	tristate "Generic ESS ES688/ES1688 driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -181,7 +180,6 @@ config SND_ES1688
 
 config SND_ES18XX
 	tristate "Generic ESS ES18xx driver"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -193,7 +191,7 @@ config SND_ES18XX
 
 config SND_SC6000
 	tristate "Gallant SC-6000, Audio Excel DSP 16"
-	depends on SND && HAS_IOPORT
+	depends on HAS_IOPORT
 	select SND_AD1848_LIB
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -204,15 +202,10 @@ config SND_SC6000
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-sc6000.
 
-config SND_GUS_SYNTH
-	tristate
-
 config SND_GUSCLASSIC
 	tristate "Gravis UltraSound Classic"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_PCM
-	select SND_GUS_SYNTH
 	help
 	  Say Y here to include support for Gravis UltraSound Classic
 	  soundcards.
@@ -222,11 +215,9 @@ config SND_GUSCLASSIC
 
 config SND_GUSEXTREME
 	tristate "Gravis UltraSound Extreme"
-	depends on SND
 	select SND_HWDEP
 	select SND_MPU401_UART
 	select SND_PCM
-	select SND_GUS_SYNTH
 	help
 	  Say Y here to include support for Gravis UltraSound Extreme
 	  soundcards.
@@ -236,10 +227,8 @@ config SND_GUSEXTREME
 
 config SND_GUSMAX
 	tristate "Gravis UltraSound MAX"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_CS4231_LIB
-	select SND_GUS_SYNTH
 	help
 	  Say Y here to include support for Gravis UltraSound MAX
 	  soundcards.
@@ -249,10 +238,9 @@ config SND_GUSMAX
 
 config SND_INTERWAVE
 	tristate "AMD InterWave, Gravis UltraSound PnP"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select SND_RAWMIDI
 	select SND_CS4231_LIB
-	select SND_GUS_SYNTH
 	help
 	  Say Y here to include support for AMD InterWave based
 	  soundcards (Gravis UltraSound Plug & Play, STB SoundRage32,
@@ -263,10 +251,9 @@ config SND_INTERWAVE
 
 config SND_INTERWAVE_STB
 	tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)"
-	depends on SND && PNP && ISA
+	depends on PNP
 	select SND_RAWMIDI
 	select SND_CS4231_LIB
-	select SND_GUS_SYNTH
 	help
 	  Say Y here to include support for AMD InterWave based
 	  soundcards with a TEA6330T bass and treble regulator
@@ -277,7 +264,6 @@ config SND_INTERWAVE_STB
 
 config SND_OPL3SA2
 	tristate "Yamaha OPL3-SA2/SA3"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -290,7 +276,6 @@ config SND_OPL3SA2
 
 config SND_OPTI92X_AD1848
 	tristate "OPTi 82C92x - AD1848"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_OPL4_LIB
 	select SND_MPU401_UART
@@ -304,7 +289,6 @@ config SND_OPTI92X_AD1848
 
 config SND_OPTI92X_CS4231
 	tristate "OPTi 82C92x - CS4231"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_OPL4_LIB
 	select SND_MPU401_UART
@@ -318,10 +302,9 @@ config SND_OPTI92X_CS4231
 
 config SND_OPTI93X
 	tristate "OPTi 82C93x"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
-	select SND_PCM
+	select SND_CS4231_LIB
 	help
 	  Say Y here to include support for soundcards based on Opti
 	  82C93x chips.
@@ -331,7 +314,6 @@ config SND_OPTI93X
 
 config SND_MIRO
 	tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver"
-	depends on SND
 	select SND_OPL4_LIB
 	select SND_CS4231_LIB
 	select SND_MPU401_UART
@@ -345,7 +327,6 @@ config SND_MIRO
 
 config SND_SB8
 	tristate "Sound Blaster 1.0/2.0/Pro (8-bit)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_RAWMIDI
 	select SND_SB8_DSP
@@ -358,7 +339,6 @@ config SND_SB8
 
 config SND_SB16
 	tristate "Sound Blaster 16 (PnP)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_SB16_DSP
@@ -371,7 +351,6 @@ config SND_SB16
 
 config SND_SBAWE
 	tristate "Sound Blaster AWE (32,64) (PnP)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_SB16_DSP
@@ -402,7 +381,6 @@ config SND_SB16_CSP_FIRMWARE_IN_KERNEL
 
 config SND_SGALAXY
 	tristate "Aztech Sound Galaxy"
-	depends on SND
 	select SND_AD1848_LIB
 	help
 	  Say Y here to include support for Aztech Sound Galaxy
@@ -413,7 +391,6 @@ config SND_SGALAXY
 
 config SND_SSCAPE
 	tristate "Ensoniq SoundScape PnP driver"
-	depends on SND
 	select SND_HWDEP
 	select SND_MPU401_UART
 	select SND_CS4231_LIB
@@ -426,7 +403,6 @@ config SND_SSCAPE
 
 config SND_WAVEFRONT
 	tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)"
-	depends on SND
 	select FW_LOADER
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -448,4 +424,5 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL
 	  you need to install the firmware files from the
 	  alsa-firmware package.
 
-endmenu
+endif	# SND_ISA
+

sound/isa/cs423x/cs4231_lib.c

@@ -119,6 +119,42 @@ static unsigned char snd_cs4231_original_image[32] =
 	0x00,			/* 1f/31 - cbrl */
 };
 
+static unsigned char snd_opti93x_original_image[32] =
+{
+	0x00,		/* 00/00 - l_mixout_outctrl */
+	0x00,		/* 01/01 - r_mixout_outctrl */
+	0x88,		/* 02/02 - l_cd_inctrl */
+	0x88,		/* 03/03 - r_cd_inctrl */
+	0x88,		/* 04/04 - l_a1/fm_inctrl */
+	0x88,		/* 05/05 - r_a1/fm_inctrl */
+	0x80,		/* 06/06 - l_dac_inctrl */
+	0x80,		/* 07/07 - r_dac_inctrl */
+	0x00,		/* 08/08 - ply_dataform_reg */
+	0x00,		/* 09/09 - if_conf */
+	0x00,		/* 0a/10 - pin_ctrl */
+	0x00,		/* 0b/11 - err_init_reg */
+	0x0a,		/* 0c/12 - id_reg */
+	0x00,		/* 0d/13 - reserved */
+	0x00,		/* 0e/14 - ply_upcount_reg */
+	0x00,		/* 0f/15 - ply_lowcount_reg */
+	0x88,		/* 10/16 - reserved/l_a1_inctrl */
+	0x88,		/* 11/17 - reserved/r_a1_inctrl */
+	0x88,		/* 12/18 - l_line_inctrl */
+	0x88,		/* 13/19 - r_line_inctrl */
+	0x88,		/* 14/20 - l_mic_inctrl */
+	0x88,		/* 15/21 - r_mic_inctrl */
+	0x80,		/* 16/22 - l_out_outctrl */
+	0x80,		/* 17/23 - r_out_outctrl */
+	0x00,		/* 18/24 - reserved */
+	0x00,		/* 19/25 - reserved */
+	0x00,		/* 1a/26 - reserved */
+	0x00,		/* 1b/27 - reserved */
+	0x00,		/* 1c/28 - cap_dataform_reg */
+	0x00,		/* 1d/29 - reserved */
+	0x00,		/* 1e/30 - cap_upcount_reg */
+	0x00		/* 1f/31 - cap_lowcount_reg */
+};
+
 /*
  *  Basic I/O functions
  */
@@ -895,7 +931,7 @@ static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void snd_cs4231_overrange(struct snd_cs4231 *chip)
+void snd_cs4231_overrange(struct snd_cs4231 *chip)
 {
 	unsigned long flags;
 	unsigned char res;
@@ -1054,8 +1090,11 @@ static int snd_cs4231_probe(struct snd_cs4231 *chip)
 	chip->image[CS4231_IFACE_CTRL] =
 	    (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) |
 	    (chip->single_dma ? CS4231_SINGLE_DMA : 0);
-	chip->image[CS4231_ALT_FEATURE_1] = 0x80;
-	chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01;
+	if (chip->hardware != CS4231_HW_OPTI93X) {
+		chip->image[CS4231_ALT_FEATURE_1] = 0x80;
+		chip->image[CS4231_ALT_FEATURE_2] =
+			chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01;
+	}
 	ptr = (unsigned char *) &chip->image;
 	snd_cs4231_mce_down(chip);
 	spin_lock_irqsave(&chip->reg_lock, flags);
@@ -1376,6 +1415,7 @@ const char *snd_cs4231_chip_id(struct snd_cs4231 *chip)
 	case CS4231_HW_INTERWAVE: return "AMD InterWave";
 	case CS4231_HW_OPL3SA2: return chip->card->shortname;
 	case CS4231_HW_AD1845: return "AD1845";
+	case CS4231_HW_OPTI93X: return "OPTi 93x";
 	default: return "???";
 	}
 }
@@ -1401,8 +1441,13 @@ static int snd_cs4231_new(struct snd_card *card,
 	chip->rate_constraint = snd_cs4231_xrate;
 	chip->set_playback_format = snd_cs4231_playback_format;
 	chip->set_capture_format = snd_cs4231_capture_format;
-        memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image));
-        
+	if (chip->hardware == CS4231_HW_OPTI93X)
+		memcpy(&chip->image, &snd_opti93x_original_image,
+		       sizeof(snd_opti93x_original_image));
+	else
+		memcpy(&chip->image, &snd_cs4231_original_image,
+		       sizeof(snd_cs4231_original_image));
+
         *rchip = chip;
         return 0;
 }
@@ -1790,6 +1835,48 @@ CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0),
 CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1)
 };
                                         
+static struct snd_kcontrol_new snd_opti93x_controls[] = {
+CS4231_DOUBLE("Master Playback Switch", 0,
+	      OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1),
+CS4231_DOUBLE("Master Playback Volume", 0,
+	      OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1),
+CS4231_DOUBLE("PCM Playback Switch", 0,
+	      CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("PCM Playback Volume", 0,
+	      CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1),
+CS4231_DOUBLE("FM Playback Switch", 0,
+	      CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("FM Playback Volume", 0,
+	      CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1),
+CS4231_DOUBLE("Line Playback Switch", 0,
+	      CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+CS4231_DOUBLE("Line Playback Volume", 0,
+	      CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1),
+CS4231_DOUBLE("Mic Playback Switch", 0,
+	      OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("Mic Playback Volume", 0,
+	      OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1),
+CS4231_DOUBLE("Mic Boost", 0,
+	      CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0),
+CS4231_DOUBLE("CD Playback Switch", 0,
+	      CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("CD Playback Volume", 0,
+	      CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1),
+CS4231_DOUBLE("Aux Playback Switch", 0,
+	      OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("Aux Playback Volume", 0,
+	      OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1),
+CS4231_DOUBLE("Capture Volume", 0,
+	      CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0),
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Capture Source",
+	.info = snd_cs4231_info_mux,
+	.get = snd_cs4231_get_mux,
+	.put = snd_cs4231_put_mux,
+}
+};
+
 int snd_cs4231_mixer(struct snd_cs4231 *chip)
 {
 	struct snd_card *card;
@@ -1802,10 +1889,22 @@ int snd_cs4231_mixer(struct snd_cs4231 *chip)
 
 	strcpy(card->mixername, chip->pcm->name);
 
-	for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) {
-		if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0)
-			return err;
-	}
+	if (chip->hardware == CS4231_HW_OPTI93X)
+		for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) {
+			err = snd_ctl_add(card,
+					snd_ctl_new1(&snd_opti93x_controls[idx],
+						     chip));
+			if (err < 0)
+				return err;
+		}
+	else
+		for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) {
+			err = snd_ctl_add(card,
+					snd_ctl_new1(&snd_cs4231_controls[idx],
+						     chip));
+			if (err < 0)
+				return err;
+		}
 	return 0;
 }
 
@@ -1815,6 +1914,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out);
 EXPORT_SYMBOL(snd_cs4236_ext_in);
 EXPORT_SYMBOL(snd_cs4231_mce_up);
 EXPORT_SYMBOL(snd_cs4231_mce_down);
+EXPORT_SYMBOL(snd_cs4231_overrange);
 EXPORT_SYMBOL(snd_cs4231_interrupt);
 EXPORT_SYMBOL(snd_cs4231_chip_id);
 EXPORT_SYMBOL(snd_cs4231_create);

sound/isa/sb/Makefile

@@ -34,5 +34,3 @@ ifeq ($(CONFIG_SND_SB16_CSP),y)
   obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o
 endif
 obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o
-
-obj-m := $(sort $(obj-m))

sound/isa/wavefront/wavefront_synth.c

@@ -1939,7 +1939,7 @@ static int __devinit
 wavefront_download_firmware (snd_wavefront_t *dev, char *path)
 
 {
-	unsigned char *buf;
+	const unsigned char *buf;
 	int len, err;
 	int section_cnt_downloaded = 0;
 	const struct firmware *firmware;

sound/mips/Kconfig

@@ -1,15 +1,34 @@
 # ALSA MIPS drivers
 
-menu "ALSA MIPS devices"
-	depends on SND!=n && MIPS
+menuconfig SND_MIPS
+	bool "MIPS sound devices"
+	depends on MIPS
+	default y
+	help
+	  Support for sound devices of MIPS architectures.
+
+if SND_MIPS
+
+config SND_SGI_O2
+	tristate "SGI O2 Audio"
+	depends on SGI_IP32
+        help
+                Sound support for the SGI O2 Workstation. 
+
+config SND_SGI_HAL2
+        tristate "SGI HAL2 Audio"
+        depends on SGI_HAS_HAL2
+        help
+                Sound support for the SGI Indy and Indigo2 Workstation.
+
 
 config SND_AU1X00
 	tristate "Au1x00 AC97 Port Driver"
-	depends on (SOC_AU1000 || SOC_AU1100 || SOC_AU1500) && SND
+	depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
 	  ALSA Sound driver for the Au1x00's AC97 port.
 
-endmenu
+endif	# SND_MIPS
 

sound/mips/Makefile

@@ -3,6 +3,10 @@
 #
 
 snd-au1x00-objs := au1x00.o
+snd-sgi-o2-objs := sgio2audio.o ad1843.o
+snd-sgi-hal2-objs := hal2.o
 
 # Toplevel Module Dependency
 obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o
+obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o
+obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o

sound/mips/ad1843.c

@@ -0,0 +1,561 @@
+/*
+ *   AD1843 low level driver
+ *
+ *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ *
+ *   inspired from vwsnd.c (SGI VW audio driver)
+ *     Copyright 1999 Silicon Graphics, Inc.  All rights reserved.
+ *
+ *   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/sched.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ad1843.h>
+
+/*
+ * AD1843 bitfield definitions.  All are named as in the AD1843 data
+ * sheet, with ad1843_ prepended and individual bit numbers removed.
+ *
+ * E.g., bits LSS0 through LSS2 become ad1843_LSS.
+ *
+ * Only the bitfields we need are defined.
+ */
+
+struct ad1843_bitfield {
+	char reg;
+	char lo_bit;
+	char nbits;
+};
+
+static const struct ad1843_bitfield
+	ad1843_PDNO   = {  0, 14,  1 },	/* Converter Power-Down Flag */
+	ad1843_INIT   = {  0, 15,  1 },	/* Clock Initialization Flag */
+	ad1843_RIG    = {  2,  0,  4 },	/* Right ADC Input Gain */
+	ad1843_RMGE   = {  2,  4,  1 },	/* Right ADC Mic Gain Enable */
+	ad1843_RSS    = {  2,  5,  3 },	/* Right ADC Source Select */
+	ad1843_LIG    = {  2,  8,  4 },	/* Left ADC Input Gain */
+	ad1843_LMGE   = {  2, 12,  1 },	/* Left ADC Mic Gain Enable */
+	ad1843_LSS    = {  2, 13,  3 },	/* Left ADC Source Select */
+	ad1843_RD2M   = {  3,  0,  5 },	/* Right DAC 2 Mix Gain/Atten */
+	ad1843_RD2MM  = {  3,  7,  1 },	/* Right DAC 2 Mix Mute */
+	ad1843_LD2M   = {  3,  8,  5 },	/* Left DAC 2 Mix Gain/Atten */
+	ad1843_LD2MM  = {  3, 15,  1 },	/* Left DAC 2 Mix Mute */
+	ad1843_RX1M   = {  4,  0,  5 },	/* Right Aux 1 Mix Gain/Atten */
+	ad1843_RX1MM  = {  4,  7,  1 },	/* Right Aux 1 Mix Mute */
+	ad1843_LX1M   = {  4,  8,  5 },	/* Left Aux 1 Mix Gain/Atten */
+	ad1843_LX1MM  = {  4, 15,  1 },	/* Left Aux 1 Mix Mute */
+	ad1843_RX2M   = {  5,  0,  5 },	/* Right Aux 2 Mix Gain/Atten */
+	ad1843_RX2MM  = {  5,  7,  1 },	/* Right Aux 2 Mix Mute */
+	ad1843_LX2M   = {  5,  8,  5 },	/* Left Aux 2 Mix Gain/Atten */
+	ad1843_LX2MM  = {  5, 15,  1 },	/* Left Aux 2 Mix Mute */
+	ad1843_RMCM   = {  7,  0,  5 },	/* Right Mic Mix Gain/Atten */
+	ad1843_RMCMM  = {  7,  7,  1 },	/* Right Mic Mix Mute */
+	ad1843_LMCM   = {  7,  8,  5 },	/* Left Mic Mix Gain/Atten */
+	ad1843_LMCMM  = {  7, 15,  1 },	/* Left Mic Mix Mute */
+	ad1843_HPOS   = {  8,  4,  1 },	/* Headphone Output Voltage Swing */
+	ad1843_HPOM   = {  8,  5,  1 },	/* Headphone Output Mute */
+	ad1843_MPOM   = {  8,  6,  1 },	/* Mono Output Mute */
+	ad1843_RDA1G  = {  9,  0,  6 },	/* Right DAC1 Analog/Digital Gain */
+	ad1843_RDA1GM = {  9,  7,  1 },	/* Right DAC1 Analog Mute */
+	ad1843_LDA1G  = {  9,  8,  6 },	/* Left DAC1 Analog/Digital Gain */
+	ad1843_LDA1GM = {  9, 15,  1 },	/* Left DAC1 Analog Mute */
+	ad1843_RDA2G  = { 10,  0,  6 },	/* Right DAC2 Analog/Digital Gain */
+	ad1843_RDA2GM = { 10,  7,  1 },	/* Right DAC2 Analog Mute */
+	ad1843_LDA2G  = { 10,  8,  6 },	/* Left DAC2 Analog/Digital Gain */
+	ad1843_LDA2GM = { 10, 15,  1 },	/* Left DAC2 Analog Mute */
+	ad1843_RDA1AM = { 11,  7,  1 },	/* Right DAC1 Digital Mute */
+	ad1843_LDA1AM = { 11, 15,  1 },	/* Left DAC1 Digital Mute */
+	ad1843_RDA2AM = { 12,  7,  1 },	/* Right DAC2 Digital Mute */
+	ad1843_LDA2AM = { 12, 15,  1 },	/* Left DAC2 Digital Mute */
+	ad1843_ADLC   = { 15,  0,  2 },	/* ADC Left Sample Rate Source */
+	ad1843_ADRC   = { 15,  2,  2 },	/* ADC Right Sample Rate Source */
+	ad1843_DA1C   = { 15,  8,  2 },	/* DAC1 Sample Rate Source */
+	ad1843_DA2C   = { 15, 10,  2 },	/* DAC2 Sample Rate Source */
+	ad1843_C1C    = { 17,  0, 16 },	/* Clock 1 Sample Rate Select */
+	ad1843_C2C    = { 20,  0, 16 },	/* Clock 2 Sample Rate Select */
+	ad1843_C3C    = { 23,  0, 16 },	/* Clock 3 Sample Rate Select */
+	ad1843_DAADL  = { 25,  4,  2 },	/* Digital ADC Left Source Select */
+	ad1843_DAADR  = { 25,  6,  2 },	/* Digital ADC Right Source Select */
+	ad1843_DAMIX  = { 25, 14,  1 },	/* DAC Digital Mix Enable */
+	ad1843_DRSFLT = { 25, 15,  1 },	/* Digital Reampler Filter Mode */
+	ad1843_ADLF   = { 26,  0,  2 }, /* ADC Left Channel Data Format */
+	ad1843_ADRF   = { 26,  2,  2 }, /* ADC Right Channel Data Format */
+	ad1843_ADTLK  = { 26,  4,  1 },	/* ADC Transmit Lock Mode Select */
+	ad1843_SCF    = { 26,  7,  1 },	/* SCLK Frequency Select */
+	ad1843_DA1F   = { 26,  8,  2 },	/* DAC1 Data Format Select */
+	ad1843_DA2F   = { 26, 10,  2 },	/* DAC2 Data Format Select */
+	ad1843_DA1SM  = { 26, 14,  1 },	/* DAC1 Stereo/Mono Mode Select */
+	ad1843_DA2SM  = { 26, 15,  1 },	/* DAC2 Stereo/Mono Mode Select */
+	ad1843_ADLEN  = { 27,  0,  1 },	/* ADC Left Channel Enable */
+	ad1843_ADREN  = { 27,  1,  1 },	/* ADC Right Channel Enable */
+	ad1843_AAMEN  = { 27,  4,  1 },	/* Analog to Analog Mix Enable */
+	ad1843_ANAEN  = { 27,  7,  1 },	/* Analog Channel Enable */
+	ad1843_DA1EN  = { 27,  8,  1 },	/* DAC1 Enable */
+	ad1843_DA2EN  = { 27,  9,  1 },	/* DAC2 Enable */
+	ad1843_DDMEN  = { 27, 12,  1 },	/* DAC2 to DAC1 Mix  Enable */
+	ad1843_C1EN   = { 28, 11,  1 },	/* Clock Generator 1 Enable */
+	ad1843_C2EN   = { 28, 12,  1 },	/* Clock Generator 2 Enable */
+	ad1843_C3EN   = { 28, 13,  1 },	/* Clock Generator 3 Enable */
+	ad1843_PDNI   = { 28, 15,  1 };	/* Converter Power Down */
+
+/*
+ * The various registers of the AD1843 use three different formats for
+ * specifying gain.  The ad1843_gain structure parameterizes the
+ * formats.
+ */
+
+struct ad1843_gain {
+	int	negative;		/* nonzero if gain is negative. */
+	const struct ad1843_bitfield *lfield;
+	const struct ad1843_bitfield *rfield;
+	const struct ad1843_bitfield *lmute;
+	const struct ad1843_bitfield *rmute;
+};
+
+static const struct ad1843_gain ad1843_gain_RECLEV = {
+	.negative = 0,
+	.lfield   = &ad1843_LIG,
+	.rfield   = &ad1843_RIG
+};
+static const struct ad1843_gain ad1843_gain_LINE = {
+	.negative = 1,
+	.lfield   = &ad1843_LX1M,
+	.rfield   = &ad1843_RX1M,
+	.lmute    = &ad1843_LX1MM,
+	.rmute    = &ad1843_RX1MM
+};
+static const struct ad1843_gain ad1843_gain_LINE_2 = {
+	.negative = 1,
+	.lfield   = &ad1843_LDA2G,
+	.rfield   = &ad1843_RDA2G,
+	.lmute    = &ad1843_LDA2GM,
+	.rmute    = &ad1843_RDA2GM
+};
+static const struct ad1843_gain ad1843_gain_MIC = {
+	.negative = 1,
+	.lfield   = &ad1843_LMCM,
+	.rfield   = &ad1843_RMCM,
+	.lmute    = &ad1843_LMCMM,
+	.rmute    = &ad1843_RMCMM
+};
+static const struct ad1843_gain ad1843_gain_PCM_0 = {
+	.negative = 1,
+	.lfield   = &ad1843_LDA1G,
+	.rfield   = &ad1843_RDA1G,
+	.lmute    = &ad1843_LDA1GM,
+	.rmute    = &ad1843_RDA1GM
+};
+static const struct ad1843_gain ad1843_gain_PCM_1 = {
+	.negative = 1,
+	.lfield   = &ad1843_LD2M,
+	.rfield   = &ad1843_RD2M,
+	.lmute    = &ad1843_LD2MM,
+	.rmute    = &ad1843_RD2MM
+};
+
+static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] =
+{
+	&ad1843_gain_RECLEV,
+	&ad1843_gain_LINE,
+	&ad1843_gain_LINE_2,
+	&ad1843_gain_MIC,
+	&ad1843_gain_PCM_0,
+	&ad1843_gain_PCM_1,
+};
+
+/* read the current value of an AD1843 bitfield. */
+
+static int ad1843_read_bits(struct snd_ad1843 *ad1843,
+			    const struct ad1843_bitfield *field)
+{
+	int w;
+
+	w = ad1843->read(ad1843->chip, field->reg);
+	return w >> field->lo_bit & ((1 << field->nbits) - 1);
+}
+
+/*
+ * write a new value to an AD1843 bitfield and return the old value.
+ */
+
+static int ad1843_write_bits(struct snd_ad1843 *ad1843,
+			     const struct ad1843_bitfield *field,
+			     int newval)
+{
+	int w, mask, oldval, newbits;
+
+	w = ad1843->read(ad1843->chip, field->reg);
+	mask = ((1 << field->nbits) - 1) << field->lo_bit;
+	oldval = (w & mask) >> field->lo_bit;
+	newbits = (newval << field->lo_bit) & mask;
+	w = (w & ~mask) | newbits;
+	ad1843->write(ad1843->chip, field->reg, w);
+
+	return oldval;
+}
+
+/*
+ * ad1843_read_multi reads multiple bitfields from the same AD1843
+ * register.  It uses a single read cycle to do it.  (Reading the
+ * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20
+ * microseconds.)
+ *
+ * Called like this.
+ *
+ *  ad1843_read_multi(ad1843, nfields,
+ *		      &ad1843_FIELD1, &val1,
+ *		      &ad1843_FIELD2, &val2, ...);
+ */
+
+static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+	va_list ap;
+	const struct ad1843_bitfield *fp;
+	int w = 0, mask, *value, reg = -1;
+
+	va_start(ap, argcount);
+	while (--argcount >= 0) {
+		fp = va_arg(ap, const struct ad1843_bitfield *);
+		value = va_arg(ap, int *);
+		if (reg == -1) {
+			reg = fp->reg;
+			w = ad1843->read(ad1843->chip, reg);
+		}
+
+		mask = (1 << fp->nbits) - 1;
+		*value = w >> fp->lo_bit & mask;
+	}
+	va_end(ap);
+}
+
+/*
+ * ad1843_write_multi stores multiple bitfields into the same AD1843
+ * register.  It uses one read and one write cycle to do it.
+ *
+ * Called like this.
+ *
+ *  ad1843_write_multi(ad1843, nfields,
+ *		       &ad1843_FIELD1, val1,
+ *		       &ad1843_FIELF2, val2, ...);
+ */
+
+static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+	va_list ap;
+	int reg;
+	const struct ad1843_bitfield *fp;
+	int value;
+	int w, m, mask, bits;
+
+	mask = 0;
+	bits = 0;
+	reg = -1;
+
+	va_start(ap, argcount);
+	while (--argcount >= 0) {
+		fp = va_arg(ap, const struct ad1843_bitfield *);
+		value = va_arg(ap, int);
+		if (reg == -1)
+			reg = fp->reg;
+		else
+			BUG_ON(reg != fp->reg);
+		m = ((1 << fp->nbits) - 1) << fp->lo_bit;
+		mask |= m;
+		bits |= (value << fp->lo_bit) & m;
+	}
+	va_end(ap);
+
+	if (~mask & 0xFFFF)
+		w = ad1843->read(ad1843->chip, reg);
+	else
+		w = 0;
+	w = (w & ~mask) | bits;
+	ad1843->write(ad1843->chip, reg, w);
+}
+
+int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id)
+{
+	const struct ad1843_gain *gp = ad1843_gain[id];
+	int ret;
+
+	ret = (1 << gp->lfield->nbits);
+	if (!gp->lmute)
+		ret -= 1;
+	return ret;
+}
+
+/*
+ * ad1843_get_gain reads the specified register and extracts the gain value
+ * using the supplied gain type.
+ */
+
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id)
+{
+	int lg, rg, lm, rm;
+	const struct ad1843_gain *gp = ad1843_gain[id];
+	unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+	ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg);
+	if (gp->negative) {
+		lg = mask - lg;
+		rg = mask - rg;
+	}
+	if (gp->lmute) {
+		ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm);
+		if (lm)
+			lg = 0;
+		if (rm)
+			rg = 0;
+	}
+	return lg << 0 | rg << 8;
+}
+
+/*
+ * Set an audio channel's gain.
+ *
+ * Returns the new gain, which may be lower than the old gain.
+ */
+
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval)
+{
+	const struct ad1843_gain *gp = ad1843_gain[id];
+	unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+	int lg = (newval >> 0) & mask;
+	int rg = (newval >> 8) & mask;
+	int lm = (lg == 0) ? 1 : 0;
+	int rm = (rg == 0) ? 1 : 0;
+
+	if (gp->negative) {
+		lg = mask - lg;
+		rg = mask - rg;
+	}
+	if (gp->lmute)
+		ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm);
+	ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg);
+	return ad1843_get_gain(ad1843, id);
+}
+
+/* Returns the current recording source */
+
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843)
+{
+	int val = ad1843_read_bits(ad1843, &ad1843_LSS);
+
+	if (val < 0 || val > 2) {
+		val = 2;
+		ad1843_write_multi(ad1843, 2,
+				   &ad1843_LSS, val, &ad1843_RSS, val);
+	}
+	return val;
+}
+
+/*
+ * Set recording source.
+ *
+ * Returns newsrc on success, -errno on failure.
+ */
+
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc)
+{
+	if (newsrc < 0 || newsrc > 2)
+		return -EINVAL;
+
+	ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc);
+	return newsrc;
+}
+
+/* Setup ad1843 for D/A conversion. */
+
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+		      unsigned int id,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels)
+{
+	int ad_fmt = 0, ad_mode = 0;
+
+	switch (fmt) {
+	case SNDRV_PCM_FORMAT_S8:
+		ad_fmt = 0;
+		break;
+	case SNDRV_PCM_FORMAT_U8:
+		ad_fmt = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		ad_fmt = 1;
+		break;
+	case SNDRV_PCM_FORMAT_MU_LAW:
+		ad_fmt = 2;
+		break;
+	case SNDRV_PCM_FORMAT_A_LAW:
+		ad_fmt = 3;
+		break;
+	default:
+		break;
+	}
+
+	switch (channels) {
+	case 2:
+		ad_mode = 0;
+		break;
+	case 1:
+		ad_mode = 1;
+		break;
+	default:
+		break;
+	}
+
+	if (id) {
+		ad1843_write_bits(ad1843, &ad1843_C2C, framerate);
+		ad1843_write_multi(ad1843, 2,
+				   &ad1843_DA2SM, ad_mode,
+				   &ad1843_DA2F, ad_fmt);
+	} else {
+		ad1843_write_bits(ad1843, &ad1843_C1C, framerate);
+		ad1843_write_multi(ad1843, 2,
+				   &ad1843_DA1SM, ad_mode,
+				   &ad1843_DA1F, ad_fmt);
+	}
+}
+
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id)
+{
+	if (id)
+		ad1843_write_bits(ad1843, &ad1843_DA2F, 1);
+	else
+		ad1843_write_bits(ad1843, &ad1843_DA1F, 1);
+}
+
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels)
+{
+	int da_fmt = 0;
+
+	switch (fmt) {
+	case SNDRV_PCM_FORMAT_S8:	da_fmt = 0; break;
+	case SNDRV_PCM_FORMAT_U8:	da_fmt = 0; break;
+	case SNDRV_PCM_FORMAT_S16_LE:	da_fmt = 1; break;
+	case SNDRV_PCM_FORMAT_MU_LAW:	da_fmt = 2; break;
+	case SNDRV_PCM_FORMAT_A_LAW:	da_fmt = 3; break;
+	default:		break;
+	}
+
+	ad1843_write_bits(ad1843, &ad1843_C3C, framerate);
+	ad1843_write_multi(ad1843, 2,
+			   &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt);
+}
+
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843)
+{
+	/* nothing to do */
+}
+
+/*
+ * Fully initialize the ad1843.  As described in the AD1843 data
+ * sheet, section "START-UP SEQUENCE".  The numbered comments are
+ * subsection headings from the data sheet.  See the data sheet, pages
+ * 52-54, for more info.
+ *
+ * return 0 on success, -errno on failure.  */
+
+int ad1843_init(struct snd_ad1843 *ad1843)
+{
+	unsigned long later;
+
+	if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) {
+		printk(KERN_ERR "ad1843: AD1843 won't initialize\n");
+		return -EIO;
+	}
+
+	ad1843_write_bits(ad1843, &ad1843_SCF, 1);
+
+	/* 4. Put the conversion resources into standby. */
+	ad1843_write_bits(ad1843, &ad1843_PDNI, 0);
+	later = jiffies + msecs_to_jiffies(500);
+
+	while (ad1843_read_bits(ad1843, &ad1843_PDNO)) {
+		if (time_after(jiffies, later)) {
+			printk(KERN_ERR
+			       "ad1843: AD1843 won't power up\n");
+			return -EIO;
+		}
+		schedule_timeout_interruptible(5);
+	}
+
+	/* 5. Power up the clock generators and enable clock output pins. */
+	ad1843_write_multi(ad1843, 3,
+			   &ad1843_C1EN, 1,
+			   &ad1843_C2EN, 1,
+			   &ad1843_C3EN, 1);
+
+	/* 6. Configure conversion resources while they are in standby. */
+
+	/* DAC1/2 use clock 1/2 as source, ADC uses clock 3.  Always. */
+	ad1843_write_multi(ad1843, 4,
+			   &ad1843_DA1C, 1,
+			   &ad1843_DA2C, 2,
+			   &ad1843_ADLC, 3,
+			   &ad1843_ADRC, 3);
+
+	/* 7. Enable conversion resources. */
+	ad1843_write_bits(ad1843, &ad1843_ADTLK, 1);
+	ad1843_write_multi(ad1843, 7,
+			   &ad1843_ANAEN, 1,
+			   &ad1843_AAMEN, 1,
+			   &ad1843_DA1EN, 1,
+			   &ad1843_DA2EN, 1,
+			   &ad1843_DDMEN, 1,
+			   &ad1843_ADLEN, 1,
+			   &ad1843_ADREN, 1);
+
+	/* 8. Configure conversion resources while they are enabled. */
+
+	/* set gain to 0 for all channels */
+	ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0);
+
+	/* Unmute all channels. */
+	/* DAC1 */
+	ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0);
+	/* DAC2 */
+	ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0);
+
+	/* Set default recording source to Line In and set
+	 * mic gain to +20 dB.
+	 */
+	ad1843_set_recsrc(ad1843, 2);
+	ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1);
+
+	/* Set Speaker Out level to +/- 4V and unmute it. */
+	ad1843_write_multi(ad1843, 3,
+			   &ad1843_HPOS, 1,
+			   &ad1843_HPOM, 0,
+			   &ad1843_MPOM, 0);
+
+	return 0;
+}

sound/mips/hal2.c

@@ -0,0 +1,947 @@
+/*
+ *  Driver for A2 audio system used in SGI machines
+ *  Copyright (c) 2008 Thomas Bogendoerfer <tsbogend@alpha.fanken.de>
+ *
+ *  Based on OSS code from Ladislav Michl <ladis@linux-mips.org>, which
+ *  was based on code from Ulf Carlsson
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ *  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., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <asm/sgi/hpc3.h>
+#include <asm/sgi/ip22.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm-indirect.h>
+#include <sound/initval.h>
+
+#include "hal2.h"
+
+static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SGI HAL2 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SGI HAL2 soundcard.");
+MODULE_DESCRIPTION("ALSA driver for SGI HAL2 audio");
+MODULE_AUTHOR("Thomas Bogendoerfer");
+MODULE_LICENSE("GPL");
+
+
+#define H2_BLOCK_SIZE	1024
+#define H2_BUF_SIZE	16384
+
+struct hal2_pbus {
+	struct hpc3_pbus_dmacregs *pbus;
+	int pbusnr;
+	unsigned int ctrl;		/* Current state of pbus->pbdma_ctrl */
+};
+
+struct hal2_desc {
+	struct hpc_dma_desc desc;
+	u32 pad;			/* padding */
+};
+
+struct hal2_codec {
+	struct snd_pcm_indirect pcm_indirect;
+	struct snd_pcm_substream *substream;
+
+	unsigned char *buffer;
+	dma_addr_t buffer_dma;
+	struct hal2_desc *desc;
+	dma_addr_t desc_dma;
+	int desc_count;
+	struct hal2_pbus pbus;
+	int voices;			/* mono/stereo */
+	unsigned int sample_rate;
+	unsigned int master;		/* Master frequency */
+	unsigned short mod;		/* MOD value */
+	unsigned short inc;		/* INC value */
+};
+
+#define H2_MIX_OUTPUT_ATT	0
+#define H2_MIX_INPUT_GAIN	1
+
+struct snd_hal2 {
+	struct snd_card *card;
+
+	struct hal2_ctl_regs *ctl_regs;	/* HAL2 ctl registers */
+	struct hal2_aes_regs *aes_regs;	/* HAL2 aes registers */
+	struct hal2_vol_regs *vol_regs;	/* HAL2 vol registers */
+	struct hal2_syn_regs *syn_regs;	/* HAL2 syn registers */
+
+	struct hal2_codec dac;
+	struct hal2_codec adc;
+};
+
+#define H2_INDIRECT_WAIT(regs)	while (hal2_read(&regs->isr) & H2_ISR_TSTATUS);
+
+#define H2_READ_ADDR(addr)	(addr | (1<<7))
+#define H2_WRITE_ADDR(addr)	(addr)
+
+static inline u32 hal2_read(u32 *reg)
+{
+	return __raw_readl(reg);
+}
+
+static inline void hal2_write(u32 val, u32 *reg)
+{
+	__raw_writel(val, reg);
+}
+
+
+static u32 hal2_i_read32(struct snd_hal2 *hal2, u16 addr)
+{
+	u32 ret;
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(H2_READ_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	ret = hal2_read(&regs->idr0) & 0xffff;
+	hal2_write(H2_READ_ADDR(addr) | 0x1, &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	ret |= (hal2_read(&regs->idr0) & 0xffff) << 16;
+	return ret;
+}
+
+static void hal2_i_write16(struct snd_hal2 *hal2, u16 addr, u16 val)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(val, &regs->idr0);
+	hal2_write(0, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_write32(struct snd_hal2 *hal2, u16 addr, u32 val)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(val & 0xffff, &regs->idr0);
+	hal2_write(val >> 16, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_setbit16(struct snd_hal2 *hal2, u16 addr, u16 bit)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(H2_READ_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	hal2_write((hal2_read(&regs->idr0) & 0xffff) | bit, &regs->idr0);
+	hal2_write(0, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_clearbit16(struct snd_hal2 *hal2, u16 addr, u16 bit)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(H2_READ_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	hal2_write((hal2_read(&regs->idr0) & 0xffff) & ~bit, &regs->idr0);
+	hal2_write(0, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static int hal2_gain_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	switch ((int)kcontrol->private_value) {
+	case H2_MIX_OUTPUT_ATT:
+		uinfo->value.integer.max = 31;
+		break;
+	case H2_MIX_INPUT_GAIN:
+		uinfo->value.integer.max = 15;
+		break;
+	}
+	return 0;
+}
+
+static int hal2_gain_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol);
+	u32 tmp;
+	int l, r;
+
+	switch ((int)kcontrol->private_value) {
+	case H2_MIX_OUTPUT_ATT:
+		tmp = hal2_i_read32(hal2, H2I_DAC_C2);
+		if (tmp & H2I_C2_MUTE) {
+			l = 0;
+			r = 0;
+		} else {
+			l = 31 - ((tmp >> H2I_C2_L_ATT_SHIFT) & 31);
+			r = 31 - ((tmp >> H2I_C2_R_ATT_SHIFT) & 31);
+		}
+		break;
+	case H2_MIX_INPUT_GAIN:
+		tmp = hal2_i_read32(hal2, H2I_ADC_C2);
+		l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15;
+		r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15;
+		break;
+	}
+	ucontrol->value.integer.value[0] = l;
+	ucontrol->value.integer.value[1] = r;
+
+	return 0;
+}
+
+static int hal2_gain_put(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol);
+	u32 old, new;
+	int l, r;
+
+	l = ucontrol->value.integer.value[0];
+	r = ucontrol->value.integer.value[1];
+
+	switch ((int)kcontrol->private_value) {
+	case H2_MIX_OUTPUT_ATT:
+		old = hal2_i_read32(hal2, H2I_DAC_C2);
+		new = old & ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE);
+		if (l | r) {
+			l = 31 - l;
+			r = 31 - r;
+			new |= (l << H2I_C2_L_ATT_SHIFT);
+			new |= (r << H2I_C2_R_ATT_SHIFT);
+		} else
+			new |= H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE;
+		hal2_i_write32(hal2, H2I_DAC_C2, new);
+		break;
+	case H2_MIX_INPUT_GAIN:
+		old = hal2_i_read32(hal2, H2I_ADC_C2);
+		new = old & ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M);
+		new |= (l << H2I_C2_L_GAIN_SHIFT);
+		new |= (r << H2I_C2_R_GAIN_SHIFT);
+		hal2_i_write32(hal2, H2I_ADC_C2, new);
+		break;
+	}
+	return old != new;
+}
+
+static struct snd_kcontrol_new hal2_ctrl_headphone __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Headphone Playback Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = H2_MIX_OUTPUT_ATT,
+	.info           = hal2_gain_info,
+	.get            = hal2_gain_get,
+	.put            = hal2_gain_put,
+};
+
+static struct snd_kcontrol_new hal2_ctrl_mic __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Mic Capture Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = H2_MIX_INPUT_GAIN,
+	.info           = hal2_gain_info,
+	.get            = hal2_gain_get,
+	.put            = hal2_gain_put,
+};
+
+static int __devinit hal2_mixer_create(struct snd_hal2 *hal2)
+{
+	int err;
+
+	/* mute DAC */
+	hal2_i_write32(hal2, H2I_DAC_C2,
+		       H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE);
+	/* mute ADC */
+	hal2_i_write32(hal2, H2I_ADC_C2, 0);
+
+	err = snd_ctl_add(hal2->card,
+			  snd_ctl_new1(&hal2_ctrl_headphone, hal2));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(hal2->card,
+			  snd_ctl_new1(&hal2_ctrl_mic, hal2));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static irqreturn_t hal2_interrupt(int irq, void *dev_id)
+{
+	struct snd_hal2 *hal2 = dev_id;
+	irqreturn_t ret = IRQ_NONE;
+
+	/* decide what caused this interrupt */
+	if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) {
+		snd_pcm_period_elapsed(hal2->dac.substream);
+		ret = IRQ_HANDLED;
+	}
+	if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) {
+		snd_pcm_period_elapsed(hal2->adc.substream);
+		ret = IRQ_HANDLED;
+	}
+	return ret;
+}
+
+static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate)
+{
+	unsigned short mod;
+
+	if (44100 % rate < 48000 % rate) {
+		mod = 4 * 44100 / rate;
+		codec->master = 44100;
+	} else {
+		mod = 4 * 48000 / rate;
+		codec->master = 48000;
+	}
+
+	codec->inc = 4;
+	codec->mod = mod;
+	rate = 4 * codec->master / mod;
+
+	return rate;
+}
+
+static void hal2_set_dac_rate(struct snd_hal2 *hal2)
+{
+	unsigned int master = hal2->dac.master;
+	int inc = hal2->dac.inc;
+	int mod = hal2->dac.mod;
+
+	hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0);
+	hal2_i_write32(hal2, H2I_BRES1_C2,
+		       ((0xffff & (inc - mod - 1)) << 16) | inc);
+}
+
+static void hal2_set_adc_rate(struct snd_hal2 *hal2)
+{
+	unsigned int master = hal2->adc.master;
+	int inc = hal2->adc.inc;
+	int mod = hal2->adc.mod;
+
+	hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0);
+	hal2_i_write32(hal2, H2I_BRES2_C2,
+		       ((0xffff & (inc - mod - 1)) << 16) | inc);
+}
+
+static void hal2_setup_dac(struct snd_hal2 *hal2)
+{
+	unsigned int fifobeg, fifoend, highwater, sample_size;
+	struct hal2_pbus *pbus = &hal2->dac.pbus;
+
+	/* Now we set up some PBUS information. The PBUS needs information about
+	 * what portion of the fifo it will use. If it's receiving or
+	 * transmitting, and finally whether the stream is little endian or big
+	 * endian. The information is written later, on the start call.
+	 */
+	sample_size = 2 * hal2->dac.voices;
+	/* Fifo should be set to hold exactly four samples. Highwater mark
+	 * should be set to two samples. */
+	highwater = (sample_size * 2) >> 1;	/* halfwords */
+	fifobeg = 0;				/* playback is first */
+	fifoend = (sample_size * 4) >> 3;	/* doublewords */
+	pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD |
+		     (highwater << 8) | (fifobeg << 16) | (fifoend << 24);
+	/* We disable everything before we do anything at all */
+	pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+	hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX);
+	/* Setup the HAL2 for playback */
+	hal2_set_dac_rate(hal2);
+	/* Set endianess */
+	hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX);
+	/* Set DMA bus */
+	hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr));
+	/* We are using 1st Bresenham clock generator for playback */
+	hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT)
+			| (1 << H2I_C1_CLKID_SHIFT)
+			| (hal2->dac.voices << H2I_C1_DATAT_SHIFT));
+}
+
+static void hal2_setup_adc(struct snd_hal2 *hal2)
+{
+	unsigned int fifobeg, fifoend, highwater, sample_size;
+	struct hal2_pbus *pbus = &hal2->adc.pbus;
+
+	sample_size = 2 * hal2->adc.voices;
+	highwater = (sample_size * 2) >> 1;		/* halfwords */
+	fifobeg = (4 * 4) >> 3;				/* record is second */
+	fifoend = (4 * 4 + sample_size * 4) >> 3;	/* doublewords */
+	pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD |
+		     (highwater << 8) | (fifobeg << 16) | (fifoend << 24);
+	pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+	hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR);
+	/* Setup the HAL2 for record */
+	hal2_set_adc_rate(hal2);
+	/* Set endianess */
+	hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR);
+	/* Set DMA bus */
+	hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr));
+	/* We are using 2nd Bresenham clock generator for record */
+	hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT)
+			| (2 << H2I_C1_CLKID_SHIFT)
+			| (hal2->adc.voices << H2I_C1_DATAT_SHIFT));
+}
+
+static void hal2_start_dac(struct snd_hal2 *hal2)
+{
+	struct hal2_pbus *pbus = &hal2->dac.pbus;
+
+	pbus->pbus->pbdma_dptr = hal2->dac.desc_dma;
+	pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT;
+	/* enable DAC */
+	hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX);
+}
+
+static void hal2_start_adc(struct snd_hal2 *hal2)
+{
+	struct hal2_pbus *pbus = &hal2->adc.pbus;
+
+	pbus->pbus->pbdma_dptr = hal2->adc.desc_dma;
+	pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT;
+	/* enable ADC */
+	hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR);
+}
+
+static inline void hal2_stop_dac(struct snd_hal2 *hal2)
+{
+	hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+	/* The HAL2 itself may remain enabled safely */
+}
+
+static inline void hal2_stop_adc(struct snd_hal2 *hal2)
+{
+	hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+}
+
+static int hal2_alloc_dmabuf(struct hal2_codec *codec)
+{
+	struct hal2_desc *desc;
+	dma_addr_t desc_dma, buffer_dma;
+	int count = H2_BUF_SIZE / H2_BLOCK_SIZE;
+	int i;
+
+	codec->buffer = dma_alloc_noncoherent(NULL, H2_BUF_SIZE,
+					      &buffer_dma, GFP_KERNEL);
+	if (!codec->buffer)
+		return -ENOMEM;
+	desc = dma_alloc_noncoherent(NULL, count * sizeof(struct hal2_desc),
+				     &desc_dma, GFP_KERNEL);
+	if (!desc) {
+		dma_free_noncoherent(NULL, H2_BUF_SIZE,
+				     codec->buffer, buffer_dma);
+		return -ENOMEM;
+	}
+	codec->buffer_dma = buffer_dma;
+	codec->desc_dma = desc_dma;
+	codec->desc = desc;
+	for (i = 0; i < count; i++) {
+		desc->desc.pbuf = buffer_dma + i * H2_BLOCK_SIZE;
+		desc->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE;
+		desc->desc.pnext = (i == count - 1) ?
+		      desc_dma : desc_dma + (i + 1) * sizeof(struct hal2_desc);
+		desc++;
+	}
+	dma_cache_sync(NULL, codec->desc, count * sizeof(struct hal2_desc),
+		       DMA_TO_DEVICE);
+	codec->desc_count = count;
+	return 0;
+}
+
+static void hal2_free_dmabuf(struct hal2_codec *codec)
+{
+	dma_free_noncoherent(NULL, codec->desc_count * sizeof(struct hal2_desc),
+			     codec->desc, codec->desc_dma);
+	dma_free_noncoherent(NULL, H2_BUF_SIZE, codec->buffer,
+			     codec->buffer_dma);
+}
+
+static struct snd_pcm_hardware hal2_pcm_hw = {
+	.info = (SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER),
+	.formats =          SNDRV_PCM_FMTBIT_S16_BE,
+	.rates =            SNDRV_PCM_RATE_8000_48000,
+	.rate_min =         8000,
+	.rate_max =         48000,
+	.channels_min =     2,
+	.channels_max =     2,
+	.buffer_bytes_max = 65536,
+	.period_bytes_min = 1024,
+	.period_bytes_max = 65536,
+	.periods_min =      2,
+	.periods_max =      1024,
+};
+
+static int hal2_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	int err;
+
+	err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static int hal2_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int hal2_playback_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	int err;
+
+	runtime->hw = hal2_pcm_hw;
+
+	err = hal2_alloc_dmabuf(&hal2->dac);
+	if (err)
+		return err;
+	return 0;
+}
+
+static int hal2_playback_close(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	hal2_free_dmabuf(&hal2->dac);
+	return 0;
+}
+
+static int hal2_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hal2_codec *dac = &hal2->dac;
+
+	dac->voices = runtime->channels;
+	dac->sample_rate = hal2_compute_rate(dac, runtime->rate);
+	memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect));
+	dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+	dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	dac->substream = substream;
+	hal2_setup_dac(hal2);
+	return 0;
+}
+
+static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma;
+		hal2->dac.pcm_indirect.hw_data = 0;
+		substream->ops->ack(substream);
+		hal2_start_dac(hal2);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		hal2_stop_dac(hal2);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t
+hal2_playback_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *dac = &hal2->dac;
+
+	return snd_pcm_indirect_playback_pointer(substream, &dac->pcm_indirect,
+						 dac->pbus.pbus->pbdma_bptr);
+}
+
+static void hal2_playback_transfer(struct snd_pcm_substream *substream,
+				   struct snd_pcm_indirect *rec, size_t bytes)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	unsigned char *buf = hal2->dac.buffer + rec->hw_data;
+
+	memcpy(buf, substream->runtime->dma_area + rec->sw_data, bytes);
+	dma_cache_sync(NULL, buf, bytes, DMA_TO_DEVICE);
+
+}
+
+static int hal2_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *dac = &hal2->dac;
+
+	dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+	snd_pcm_indirect_playback_transfer(substream,
+					   &dac->pcm_indirect,
+					   hal2_playback_transfer);
+	return 0;
+}
+
+static int hal2_capture_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *adc = &hal2->adc;
+	int err;
+
+	runtime->hw = hal2_pcm_hw;
+
+	err = hal2_alloc_dmabuf(adc);
+	if (err)
+		return err;
+	return 0;
+}
+
+static int hal2_capture_close(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	hal2_free_dmabuf(&hal2->adc);
+	return 0;
+}
+
+static int hal2_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hal2_codec *adc = &hal2->adc;
+
+	adc->voices = runtime->channels;
+	adc->sample_rate = hal2_compute_rate(adc, runtime->rate);
+	memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect));
+	adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+	adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+	adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	adc->substream = substream;
+	hal2_setup_adc(hal2);
+	return 0;
+}
+
+static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma;
+		hal2->adc.pcm_indirect.hw_data = 0;
+		printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma);
+		hal2_start_adc(hal2);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		hal2_stop_adc(hal2);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t
+hal2_capture_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *adc = &hal2->adc;
+
+	return snd_pcm_indirect_capture_pointer(substream, &adc->pcm_indirect,
+						adc->pbus.pbus->pbdma_bptr);
+}
+
+static void hal2_capture_transfer(struct snd_pcm_substream *substream,
+				  struct snd_pcm_indirect *rec, size_t bytes)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	unsigned char *buf = hal2->adc.buffer + rec->hw_data;
+
+	dma_cache_sync(NULL, buf, bytes, DMA_FROM_DEVICE);
+	memcpy(substream->runtime->dma_area + rec->sw_data, buf, bytes);
+}
+
+static int hal2_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *adc = &hal2->adc;
+
+	snd_pcm_indirect_capture_transfer(substream,
+					  &adc->pcm_indirect,
+					  hal2_capture_transfer);
+	return 0;
+}
+
+static struct snd_pcm_ops hal2_playback_ops = {
+	.open =        hal2_playback_open,
+	.close =       hal2_playback_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   hal2_pcm_hw_params,
+	.hw_free =     hal2_pcm_hw_free,
+	.prepare =     hal2_playback_prepare,
+	.trigger =     hal2_playback_trigger,
+	.pointer =     hal2_playback_pointer,
+	.ack =         hal2_playback_ack,
+};
+
+static struct snd_pcm_ops hal2_capture_ops = {
+	.open =        hal2_capture_open,
+	.close =       hal2_capture_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   hal2_pcm_hw_params,
+	.hw_free =     hal2_pcm_hw_free,
+	.prepare =     hal2_capture_prepare,
+	.trigger =     hal2_capture_trigger,
+	.pointer =     hal2_capture_pointer,
+	.ack =         hal2_capture_ack,
+};
+
+static int __devinit hal2_pcm_create(struct snd_hal2 *hal2)
+{
+	struct snd_pcm *pcm;
+	int err;
+
+	/* create first pcm device with one outputs and one input */
+	err = snd_pcm_new(hal2->card, "SGI HAL2 Audio", 0, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = hal2;
+	strcpy(pcm->name, "SGI HAL2");
+
+	/* set operators */
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+			&hal2_playback_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+			&hal2_capture_ops);
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+					   snd_dma_continuous_data(GFP_KERNEL),
+					   0, 1024 * 1024);
+
+	return 0;
+}
+
+static int hal2_dev_free(struct snd_device *device)
+{
+	struct snd_hal2 *hal2 = device->device_data;
+
+	free_irq(SGI_HPCDMA_IRQ, hal2);
+	kfree(hal2);
+	return 0;
+}
+
+static struct snd_device_ops hal2_ops = {
+	.dev_free = hal2_dev_free,
+};
+
+static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3,
+			    int index)
+{
+	codec->pbus.pbusnr = index;
+	codec->pbus.pbus = &hpc3->pbdma[index];
+}
+
+static int hal2_detect(struct snd_hal2 *hal2)
+{
+	unsigned short board, major, minor;
+	unsigned short rev;
+
+	/* reset HAL2 */
+	hal2_write(0, &hal2->ctl_regs->isr);
+
+	/* release reset */
+	hal2_write(H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N,
+		   &hal2->ctl_regs->isr);
+
+
+	hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE);
+	rev = hal2_read(&hal2->ctl_regs->rev);
+	if (rev & H2_REV_AUDIO_PRESENT)
+		return -ENODEV;
+
+	board = (rev & H2_REV_BOARD_M) >> 12;
+	major = (rev & H2_REV_MAJOR_CHIP_M) >> 4;
+	minor = (rev & H2_REV_MINOR_CHIP_M);
+
+	printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n",
+	       board, major, minor);
+
+	return 0;
+}
+
+static int hal2_create(struct snd_card *card, struct snd_hal2 **rchip)
+{
+	struct snd_hal2 *hal2;
+	struct hpc3_regs *hpc3 = hpc3c0;
+	int err;
+
+	hal2 = kzalloc(sizeof(struct snd_hal2), GFP_KERNEL);
+	if (!hal2)
+		return -ENOMEM;
+
+	hal2->card = card;
+
+	if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED,
+			"SGI HAL2", hal2)) {
+		printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ);
+		kfree(hal2);
+		return -EAGAIN;
+	}
+
+	hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0];
+	hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1];
+	hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2];
+	hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3];
+
+	if (hal2_detect(hal2) < 0) {
+		kfree(hal2);
+		return -ENODEV;
+	}
+
+	hal2_init_codec(&hal2->dac, hpc3, 0);
+	hal2_init_codec(&hal2->adc, hpc3, 1);
+
+	/*
+	 * All DMA channel interfaces in HAL2 are designed to operate with
+	 * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles
+	 * in D5. HAL2 is a 16-bit device which can accept both big and little
+	 * endian format. It assumes that even address bytes are on high
+	 * portion of PBUS (15:8) and assumes that HPC3 is programmed to
+	 * accept a live (unsynchronized) version of P_DREQ_N from HAL2.
+	 */
+#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \
+			  (2 << HPC3_DMACFG_D4R_SHIFT) | \
+			  (2 << HPC3_DMACFG_D5R_SHIFT) | \
+			  (0 << HPC3_DMACFG_D3W_SHIFT) | \
+			  (2 << HPC3_DMACFG_D4W_SHIFT) | \
+			  (2 << HPC3_DMACFG_D5W_SHIFT) | \
+				HPC3_DMACFG_DS16 | \
+				HPC3_DMACFG_EVENHI | \
+				HPC3_DMACFG_RTIME | \
+			  (8 << HPC3_DMACFG_BURST_SHIFT) | \
+				HPC3_DMACFG_DRQLIVE)
+	/*
+	 * Ignore what's mentioned in the specification and write value which
+	 * works in The Real World (TM)
+	 */
+	hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844;
+	hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844;
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, hal2, &hal2_ops);
+	if (err < 0) {
+		free_irq(SGI_HPCDMA_IRQ, hal2);
+		kfree(hal2);
+		return err;
+	}
+	*rchip = hal2;
+	return 0;
+}
+
+static int __devinit hal2_probe(struct platform_device *pdev)
+{
+	struct snd_card *card;
+	struct snd_hal2 *chip;
+	int err;
+
+	card = snd_card_new(index, id, THIS_MODULE, 0);
+	if (card == NULL)
+		return -ENOMEM;
+
+	err = hal2_create(card, &chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	snd_card_set_dev(card, &pdev->dev);
+
+	err = hal2_pcm_create(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	err = hal2_mixer_create(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	strcpy(card->driver, "SGI HAL2 Audio");
+	strcpy(card->shortname, "SGI HAL2 Audio");
+	sprintf(card->longname, "%s irq %i",
+		card->shortname,
+		SGI_HPCDMA_IRQ);
+
+	err = snd_card_register(card);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	platform_set_drvdata(pdev, card);
+	return 0;
+}
+
+static int __exit hal2_remove(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+
+	snd_card_free(card);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver hal2_driver = {
+	.probe	= hal2_probe,
+	.remove	= __devexit_p(hal2_remove),
+	.driver = {
+		.name	= "sgihal2",
+		.owner	= THIS_MODULE,
+	}
+};
+
+static int __init alsa_card_hal2_init(void)
+{
+	return platform_driver_register(&hal2_driver);
+}
+
+static void __exit alsa_card_hal2_exit(void)
+{
+	platform_driver_unregister(&hal2_driver);
+}
+
+module_init(alsa_card_hal2_init);
+module_exit(alsa_card_hal2_exit);

sound/mips/hal2.h

@@ -0,0 +1,245 @@
+#ifndef __HAL2_H
+#define __HAL2_H
+
+/*
+ *  Driver for HAL2 sound processors
+ *  Copyright (c) 1999 Ulf Carlsson <ulfc@bun.falkenberg.se>
+ *  Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis@linux-mips.org>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ *  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., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/types.h>
+
+/* Indirect status register */
+
+#define H2_ISR_TSTATUS		0x01	/* RO: transaction status 1=busy */
+#define H2_ISR_USTATUS		0x02	/* RO: utime status bit 1=armed */
+#define H2_ISR_QUAD_MODE	0x04	/* codec mode 0=indigo 1=quad */
+#define H2_ISR_GLOBAL_RESET_N	0x08	/* chip global reset 0=reset */
+#define H2_ISR_CODEC_RESET_N	0x10	/* codec/synth reset 0=reset  */
+
+/* Revision register */
+
+#define H2_REV_AUDIO_PRESENT	0x8000	/* RO: audio present 0=present */
+#define H2_REV_BOARD_M		0x7000	/* RO: bits 14:12, board revision */
+#define H2_REV_MAJOR_CHIP_M	0x00F0	/* RO: bits 7:4, major chip revision */
+#define H2_REV_MINOR_CHIP_M	0x000F	/* RO: bits 3:0, minor chip revision */
+
+/* Indirect address register */
+
+/*
+ * Address of indirect internal register to be accessed. A write to this
+ * register initiates read or write access to the indirect registers in the
+ * HAL2. Note that there af four indirect data registers for write access to
+ * registers larger than 16 byte.
+ */
+
+#define H2_IAR_TYPE_M		0xF000	/* bits 15:12, type of functional */
+					/* block the register resides in */
+					/* 1=DMA Port */
+					/* 9=Global DMA Control */
+					/* 2=Bresenham */
+					/* 3=Unix Timer */
+#define H2_IAR_NUM_M		0x0F00	/* bits 11:8 instance of the */
+					/* blockin which the indirect */
+					/* register resides */
+					/* If IAR_TYPE_M=DMA Port: */
+					/* 1=Synth In */
+					/* 2=AES In */
+					/* 3=AES Out */
+					/* 4=DAC Out */
+					/* 5=ADC Out */
+					/* 6=Synth Control */
+					/* If IAR_TYPE_M=Global DMA Control: */
+					/* 1=Control */
+					/* If IAR_TYPE_M=Bresenham: */
+					/* 1=Bresenham Clock Gen 1 */
+					/* 2=Bresenham Clock Gen 2 */
+					/* 3=Bresenham Clock Gen 3 */
+					/* If IAR_TYPE_M=Unix Timer: */
+					/* 1=Unix Timer */
+#define H2_IAR_ACCESS_SELECT	0x0080	/* 1=read 0=write */
+#define H2_IAR_PARAM		0x000C	/* Parameter Select */
+#define H2_IAR_RB_INDEX_M	0x0003	/* Read Back Index */
+					/* 00:word0 */
+					/* 01:word1 */
+					/* 10:word2 */
+					/* 11:word3 */
+/*
+ * HAL2 internal addressing
+ *
+ * The HAL2 has "indirect registers" (idr) which are accessed by writing to the
+ * Indirect Data registers. Write the address to the Indirect Address register
+ * to transfer the data.
+ *
+ * We define the H2IR_* to the read address and H2IW_* to the write address and
+ * H2I_* to be fields in whatever register is referred to.
+ *
+ * When we write to indirect registers which are larger than one word (16 bit)
+ * we have to fill more than one indirect register before writing. When we read
+ * back however we have to read several times, each time with different Read
+ * Back Indexes (there are defs for doing this easily).
+ */
+
+/*
+ * Relay Control
+ */
+#define H2I_RELAY_C		0x9100
+#define H2I_RELAY_C_STATE	0x01		/* state of RELAY pin signal */
+
+/* DMA port enable */
+
+#define H2I_DMA_PORT_EN		0x9104
+#define H2I_DMA_PORT_EN_SY_IN	0x01		/* Synth_in DMA port */
+#define H2I_DMA_PORT_EN_AESRX	0x02		/* AES receiver DMA port */
+#define H2I_DMA_PORT_EN_AESTX	0x04		/* AES transmitter DMA port */
+#define H2I_DMA_PORT_EN_CODECTX	0x08		/* CODEC transmit DMA port */
+#define H2I_DMA_PORT_EN_CODECR	0x10		/* CODEC receive DMA port */
+
+#define H2I_DMA_END		0x9108 		/* global dma endian select */
+#define H2I_DMA_END_SY_IN	0x01		/* Synth_in DMA port */
+#define H2I_DMA_END_AESRX	0x02		/* AES receiver DMA port */
+#define H2I_DMA_END_AESTX	0x04		/* AES transmitter DMA port */
+#define H2I_DMA_END_CODECTX	0x08		/* CODEC transmit DMA port */
+#define H2I_DMA_END_CODECR	0x10		/* CODEC receive DMA port */
+						/* 0=b_end 1=l_end */
+
+#define H2I_DMA_DRV		0x910C  	/* global PBUS DMA enable */
+
+#define H2I_SYNTH_C		0x1104		/* Synth DMA control */
+
+#define H2I_AESRX_C		0x1204	 	/* AES RX dma control */
+
+#define H2I_C_TS_EN		0x20		/* Timestamp enable */
+#define H2I_C_TS_FRMT		0x40		/* Timestamp format */
+#define H2I_C_NAUDIO		0x80		/* Sign extend */
+
+/* AESRX CTL, 16 bit */
+
+#define H2I_AESTX_C		0x1304		/* AES TX DMA control */
+#define H2I_AESTX_C_CLKID_SHIFT	3		/* Bresenham Clock Gen 1-3 */
+#define H2I_AESTX_C_CLKID_M	0x18
+#define H2I_AESTX_C_DATAT_SHIFT	8		/* 1=mono 2=stereo (3=quad) */
+#define H2I_AESTX_C_DATAT_M	0x300
+
+/* CODEC registers */
+
+#define H2I_DAC_C1		0x1404 		/* DAC DMA control, 16 bit */
+#define H2I_DAC_C2		0x1408		/* DAC DMA control, 32 bit */
+#define H2I_ADC_C1		0x1504 		/* ADC DMA control, 16 bit */
+#define H2I_ADC_C2		0x1508		/* ADC DMA control, 32 bit */
+
+/* Bits in CTL1 register */
+
+#define H2I_C1_DMA_SHIFT	0		/* DMA channel */
+#define H2I_C1_DMA_M		0x7
+#define H2I_C1_CLKID_SHIFT	3		/* Bresenham Clock Gen 1-3 */
+#define H2I_C1_CLKID_M		0x18
+#define H2I_C1_DATAT_SHIFT	8		/* 1=mono 2=stereo (3=quad) */
+#define H2I_C1_DATAT_M		0x300
+
+/* Bits in CTL2 register */
+
+#define H2I_C2_R_GAIN_SHIFT	0		/* right a/d input gain */
+#define H2I_C2_R_GAIN_M		0xf
+#define H2I_C2_L_GAIN_SHIFT	4		/* left a/d input gain */
+#define H2I_C2_L_GAIN_M		0xf0
+#define H2I_C2_R_SEL		0x100		/* right input select */
+#define H2I_C2_L_SEL		0x200		/* left input select */
+#define H2I_C2_MUTE		0x400		/* mute */
+#define H2I_C2_DO1		0x00010000	/* digital output port bit 0 */
+#define H2I_C2_DO2		0x00020000	/* digital output port bit 1 */
+#define H2I_C2_R_ATT_SHIFT	18		/* right d/a output - */
+#define H2I_C2_R_ATT_M		0x007c0000	/* attenuation */
+#define H2I_C2_L_ATT_SHIFT	23		/* left d/a output - */
+#define H2I_C2_L_ATT_M		0x0f800000	/* attenuation */
+
+#define H2I_SYNTH_MAP_C		0x1104		/* synth dma handshake ctrl */
+
+/* Clock generator CTL 1, 16 bit */
+
+#define H2I_BRES1_C1		0x2104
+#define H2I_BRES2_C1		0x2204
+#define H2I_BRES3_C1		0x2304
+
+#define H2I_BRES_C1_SHIFT	0		/* 0=48.0 1=44.1 2=aes_rx */
+#define H2I_BRES_C1_M		0x03
+
+/* Clock generator CTL 2, 32 bit */
+
+#define H2I_BRES1_C2		0x2108
+#define H2I_BRES2_C2		0x2208
+#define H2I_BRES3_C2		0x2308
+
+#define H2I_BRES_C2_INC_SHIFT	0		/* increment value */
+#define H2I_BRES_C2_INC_M	0xffff
+#define H2I_BRES_C2_MOD_SHIFT	16		/* modcontrol value */
+#define H2I_BRES_C2_MOD_M	0xffff0000	/* modctrl=0xffff&(modinc-1) */
+
+/* Unix timer, 64 bit */
+
+#define H2I_UTIME		0x3104
+#define H2I_UTIME_0_LD		0xffff		/* microseconds, LSB's */
+#define H2I_UTIME_1_LD0		0x0f		/* microseconds, MSB's */
+#define H2I_UTIME_1_LD1		0xf0		/* tenths of microseconds */
+#define H2I_UTIME_2_LD		0xffff		/* seconds, LSB's */
+#define H2I_UTIME_3_LD		0xffff		/* seconds, MSB's */
+
+struct hal2_ctl_regs {
+	u32 _unused0[4];
+	u32 isr;		/* 0x10 Status Register */
+	u32 _unused1[3];
+	u32 rev;		/* 0x20 Revision Register */
+	u32 _unused2[3];
+	u32 iar;		/* 0x30 Indirect Address Register */
+	u32 _unused3[3];
+	u32 idr0;		/* 0x40 Indirect Data Register 0 */
+	u32 _unused4[3];
+	u32 idr1;		/* 0x50 Indirect Data Register 1 */
+	u32 _unused5[3];
+	u32 idr2;		/* 0x60 Indirect Data Register 2 */
+	u32 _unused6[3];
+	u32 idr3;		/* 0x70 Indirect Data Register 3 */
+};
+
+struct hal2_aes_regs {
+	u32 rx_stat[2];	/* Status registers */
+	u32 rx_cr[2];		/* Control registers */
+	u32 rx_ud[4];		/* User data window */
+	u32 rx_st[24];		/* Channel status data */
+
+	u32 tx_stat[1];	/* Status register */
+	u32 tx_cr[3];		/* Control registers */
+	u32 tx_ud[4];		/* User data window */
+	u32 tx_st[24];		/* Channel status data */
+};
+
+struct hal2_vol_regs {
+	u32 right;		/* Right volume */
+	u32 left;		/* Left volume */
+};
+
+struct hal2_syn_regs {
+	u32 _unused0[2];
+	u32 page;		/* DOC Page register */
+	u32 regsel;		/* DOC Register selection */
+	u32 dlow;		/* DOC Data low */
+	u32 dhigh;		/* DOC Data high */
+	u32 irq;		/* IRQ Status */
+	u32 dram;		/* DRAM Access */
+};
+
+#endif	/* __HAL2_H */

sound/mips/sgio2audio.c

@@ -0,0 +1,1006 @@
+/*
+ *   Sound driver for Silicon Graphics O2 Workstations A/V board audio.
+ *
+ *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ *   Mxier part taken from mace_audio.c:
+ *   Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
+ *
+ *   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/delay.h>
+#include <linux/spinlock.h>
+#include <linux/gfp.h>
+#include <linux/vmalloc.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <asm/ip32/ip32_ints.h>
+#include <asm/ip32/mace.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#define SNDRV_GET_ID
+#include <sound/initval.h>
+#include <sound/ad1843.h>
+
+
+MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
+MODULE_DESCRIPTION("SGI O2 Audio");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
+
+static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
+
+
+#define AUDIO_CONTROL_RESET              BIT(0) /* 1: reset audio interface */
+#define AUDIO_CONTROL_CODEC_PRESENT      BIT(1) /* 1: codec detected */
+
+#define CODEC_CONTROL_WORD_SHIFT        0
+#define CODEC_CONTROL_READ              BIT(16)
+#define CODEC_CONTROL_ADDRESS_SHIFT     17
+
+#define CHANNEL_CONTROL_RESET           BIT(10) /* 1: reset channel */
+#define CHANNEL_DMA_ENABLE              BIT(9)  /* 1: enable DMA transfer */
+#define CHANNEL_INT_THRESHOLD_DISABLED  (0 << 5) /* interrupt disabled */
+#define CHANNEL_INT_THRESHOLD_25        (1 << 5) /* int on buffer >25% full */
+#define CHANNEL_INT_THRESHOLD_50        (2 << 5) /* int on buffer >50% full */
+#define CHANNEL_INT_THRESHOLD_75        (3 << 5) /* int on buffer >75% full */
+#define CHANNEL_INT_THRESHOLD_EMPTY     (4 << 5) /* int on buffer empty */
+#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
+#define CHANNEL_INT_THRESHOLD_FULL      (6 << 5) /* int on buffer empty */
+#define CHANNEL_INT_THRESHOLD_NOT_FULL  (7 << 5) /* int on buffer !empty */
+
+#define CHANNEL_RING_SHIFT              12
+#define CHANNEL_RING_SIZE               (1 << CHANNEL_RING_SHIFT)
+#define CHANNEL_RING_MASK               (CHANNEL_RING_SIZE - 1)
+
+#define CHANNEL_LEFT_SHIFT 40
+#define CHANNEL_RIGHT_SHIFT 8
+
+struct snd_sgio2audio_chan {
+	int idx;
+	struct snd_pcm_substream *substream;
+	int pos;
+	snd_pcm_uframes_t size;
+	spinlock_t lock;
+};
+
+/* definition of the chip-specific record */
+struct snd_sgio2audio {
+	struct snd_card *card;
+
+	/* codec */
+	struct snd_ad1843 ad1843;
+	spinlock_t ad1843_lock;
+
+	/* channels */
+	struct snd_sgio2audio_chan channel[3];
+
+	/* resources */
+	void *ring_base;
+	dma_addr_t ring_base_dma;
+};
+
+/* AD1843 access */
+
+/*
+ * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
+ *
+ * Returns unsigned register value on success, -errno on failure.
+ */
+static int read_ad1843_reg(void *priv, int reg)
+{
+	struct snd_sgio2audio *chip = priv;
+	int val;
+	unsigned long flags;
+
+	spin_lock_irqsave(&chip->ad1843_lock, flags);
+
+	writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
+	       CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
+	wmb();
+	val = readq(&mace->perif.audio.codec_control); /* flush bus */
+	udelay(200);
+
+	val = readq(&mace->perif.audio.codec_read);
+
+	spin_unlock_irqrestore(&chip->ad1843_lock, flags);
+	return val;
+}
+
+/*
+ * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
+ */
+static int write_ad1843_reg(void *priv, int reg, int word)
+{
+	struct snd_sgio2audio *chip = priv;
+	int val;
+	unsigned long flags;
+
+	spin_lock_irqsave(&chip->ad1843_lock, flags);
+
+	writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
+	       (word << CODEC_CONTROL_WORD_SHIFT),
+	       &mace->perif.audio.codec_control);
+	wmb();
+	val = readq(&mace->perif.audio.codec_control); /* flush bus */
+	udelay(200);
+
+	spin_unlock_irqrestore(&chip->ad1843_lock, flags);
+	return 0;
+}
+
+static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
+					     (int)kcontrol->private_value);
+	return 0;
+}
+
+static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+	int vol;
+
+	vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
+
+	ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
+	ucontrol->value.integer.value[1] = vol & 0xFF;
+
+	return 0;
+}
+
+static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+	int newvol, oldvol;
+
+	oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
+	newvol = (ucontrol->value.integer.value[0] << 8) |
+		ucontrol->value.integer.value[1];
+
+	newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
+		newvol);
+
+	return newvol != oldvol;
+}
+
+static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	static const char *texts[3] = {
+		"Cam Mic", "Mic", "Line"
+	};
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 3;
+	if (uinfo->value.enumerated.item >= 3)
+		uinfo->value.enumerated.item = 1;
+	strcpy(uinfo->value.enumerated.name,
+	       texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
+	return 0;
+}
+
+static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+	int newsrc, oldsrc;
+
+	oldsrc = ad1843_get_recsrc(&chip->ad1843);
+	newsrc = ad1843_set_recsrc(&chip->ad1843,
+				   ucontrol->value.enumerated.item[0]);
+
+	return newsrc != oldsrc;
+}
+
+/* dac1/pcm0 mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "PCM Playback Volume",
+	.index          = 0,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_PCM_0,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* dac2/pcm1 mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "PCM Playback Volume",
+	.index          = 1,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_PCM_1,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* record level mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Capture Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_RECLEV,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* record level source control */
+static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Capture Source",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info           = sgio2audio_source_info,
+	.get            = sgio2audio_source_get,
+	.put            = sgio2audio_source_put,
+};
+
+/* line mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Line Playback Volume",
+	.index          = 0,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_LINE,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* cd mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Line Playback Volume",
+	.index          = 1,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_LINE_2,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* mic mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Mic Playback Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_MIC,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+
+static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
+{
+	int err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
+	if (err < 0)
+		return err;
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_line, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+/* low-level audio interface DMA */
+
+/* get data out of bounce buffer, count must be a multiple of 32 */
+/* returns 1 if a period has elapsed */
+static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
+					unsigned int ch, unsigned int count)
+{
+	int ret;
+	unsigned long src_base, src_pos, dst_mask;
+	unsigned char *dst_base;
+	int dst_pos;
+	u64 *src;
+	s16 *dst;
+	u64 x;
+	unsigned long flags;
+	struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
+
+	spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+	src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
+	src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
+	dst_base = runtime->dma_area;
+	dst_pos = chip->channel[ch].pos;
+	dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
+
+	/* check if a period has elapsed */
+	chip->channel[ch].size += (count >> 3); /* in frames */
+	ret = chip->channel[ch].size >= runtime->period_size;
+	chip->channel[ch].size %= runtime->period_size;
+
+	while (count) {
+		src = (u64 *)(src_base + src_pos);
+		dst = (s16 *)(dst_base + dst_pos);
+
+		x = *src;
+		dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
+		dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
+
+		src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
+		dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
+		count -= sizeof(u64);
+	}
+
+	writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
+	chip->channel[ch].pos = dst_pos;
+
+	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+	return ret;
+}
+
+/* put some DMA data in bounce buffer, count must be a multiple of 32 */
+/* returns 1 if a period has elapsed */
+static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
+					unsigned int ch, unsigned int count)
+{
+	int ret;
+	s64 l, r;
+	unsigned long dst_base, dst_pos, src_mask;
+	unsigned char *src_base;
+	int src_pos;
+	u64 *dst;
+	s16 *src;
+	unsigned long flags;
+	struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
+
+	spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+	dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
+	dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
+	src_base = runtime->dma_area;
+	src_pos = chip->channel[ch].pos;
+	src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
+
+	/* check if a period has elapsed */
+	chip->channel[ch].size += (count >> 3); /* in frames */
+	ret = chip->channel[ch].size >= runtime->period_size;
+	chip->channel[ch].size %= runtime->period_size;
+
+	while (count) {
+		src = (s16 *)(src_base + src_pos);
+		dst = (u64 *)(dst_base + dst_pos);
+
+		l = src[0]; /* sign extend */
+		r = src[1]; /* sign extend */
+
+		*dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
+			((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
+
+		dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
+		src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
+		count -= sizeof(u64);
+	}
+
+	writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
+	chip->channel[ch].pos = src_pos;
+
+	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+	return ret;
+}
+
+static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+	int ch = chan->idx;
+
+	/* reset DMA channel */
+	writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
+	udelay(10);
+	writeq(0, &mace->perif.audio.chan[ch].control);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* push a full buffer */
+		snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
+	}
+	/* set DMA to wake on 50% empty and enable interrupt */
+	writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
+	       &mace->perif.audio.chan[ch].control);
+	return 0;
+}
+
+static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+
+	writeq(0, &mace->perif.audio.chan[chan->idx].control);
+	return 0;
+}
+
+static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
+{
+	struct snd_sgio2audio_chan *chan = dev_id;
+	struct snd_pcm_substream *substream;
+	struct snd_sgio2audio *chip;
+	int count, ch;
+
+	substream = chan->substream;
+	chip = snd_pcm_substream_chip(substream);
+	ch = chan->idx;
+
+	/* empty the ring */
+	count = CHANNEL_RING_SIZE -
+		readq(&mace->perif.audio.chan[ch].depth) - 32;
+	if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
+		snd_pcm_period_elapsed(substream);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
+{
+	struct snd_sgio2audio_chan *chan = dev_id;
+	struct snd_pcm_substream *substream;
+	struct snd_sgio2audio *chip;
+	int count, ch;
+
+	substream = chan->substream;
+	chip = snd_pcm_substream_chip(substream);
+	ch = chan->idx;
+	/* fill the ring */
+	count = CHANNEL_RING_SIZE -
+		readq(&mace->perif.audio.chan[ch].depth) - 32;
+	if (snd_sgio2audio_dma_push_frag(chip, ch, count))
+		snd_pcm_period_elapsed(substream);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
+{
+	struct snd_sgio2audio_chan *chan = dev_id;
+	struct snd_pcm_substream *substream;
+
+	substream = chan->substream;
+	snd_sgio2audio_dma_stop(substream);
+	snd_sgio2audio_dma_start(substream);
+	return IRQ_HANDLED;
+}
+
+/* PCM part */
+/* PCM hardware definition */
+static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
+	.info = (SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER),
+	.formats =          SNDRV_PCM_FMTBIT_S16_BE,
+	.rates =            SNDRV_PCM_RATE_8000_48000,
+	.rate_min =         8000,
+	.rate_max =         48000,
+	.channels_min =     2,
+	.channels_max =     2,
+	.buffer_bytes_max = 65536,
+	.period_bytes_min = 32768,
+	.period_bytes_max = 65536,
+	.periods_min =      1,
+	.periods_max =      1024,
+};
+
+/* PCM playback open callback */
+static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw = snd_sgio2audio_pcm_hw;
+	runtime->private_data = &chip->channel[1];
+	return 0;
+}
+
+static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw = snd_sgio2audio_pcm_hw;
+	runtime->private_data = &chip->channel[2];
+	return 0;
+}
+
+/* PCM capture open callback */
+static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw = snd_sgio2audio_pcm_hw;
+	runtime->private_data = &chip->channel[0];
+	return 0;
+}
+
+/* PCM close callback */
+static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->private_data = NULL;
+	return 0;
+}
+
+
+/* hw_params callback */
+static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int size = params_buffer_bytes(hw_params);
+
+	/* alloc virtual 'dma' area */
+	if (runtime->dma_area)
+		vfree(runtime->dma_area);
+	runtime->dma_area = vmalloc(size);
+	if (runtime->dma_area == NULL)
+		return -ENOMEM;
+	runtime->dma_bytes = size;
+	return 0;
+}
+
+/* hw_free callback */
+static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	if (substream->runtime->dma_area)
+		vfree(substream->runtime->dma_area);
+	substream->runtime->dma_area = NULL;
+	return 0;
+}
+
+/* prepare callback */
+static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+	int ch = chan->idx;
+	unsigned long flags;
+
+	spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+	/* Setup the pseudo-dma transfer pointers.  */
+	chip->channel[ch].pos = 0;
+	chip->channel[ch].size = 0;
+	chip->channel[ch].substream = substream;
+
+	/* set AD1843 format */
+	/* hardware format is always S16_LE */
+	switch (substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		ad1843_setup_dac(&chip->ad1843,
+				 ch - 1,
+				 runtime->rate,
+				 SNDRV_PCM_FORMAT_S16_LE,
+				 runtime->channels);
+		break;
+	case SNDRV_PCM_STREAM_CAPTURE:
+		ad1843_setup_adc(&chip->ad1843,
+				 runtime->rate,
+				 SNDRV_PCM_FORMAT_S16_LE,
+				 runtime->channels);
+		break;
+	}
+	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+	return 0;
+}
+
+/* trigger callback */
+static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
+				      int cmd)
+{
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		/* start the PCM engine */
+		snd_sgio2audio_dma_start(substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		/* stop the PCM engine */
+		snd_sgio2audio_dma_stop(substream);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+/* pointer callback */
+static snd_pcm_uframes_t
+snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+
+	/* get the current hardware pointer */
+	return bytes_to_frames(substream->runtime,
+			       chip->channel[chan->idx].pos);
+}
+
+/* get the physical page pointer on the given offset */
+static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
+					unsigned long offset)
+{
+	return vmalloc_to_page(substream->runtime->dma_area + offset);
+}
+
+/* operators */
+static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
+	.open =        snd_sgio2audio_playback1_open,
+	.close =       snd_sgio2audio_pcm_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   snd_sgio2audio_pcm_hw_params,
+	.hw_free =     snd_sgio2audio_pcm_hw_free,
+	.prepare =     snd_sgio2audio_pcm_prepare,
+	.trigger =     snd_sgio2audio_pcm_trigger,
+	.pointer =     snd_sgio2audio_pcm_pointer,
+	.page =        snd_sgio2audio_page,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
+	.open =        snd_sgio2audio_playback2_open,
+	.close =       snd_sgio2audio_pcm_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   snd_sgio2audio_pcm_hw_params,
+	.hw_free =     snd_sgio2audio_pcm_hw_free,
+	.prepare =     snd_sgio2audio_pcm_prepare,
+	.trigger =     snd_sgio2audio_pcm_trigger,
+	.pointer =     snd_sgio2audio_pcm_pointer,
+	.page =        snd_sgio2audio_page,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
+	.open =        snd_sgio2audio_capture_open,
+	.close =       snd_sgio2audio_pcm_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   snd_sgio2audio_pcm_hw_params,
+	.hw_free =     snd_sgio2audio_pcm_hw_free,
+	.prepare =     snd_sgio2audio_pcm_prepare,
+	.trigger =     snd_sgio2audio_pcm_trigger,
+	.pointer =     snd_sgio2audio_pcm_pointer,
+	.page =        snd_sgio2audio_page,
+};
+
+/*
+ *  definitions of capture are omitted here...
+ */
+
+/* create a pcm device */
+static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
+{
+	struct snd_pcm *pcm;
+	int err;
+
+	/* create first pcm device with one outputs and one input */
+	err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = chip;
+	strcpy(pcm->name, "SGI O2 DAC1");
+
+	/* set operators */
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+			&snd_sgio2audio_playback1_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+			&snd_sgio2audio_capture_ops);
+
+	/* create second  pcm device with one outputs and no input */
+	err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = chip;
+	strcpy(pcm->name, "SGI O2 DAC2");
+
+	/* set operators */
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+			&snd_sgio2audio_playback2_ops);
+
+	return 0;
+}
+
+static struct {
+	int idx;
+	int irq;
+	irqreturn_t (*isr)(int, void *);
+	const char *desc;
+} snd_sgio2_isr_table[] = {
+	{
+		.idx = 0,
+		.irq = MACEISA_AUDIO1_DMAT_IRQ,
+		.isr = snd_sgio2audio_dma_in_isr,
+		.desc = "Capture DMA Channel 0"
+	}, {
+		.idx = 0,
+		.irq = MACEISA_AUDIO1_OF_IRQ,
+		.isr = snd_sgio2audio_error_isr,
+		.desc = "Capture Overflow"
+	}, {
+		.idx = 1,
+		.irq = MACEISA_AUDIO2_DMAT_IRQ,
+		.isr = snd_sgio2audio_dma_out_isr,
+		.desc = "Playback DMA Channel 1"
+	}, {
+		.idx = 1,
+		.irq = MACEISA_AUDIO2_MERR_IRQ,
+		.isr = snd_sgio2audio_error_isr,
+		.desc = "Memory Error Channel 1"
+	}, {
+		.idx = 2,
+		.irq = MACEISA_AUDIO3_DMAT_IRQ,
+		.isr = snd_sgio2audio_dma_out_isr,
+		.desc = "Playback DMA Channel 2"
+	}, {
+		.idx = 2,
+		.irq = MACEISA_AUDIO3_MERR_IRQ,
+		.isr = snd_sgio2audio_error_isr,
+		.desc = "Memory Error Channel 2"
+	}
+};
+
+/* ALSA driver */
+
+static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
+{
+	int i;
+
+	/* reset interface */
+	writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+	udelay(1);
+	writeq(0, &mace->perif.audio.control);
+
+	/* release IRQ's */
+	for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
+		free_irq(snd_sgio2_isr_table[i].irq,
+			 &chip->channel[snd_sgio2_isr_table[i].idx]);
+
+	dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
+			  chip->ring_base, chip->ring_base_dma);
+
+	/* release card data */
+	kfree(chip);
+	return 0;
+}
+
+static int snd_sgio2audio_dev_free(struct snd_device *device)
+{
+	struct snd_sgio2audio *chip = device->device_data;
+
+	return snd_sgio2audio_free(chip);
+}
+
+static struct snd_device_ops ops = {
+	.dev_free = snd_sgio2audio_dev_free,
+};
+
+static int __devinit snd_sgio2audio_create(struct snd_card *card,
+					   struct snd_sgio2audio **rchip)
+{
+	struct snd_sgio2audio *chip;
+	int i, err;
+
+	*rchip = NULL;
+
+	/* check if a codec is attached to the interface */
+	/* (Audio or Audio/Video board present) */
+	if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
+		return -ENOENT;
+
+	chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
+	if (chip == NULL)
+		return -ENOMEM;
+
+	chip->card = card;
+
+	chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
+					     &chip->ring_base_dma, GFP_USER);
+	if (chip->ring_base == NULL) {
+		printk(KERN_ERR
+		       "sgio2audio: could not allocate ring buffers\n");
+		kfree(chip);
+		return -ENOMEM;
+	}
+
+	spin_lock_init(&chip->ad1843_lock);
+
+	/* initialize channels */
+	for (i = 0; i < 3; i++) {
+		spin_lock_init(&chip->channel[i].lock);
+		chip->channel[i].idx = i;
+	}
+
+	/* allocate IRQs */
+	for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
+		if (request_irq(snd_sgio2_isr_table[i].irq,
+				snd_sgio2_isr_table[i].isr,
+				0,
+				snd_sgio2_isr_table[i].desc,
+				&chip->channel[snd_sgio2_isr_table[i].idx])) {
+			snd_sgio2audio_free(chip);
+			printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
+			       snd_sgio2_isr_table[i].irq);
+			return -EBUSY;
+		}
+	}
+
+	/* reset the interface */
+	writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+	udelay(1);
+	writeq(0, &mace->perif.audio.control);
+	msleep_interruptible(1); /* give time to recover */
+
+	/* set ring base */
+	writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
+
+	/* attach the AD1843 codec */
+	chip->ad1843.read = read_ad1843_reg;
+	chip->ad1843.write = write_ad1843_reg;
+	chip->ad1843.chip = chip;
+
+	/* initialize the AD1843 codec */
+	err = ad1843_init(&chip->ad1843);
+	if (err < 0) {
+		snd_sgio2audio_free(chip);
+		return err;
+	}
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0) {
+		snd_sgio2audio_free(chip);
+		return err;
+	}
+	*rchip = chip;
+	return 0;
+}
+
+static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
+{
+	struct snd_card *card;
+	struct snd_sgio2audio *chip;
+	int err;
+
+	card = snd_card_new(index, id, THIS_MODULE, 0);
+	if (card == NULL)
+		return -ENOMEM;
+
+	err = snd_sgio2audio_create(card, &chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	snd_card_set_dev(card, &pdev->dev);
+
+	err = snd_sgio2audio_new_pcm(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	err = snd_sgio2audio_new_mixer(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	strcpy(card->driver, "SGI O2 Audio");
+	strcpy(card->shortname, "SGI O2 Audio");
+	sprintf(card->longname, "%s irq %i-%i",
+		card->shortname,
+		MACEISA_AUDIO1_DMAT_IRQ,
+		MACEISA_AUDIO3_MERR_IRQ);
+
+	err = snd_card_register(card);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	platform_set_drvdata(pdev, card);
+	return 0;
+}
+
+static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+
+	snd_card_free(card);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver sgio2audio_driver = {
+	.probe	= snd_sgio2audio_probe,
+	.remove	= __devexit_p(snd_sgio2audio_remove),
+	.driver = {
+		.name	= "sgio2audio",
+		.owner	= THIS_MODULE,
+	}
+};
+
+static int __init alsa_card_sgio2audio_init(void)
+{
+	return platform_driver_register(&sgio2audio_driver);
+}
+
+static void __exit alsa_card_sgio2audio_exit(void)
+{
+	platform_driver_unregister(&sgio2audio_driver);
+}
+
+module_init(alsa_card_sgio2audio_init)
+module_exit(alsa_card_sgio2audio_exit)

sound/oss/Kconfig

@@ -7,7 +7,7 @@
 
 config SOUND_BCM_CS4297A
 	tristate "Crystal Sound CS4297a (for Swarm)"
-	depends on SOUND_PRIME && SIBYTE_SWARM
+	depends on SIBYTE_SWARM
 	help
 	  The BCM91250A has a Crystal CS4297a on synchronous serial
 	  port B (in addition to the DB-9 serial port).  Say Y or M
@@ -17,7 +17,7 @@ config SOUND_BCM_CS4297A
 
 config SOUND_VWSND
 	tristate "SGI Visual Workstation Sound"
-	depends on SOUND_PRIME && X86_VISWS
+	depends on X86_VISWS
 	help
 	  Say Y or M if you have an SGI Visual Workstation and you want to be
 	  able to use its on-board audio.  Read
@@ -26,19 +26,18 @@ config SOUND_VWSND
 
 config SOUND_HAL2
 	tristate "SGI HAL2 sound (EXPERIMENTAL)"
-	depends on SOUND_PRIME && SGI_IP22 && EXPERIMENTAL
+	depends on SGI_IP22 && EXPERIMENTAL
 	help
 	  Say Y or M if you have an SGI Indy or Indigo2 system and want to be able to
 	  use its on-board A2 audio system.
 
 config SOUND_AU1550_AC97
 	tristate "Au1550/Au1200 AC97 Sound"
-	select SND_AC97_CODEC
-	depends on SOUND_PRIME && (SOC_AU1550 || SOC_AU1200)
+	depends on SOC_AU1550 || SOC_AU1200
 
 config SOUND_TRIDENT
 	tristate "Trident 4DWave DX/NX, SiS 7018 or ALi 5451 PCI Audio Core"
-	depends on SOUND_PRIME && PCI
+	depends on PCI
 	---help---
 	  Say Y or M if you have a PCI sound card utilizing the Trident
 	  4DWave-DX/NX chipset or your mother board chipset has SiS 7018
@@ -79,7 +78,7 @@ config SOUND_TRIDENT
 
 config SOUND_MSNDCLAS
 	tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey"
-	depends on SOUND_PRIME && (m || !STANDALONE) && ISA
+	depends on (m || !STANDALONE) && ISA
 	help
 	  Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or
 	  Monterey (not for the Pinnacle or Fiji).
@@ -143,7 +142,7 @@ config MSNDCLAS_IO
 
 config SOUND_MSNDPIN
 	tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji"
-	depends on SOUND_PRIME && (m || !STANDALONE) && ISA
+	depends on (m || !STANDALONE) && ISA
 	help
 	  Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji.
 	  See <file:Documentation/sound/oss/MultiSound> for important information
@@ -229,7 +228,7 @@ config MSNDPIN_NONPNP
 	  configure the card's resources.
 
 comment "MSND Pinnacle DSP section will be configured to above parameters."
-	depends on SOUND_PRIME && SOUND_MSNDPIN=y && MSNDPIN_NONPNP
+	depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP
 
 config MSNDPIN_CFG
 	hex "MSND Pinnacle config port 250,260,270"
@@ -242,7 +241,7 @@ config MSNDPIN_CFG
 	  Mode".
 
 comment "Pinnacle-specific Device Configuration (0 disables)"
-	depends on SOUND_PRIME && SOUND_MSNDPIN=y && MSNDPIN_NONPNP
+	depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP
 
 config MSNDPIN_MPU_IO
 	hex "MSND Pinnacle MPU I/O (e.g. 330)"
@@ -294,7 +293,7 @@ config MSNDPIN_JOYSTICK_IO
 
 config MSND_FIFOSIZE
 	int "MSND buffer size (kB)"
-	depends on SOUND_PRIME && (SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y)
+	depends on SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y
 	default "128"
 	help
 	  Configures the size of each audio buffer, in kilobytes, for
@@ -302,9 +301,9 @@ config MSND_FIFOSIZE
 	  and Pinnacle). Larger values reduce the chance of data overruns at
 	  the expense of overall latency. If unsure, use the default.
 
-config SOUND_OSS
+menuconfig SOUND_OSS
 	tristate "OSS sound modules"
-	depends on SOUND_PRIME && ISA_DMA_API && VIRT_TO_BUS
+	depends on ISA_DMA_API && VIRT_TO_BUS
 	help
 	  OSS is the Open Sound System suite of sound card drivers.  They make
 	  sound programming easier since they provide a common API.  Say Y or
@@ -312,16 +311,16 @@ config SOUND_OSS
 	  driver for your sound card above, then pick your driver from the
 	  list below.
 
+if SOUND_OSS
+
 config SOUND_TRACEINIT
 	bool "Verbose initialisation"
-	depends on SOUND_OSS
 	help
 	  Verbose soundcard initialization -- affects the format of autoprobe
 	  and initialization messages at boot time.
 
 config SOUND_DMAP
 	bool "Persistent DMA buffers"
-	depends on SOUND_OSS
 	---help---
 	  Linux can often have problems allocating DMA buffers for ISA sound
 	  cards on machines with more than 16MB of RAM. This is because ISA
@@ -338,8 +337,6 @@ config SOUND_DMAP
 
 config SOUND_SSCAPE
 	tristate "Ensoniq SoundScape support"
-	depends on SOUND_OSS
-	depends on VIRT_TO_BUS
 	help
 	  Answer Y if you have a sound card based on the Ensoniq SoundScape
 	  chipset. Such cards are being manufactured at least by Ensoniq, Spea
@@ -352,13 +349,11 @@ config SOUND_SSCAPE
 
 config SOUND_VMIDI
 	tristate "Loopback MIDI device support"
-	depends on SOUND_OSS
 	help
 	  Support for MIDI loopback on port 1 or 2.
 
 config SOUND_TRIX
 	tristate "MediaTrix AudioTrix Pro support"
-	depends on SOUND_OSS
 	help
 	  Answer Y if you have the AudioTriX Pro sound card manufactured
 	  by MediaTrix.
@@ -382,7 +377,6 @@ config TRIX_BOOT_FILE
 
 config SOUND_MSS
 	tristate "Microsoft Sound System support"
-	depends on SOUND_OSS
 	---help---
 	  Again think carefully before answering Y to this question.  It's
 	  safe to answer Y if you have the original Windows Sound System card
@@ -414,7 +408,6 @@ config SOUND_MSS
 
 config SOUND_MPU401
 	tristate "MPU-401 support (NOT for SB16)"
-	depends on SOUND_OSS
 	---help---
 	  Be careful with this question.  The MPU401 interface is supported by
 	  all sound cards.  However, some natively supported cards have their
@@ -430,7 +423,6 @@ config SOUND_MPU401
 
 config SOUND_PAS
 	tristate "ProAudioSpectrum 16 support"
-	depends on SOUND_OSS
 	---help---
 	  Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
 	  16 or Logitech SoundMan 16 sound card. Answer N if you have some
@@ -452,7 +444,6 @@ config PAS_JOYSTICK
 
 config SOUND_PSS
 	tristate "PSS (AD1848, ADSP-2115, ESC614) support"
-	depends on SOUND_OSS
 	help
 	  Answer Y or M if you have an Orchid SW32, Cardinal DSP16, Beethoven
 	  ADSP-16 or some other card based on the PSS chipset (AD1848 codec +
@@ -495,7 +486,6 @@ config PSS_BOOT_FILE
 
 config SOUND_SB
 	tristate "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support"
-	depends on SOUND_OSS
 	---help---
 	  Answer Y if you have an original Sound Blaster card made by Creative
 	  Labs or a 100% hardware compatible clone (like the Thunderboard or
@@ -522,7 +512,6 @@ config SOUND_SB
 
 config SOUND_YM3812
 	tristate "Yamaha FM synthesizer (YM3812/OPL-3) support"
-	depends on SOUND_OSS
 	---help---
 	  Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
 	  Answering Y is usually a safe and recommended choice, however some
@@ -538,7 +527,6 @@ config SOUND_YM3812
 
 config SOUND_UART6850
 	tristate "6850 UART support"
-	depends on SOUND_OSS
 	help
 	  This option enables support for MIDI interfaces based on the 6850
 	  UART chip. This interface is rarely found on sound cards. It's safe
@@ -549,7 +537,6 @@ config SOUND_UART6850
 
 config SOUND_AEDSP16
 	tristate "Gallant Audio Cards (SC-6000 and SC-6600 based)"
-	depends on SOUND_OSS
 	---help---
 	  Answer Y if you have a Gallant's Audio Excel DSP 16 card. This
 	  driver supports Audio Excel DSP 16 but not the III nor PnP versions
@@ -630,14 +617,14 @@ endchoice
 
 config SOUND_VIDC
 	tristate "VIDC 16-bit sound"
-	depends on ARM && (ARCH_ACORN || ARCH_CLPS7500) && SOUND_OSS
+	depends on ARM && (ARCH_ACORN || ARCH_CLPS7500)
 	help
 	  16-bit support for the VIDC onboard sound hardware found on Acorn
 	  machines.
 
 config SOUND_WAVEARTIST
 	tristate "Netwinder WaveArtist"
-	depends on ARM && SOUND_OSS && ARCH_NETWINDER
+	depends on ARM && ARCH_NETWINDER
 	help
 	  Say Y here to include support for the Rockwell WaveArtist sound
 	  system.  This driver is mainly for the NetWinder.
@@ -646,9 +633,11 @@ config SOUND_KAHLUA
 	tristate "XpressAudio Sound Blaster emulation"
 	depends on SOUND_SB
 
+endif	# SOUND_OSS
+
 config SOUND_SH_DAC_AUDIO
 	tristate "SuperH DAC audio support"
-	depends on SOUND_PRIME && CPU_SH3
+	depends on CPU_SH3
 
 config SOUND_SH_DAC_AUDIO_CHANNEL
 	int "DAC channel"

sound/oss/dmasound/dmasound_core.c

@@ -211,10 +211,6 @@ static int state_unit = -1;
 static int irq_installed;
 #endif /* MODULE */
 
-/* software implemented recording volume! */
-uint software_input_volume = SW_INPUT_VOLUME_SCALE * SW_INPUT_VOLUME_DEFAULT;
-EXPORT_SYMBOL(software_input_volume);
-
 /* control over who can modify resources shared between play/record */
 static mode_t shared_resource_owner;
 static int shared_resources_initialised;
@@ -1188,7 +1184,7 @@ static struct {
 
 /* publish this function for use by low-level code, if required */
 
-char *get_afmt_string(int afmt)
+static char *get_afmt_string(int afmt)
 {
         switch(afmt) {
             case AFMT_MU_LAW:
@@ -1551,4 +1547,3 @@ EXPORT_SYMBOL(dmasound_catchRadius);
 EXPORT_SYMBOL(dmasound_ulaw2dma8);
 EXPORT_SYMBOL(dmasound_alaw2dma8);
 #endif
-EXPORT_SYMBOL(get_afmt_string) ;

sound/oss/dmasound/dmasound_paula.c

@@ -710,7 +710,7 @@ static MACHINE machAmiga = {
 /*** Config & Setup **********************************************************/
 
 
-int __init dmasound_paula_init(void)
+static int __init dmasound_paula_init(void)
 {
 	int err;
 

sound/oss/dmasound/dmasound_q40.c

@@ -611,7 +611,7 @@ static MACHINE machQ40 = {
 /*** Config & Setup **********************************************************/
 
 
-int __init dmasound_q40_init(void)
+static int __init dmasound_q40_init(void)
 {
 	if (MACH_IS_Q40) {
 	    dmasound.mach = machQ40;

sound/oss/msnd.c

@@ -20,8 +20,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd.c,v 1.17 1999/03/21 16:50:09 andrewtv Exp $
- *
  ********************************************************************/
 
 #include <linux/module.h>

sound/oss/msnd.h

@@ -24,8 +24,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd.h,v 1.36 1999/03/21 17:05:42 andrewtv Exp $
- *
  ********************************************************************/
 #ifndef __MSND_H
 #define __MSND_H

sound/oss/msnd_classic.h

@@ -24,8 +24,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  * 
- * $Id: msnd_classic.h,v 1.10 1999/03/21 17:36:09 andrewtv Exp $
- *
  ********************************************************************/
 #ifndef __MSND_CLASSIC_H
 #define __MSND_CLASSIC_H

sound/oss/msnd_pinnacle.c

@@ -29,13 +29,8 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd_pinnacle.c,v 1.8 2000/12/30 00:33:21 sycamore Exp $
- *
  * 12-3-2000  Modified IO port validation  Steve Sycamore
  *
- *
- * $$$: msnd_pinnacle.c,v 1.75 1999/03/21 16:50:09 andrewtv $$$ $
- *
  ********************************************************************/
 
 #include <linux/kernel.h>

sound/oss/msnd_pinnacle.h

@@ -24,8 +24,6 @@
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
- * $Id: msnd_pinnacle.h,v 1.11 1999/03/21 17:36:09 andrewtv Exp $
- *
  ********************************************************************/
 #ifndef __MSND_PINNACLE_H
 #define __MSND_PINNACLE_H

sound/parisc/Kconfig

@@ -1,15 +1,20 @@
 # ALSA PA-RISC drivers
 
-menu "GSC devices"
-	depends on SND!=n && GSC
+menuconfig SND_GSC
+	bool "GSC sound devices"
+	depends on GSC
+	default y
+	help
+	  Support for GSC sound devices on PA-RISC architectures.
+
+if SND_GSC
 
 config SND_HARMONY
 	tristate "Harmony/Vivace sound chip"
-	depends on SND
 	select SND_PCM
 	help
 	  Say 'Y' or 'M' to include support for the Harmony/Vivace sound
 	  chip found in most GSC-based PA-RISC workstations.  It's frequently
 	  provided as part of the Lasi multi-function IC.
 
-endmenu
+endif	# SND_GSC

sound/pci/Kconfig

@@ -1,11 +1,16 @@
 # ALSA PCI drivers
 
-menu "PCI devices"
-	depends on SND!=n && PCI
+menuconfig SND_PCI
+	bool "PCI sound devices"
+	depends on PCI
+	default y
+	help
+	  Support for sound devices connected via the PCI bus.
+
+if SND_PCI
 
 config SND_AD1889
 	tristate "Analog Devices AD1889"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated AC97 sound
@@ -17,7 +22,6 @@ config SND_AD1889
 
 config SND_ALS300
 	tristate "Avance Logic ALS300/ALS300+"
-	depends on SND
 	select SND_PCM
 	select SND_AC97_CODEC
 	select SND_OPL3_LIB
@@ -29,7 +33,7 @@ config SND_ALS300
 
 config SND_ALS4000
 	tristate "Avance Logic ALS4000"
-	depends on SND && ISA_DMA_API
+	depends on ISA_DMA_API
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -43,7 +47,6 @@ config SND_ALS4000
 
 config SND_ALI5451
 	tristate "ALi M5451 PCI Audio Controller"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -57,7 +60,6 @@ config SND_ALI5451
 
 config SND_ATIIXP
 	tristate "ATI IXP AC97 Controller"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated AC97 sound
@@ -69,7 +71,6 @@ config SND_ATIIXP
 
 config SND_ATIIXP_MODEM
 	tristate "ATI IXP Modem"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated MC97 modem on
@@ -80,7 +81,6 @@ config SND_ATIIXP_MODEM
 
 config SND_AU8810
 	tristate "Aureal Advantage"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -95,7 +95,6 @@ config SND_AU8810
 
 config SND_AU8820
 	tristate "Aureal Vortex"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -109,7 +108,6 @@ config SND_AU8820
 
 config SND_AU8830
 	tristate "Aureal Vortex 2"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -124,7 +122,6 @@ config SND_AU8830
 
 config SND_AW2
 	tristate "Emagic Audiowerk 2"
-	depends on SND
 	help
 	  Say Y here to include support for Emagic Audiowerk 2 soundcards.
 
@@ -139,7 +136,7 @@ config SND_AW2
 
 config SND_AZT3328
 	tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)"
-	depends on SND && EXPERIMENTAL
+	depends on EXPERIMENTAL
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -152,7 +149,6 @@ config SND_AZT3328
 
 config SND_BT87X
 	tristate "Bt87x Audio Capture"
-	depends on SND
 	select SND_PCM
 	help
 	  If you want to record audio from TV cards based on
@@ -174,7 +170,6 @@ config SND_BT87X_OVERCLOCK
 
 config SND_CA0106
 	tristate "SB Audigy LS / Live 24bit"
-	depends on SND
 	select SND_AC97_CODEC
 	select SND_RAWMIDI
 	select SND_VMASTER
@@ -187,7 +182,6 @@ config SND_CA0106
 
 config SND_CMIPCI
 	tristate "C-Media 8338, 8738, 8768, 8770"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
@@ -201,13 +195,11 @@ config SND_CMIPCI
 
 config SND_OXYGEN_LIB
         tristate
-	depends on SND
 	select SND_PCM
 	select SND_MPU401_UART
 
 config SND_OXYGEN
 	tristate "C-Media 8788 (Oxygen)"
-	depends on SND
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
@@ -225,7 +217,6 @@ config SND_OXYGEN
 
 config SND_CS4281
 	tristate "Cirrus Logic (Sound Fusion) CS4281"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
@@ -237,7 +228,6 @@ config SND_CS4281
 
 config SND_CS46XX
 	tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
 	help
@@ -258,7 +248,7 @@ config SND_CS46XX_NEW_DSP
 
 config SND_CS5530
 	tristate "CS5530 Audio"
-	depends on SND && ISA_DMA_API
+	depends on ISA_DMA_API
 	select SND_SB16_DSP
 	help
 	  Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips.
@@ -268,7 +258,7 @@ config SND_CS5530
 
 config SND_CS5535AUDIO
 	tristate "CS5535/CS5536 Audio"
-	depends on SND && X86 && !X86_64
+	depends on X86 && !X86_64
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
@@ -286,7 +276,6 @@ config SND_CS5535AUDIO
 
 config SND_DARLA20
 	tristate "(Echoaudio) Darla20"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -297,7 +286,6 @@ config SND_DARLA20
 
 config SND_GINA20
 	tristate "(Echoaudio) Gina20"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -308,7 +296,6 @@ config SND_GINA20
 
 config SND_LAYLA20
 	tristate "(Echoaudio) Layla20"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -320,7 +307,6 @@ config SND_LAYLA20
 
 config SND_DARLA24
 	tristate "(Echoaudio) Darla24"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -331,7 +317,6 @@ config SND_DARLA24
 
 config SND_GINA24
 	tristate "(Echoaudio) Gina24"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -342,7 +327,6 @@ config SND_GINA24
 
 config SND_LAYLA24
 	tristate "(Echoaudio) Layla24"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -354,7 +338,6 @@ config SND_LAYLA24
 
 config SND_MONA
 	tristate "(Echoaudio) Mona"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -366,7 +349,6 @@ config SND_MONA
 
 config SND_MIA
 	tristate "(Echoaudio) Mia"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -378,7 +360,6 @@ config SND_MIA
 
 config SND_ECHO3G
 	tristate "(Echoaudio) 3G cards"
-	depends on SND
 	select FW_LOADER
 	select SND_RAWMIDI
 	select SND_PCM
@@ -390,7 +371,6 @@ config SND_ECHO3G
 
 config SND_INDIGO
 	tristate "(Echoaudio) Indigo"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -401,7 +381,6 @@ config SND_INDIGO
 
 config SND_INDIGOIO
 	tristate "(Echoaudio) Indigo IO"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -412,7 +391,6 @@ config SND_INDIGOIO
 
 config SND_INDIGODJ
 	tristate "(Echoaudio) Indigo DJ"
-	depends on SND
 	select FW_LOADER
 	select SND_PCM
 	help
@@ -423,7 +401,6 @@ config SND_INDIGODJ
 
 config SND_EMU10K1
 	tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
-	depends on SND
 	select FW_LOADER
 	select SND_HWDEP
 	select SND_RAWMIDI
@@ -441,7 +418,6 @@ config SND_EMU10K1
 
 config SND_EMU10K1X
 	tristate "Emu10k1X (Dell OEM Version)"
-	depends on SND
 	select SND_AC97_CODEC
 	select SND_RAWMIDI
 	help
@@ -453,7 +429,6 @@ config SND_EMU10K1X
 
 config SND_ENS1370
 	tristate "(Creative) Ensoniq AudioPCI 1370"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_PCM
 	help
@@ -464,7 +439,6 @@ config SND_ENS1370
 
 config SND_ENS1371
 	tristate "(Creative) Ensoniq AudioPCI 1371/1373"
-	depends on SND
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
 	help
@@ -476,7 +450,6 @@ config SND_ENS1371
 
 config SND_ES1938
 	tristate "ESS ES1938/1946/1969 (Solo-1)"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
@@ -489,7 +462,6 @@ config SND_ES1938
 
 config SND_ES1968
 	tristate "ESS ES1968/1978 (Maestro-1/2/2E)"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -501,7 +473,6 @@ config SND_ES1968
 
 config SND_FM801
 	tristate "ForteMedia FM801"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
@@ -528,7 +499,6 @@ config SND_FM801_TEA575X
 
 config SND_HDA_INTEL
 	tristate "Intel HD Audio"
-	depends on SND
 	select SND_PCM
 	select SND_VMASTER
 	help
@@ -637,7 +607,6 @@ config SND_HDA_POWER_SAVE_DEFAULT
 
 config SND_HDSP
 	tristate "RME Hammerfall DSP Audio"
-	depends on SND
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -650,7 +619,6 @@ config SND_HDSP
 
 config SND_HDSPM
 	tristate "RME Hammerfall DSP MADI"
-	depends on SND
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_PCM
@@ -663,7 +631,6 @@ config SND_HDSPM
 
 config SND_HIFIER
 	tristate "TempoTec HiFier Fantasia"
-	depends on SND
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for the MediaTek/TempoTec HiFier
@@ -674,7 +641,6 @@ config SND_HIFIER
 
 config SND_ICE1712
 	tristate "ICEnsemble ICE1712 (Envy24)"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -691,8 +657,7 @@ config SND_ICE1712
 
 config SND_ICE1724
 	tristate "ICE/VT1724/1720 (Envy24HT/PT)"
-	depends on SND
-	select SND_MPU401_UART
+	select SND_RAWMIDI
 	select SND_AC97_CODEC
 	select SND_VMASTER
 	help
@@ -709,7 +674,6 @@ config SND_ICE1724
 
 config SND_INTEL8X0
 	tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated AC97 sound
@@ -722,7 +686,6 @@ config SND_INTEL8X0
 
 config SND_INTEL8X0M
 	tristate "Intel/SiS/nVidia/AMD MC97 Modem"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated MC97 modem on
@@ -733,7 +696,6 @@ config SND_INTEL8X0M
 
 config SND_KORG1212
 	tristate "Korg 1212 IO"
-	depends on SND
 	select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL
 	select SND_PCM
 	help
@@ -753,7 +715,6 @@ config SND_KORG1212_FIRMWARE_IN_KERNEL
 
 config SND_MAESTRO3
 	tristate "ESS Allegro/Maestro3"
-	depends on SND
 	select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL
 	select SND_AC97_CODEC
 	help
@@ -774,7 +735,6 @@ config SND_MAESTRO3_FIRMWARE_IN_KERNEL
 
 config SND_MIXART
 	tristate "Digigram miXart"
-	depends on SND
 	select SND_HWDEP
 	select SND_PCM
 	help
@@ -786,7 +746,6 @@ config SND_MIXART
 
 config SND_NM256
 	tristate "NeoMagic NM256AV/ZX"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for NeoMagic NM256AV/ZX chips.
@@ -796,7 +755,6 @@ config SND_NM256
 
 config SND_PCXHR
 	tristate "Digigram PCXHR"
-	depends on SND
 	select SND_PCM
 	select SND_HWDEP
 	help
@@ -807,7 +765,6 @@ config SND_PCXHR
 
 config SND_RIPTIDE
 	tristate "Conexant Riptide"
-	depends on SND
 	select FW_LOADER
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -820,7 +777,6 @@ config SND_RIPTIDE
 
 config SND_RME32
 	tristate "RME Digi32, 32/8, 32 PRO"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y to include support for RME Digi32, Digi32 PRO and
@@ -832,7 +788,6 @@ config SND_RME32
 
 config SND_RME96
 	tristate "RME Digi96, 96/8, 96/8 PRO"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include support for RME Digi96, Digi96/8 and
@@ -843,7 +798,6 @@ config SND_RME96
 
 config SND_RME9652
 	tristate "RME Digi9652 (Hammerfall)"
-	depends on SND
 	select SND_PCM
 	help
 	  Say Y here to include support for RME Hammerfall (RME
@@ -854,7 +808,7 @@ config SND_RME9652
 
 config SND_SIS7019
 	tristate "SiS 7019 Audio Accelerator"
-	depends on SND && X86 && !X86_64
+	depends on X86 && !X86_64
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the SiS 7019 Audio Accelerator.
@@ -864,7 +818,6 @@ config SND_SIS7019
 
 config SND_SONICVIBES
 	tristate "S3 SonicVibes"
-	depends on SND
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
@@ -877,7 +830,6 @@ config SND_SONICVIBES
 
 config SND_TRIDENT
 	tristate "Trident 4D-Wave DX/NX; SiS 7018"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -889,7 +841,6 @@ config SND_TRIDENT
 
 config SND_VIA82XX
 	tristate "VIA 82C686A/B, 8233/8235 AC97 Controller"
-	depends on SND
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	help
@@ -901,7 +852,6 @@ config SND_VIA82XX
 
 config SND_VIA82XX_MODEM
 	tristate "VIA 82C686A/B, 8233 based Modems"
-	depends on SND
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for the integrated MC97 modem on
@@ -912,7 +862,6 @@ config SND_VIA82XX_MODEM
 
 config SND_VIRTUOSO
 	tristate "Asus Virtuoso 100/200 (Xonar)"
-	depends on SND
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
@@ -923,7 +872,6 @@ config SND_VIRTUOSO
 
 config SND_VX222
 	tristate "Digigram VX222"
-	depends on SND
 	select SND_VX_LIB
 	help
 	  Say Y here to include support for Digigram VX222 soundcards.
@@ -933,7 +881,6 @@ config SND_VX222
 
 config SND_YMFPCI
 	tristate "Yamaha YMF724/740/744/754"
-	depends on SND
 	select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
@@ -954,25 +901,4 @@ config SND_YMFPCI_FIRMWARE_IN_KERNEL
 	  for the YMFPCI driver.  If you choose N here, you need to
 	  install the firmware files from the alsa-firmware package.
 
-config SND_AC97_POWER_SAVE
-	bool "AC97 Power-Saving Mode"
-	depends on SND_AC97_CODEC && EXPERIMENTAL
-	default n
-	help
-	  Say Y here to enable the aggressive power-saving support of
-	  AC97 codecs.  In this mode, the power-mode is dynamically
-	  controlled at each open/close.
-
-	  The mode is activated by passing power_save=1 option to
-	  snd-ac97-codec driver.  You can toggle it dynamically over
-	  sysfs, too.
-
-config SND_AC97_POWER_SAVE_DEFAULT
-	int "Default time-out for AC97 power-save mode"
-	depends on SND_AC97_POWER_SAVE
-	default 0
-	help
-	  The default time-out value in seconds for AC97 automatic
-	  power-save mode.  0 means to disable the power-save mode.
-
-endmenu
+endif	# SND_PCI

sound/pci/Makefile

@@ -13,7 +13,7 @@ snd-bt87x-objs := bt87x.o
 snd-cmipci-objs := cmipci.o
 snd-cs4281-objs := cs4281.o
 snd-cs5530-objs := cs5530.o
-snd-ens1370-objs := ens1370.o
+snd-ens1370-objs := ens1370.o ak4531_codec.o
 snd-ens1371-objs := ens1371.o
 snd-es1938-objs := es1938.o
 snd-es1968-objs := es1968.o

sound/pci/ac97/Makefile

@@ -3,16 +3,8 @@
 # Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
 #
 
-snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o
-
-ifneq ($(CONFIG_PROC_FS),)
-snd-ac97-codec-objs += ac97_proc.o
-endif
-
-snd-ak4531-codec-objs := ak4531_codec.o
+snd-ac97-codec-y := ac97_codec.o ac97_pcm.o
+snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o
 
 # Toplevel Module Dependency
 obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o
-obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o
-
-obj-m := $(sort $(obj-m))

sound/pci/ac97/ac97_codec.c

@@ -49,8 +49,9 @@ MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control");
 
 #ifdef CONFIG_SND_AC97_POWER_SAVE
 static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT;
-module_param(power_save, bool, 0644);
-MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control");
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+		 "(in second, 0 = disable).");
 #endif
 /*
 
@@ -2294,9 +2295,11 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97)
 	power |= AC97_PD_PR0 | AC97_PD_PR1;	/* ADC & DAC powerdown */
 	snd_ac97_write(ac97, AC97_POWERDOWN, power);
 	udelay(100);
-	power |= AC97_PD_PR2 | AC97_PD_PR3;	/* Analog Mixer powerdown */
+	power |= AC97_PD_PR2;	/* Analog Mixer powerdown (Vref on) */
 	snd_ac97_write(ac97, AC97_POWERDOWN, power);
 	if (ac97_is_power_save_mode(ac97)) {
+		power |= AC97_PD_PR3;	/* Analog Mixer powerdown */
+		snd_ac97_write(ac97, AC97_POWERDOWN, power);
 		udelay(100);
 		/* AC-link powerdown, internal Clk disable */
 		/* FIXME: this may cause click noises on some boards */
@@ -2362,7 +2365,7 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup)
 		 *  that open/close frequently)
 		 */
 		schedule_delayed_work(&ac97->power_work,
-				      msecs_to_jiffies(2000));
+				      msecs_to_jiffies(power_save * 1000));
 	else {
 		cancel_delayed_work(&ac97->power_work);
 		update_power_regs(ac97);

sound/pci/ac97/ac97_patch.c

@@ -669,6 +669,7 @@ AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
 AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
 AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
 
+AC97_SINGLE("Master Left Inv Switch", AC97_MASTER, 6, 1, 0),
 AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0),
 AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0),
 AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
@@ -3352,8 +3353,66 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0),
 AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0),
 };
 
+static const char *slave_vols_vt1616[] = {
+	"Front Playback Volume",
+	"Surround Playback Volume",
+	"Center Playback Volume",
+	"LFE Playback Volume",
+	NULL
+};
+
+static const char *slave_sws_vt1616[] = {
+	"Front Playback Switch",
+	"Surround Playback Switch",
+	"Center Playback Switch",
+	"LFE Playback Switch",
+	NULL
+};
+
+/* find a mixer control element with the given name */
+static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
+						    const char *name)
+{
+	struct snd_ctl_elem_id id;
+	memset(&id, 0, sizeof(id));
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	strcpy(id.name, name);
+	return snd_ctl_find_id(ac97->bus->card, &id);
+}
+
+/* create a virtual master control and add slaves */
+int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
+			 const unsigned int *tlv, const char **slaves)
+{
+	struct snd_kcontrol *kctl;
+	const char **s;
+	int err;
+
+	kctl = snd_ctl_make_virtual_master(name, tlv);
+	if (!kctl)
+		return -ENOMEM;
+	err = snd_ctl_add(ac97->bus->card, kctl);
+	if (err < 0)
+		return err;
+
+	for (s = slaves; *s; s++) {
+		struct snd_kcontrol *sctl;
+
+		sctl = snd_ac97_find_mixer_ctl(ac97, *s);
+		if (!sctl) {
+			snd_printdd("Cannot find slave %s, skipped\n", *s);
+			continue;
+		}
+		err = snd_ctl_add_slave(kctl, sctl);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
 static int patch_vt1616_specific(struct snd_ac97 * ac97)
 {
+	struct snd_kcontrol *kctl;
 	int err;
 
 	if (snd_ac97_try_bit(ac97, 0x5a, 9))
@@ -3361,6 +3420,24 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97)
 			return err;
 	if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[1], ARRAY_SIZE(snd_ac97_controls_vt1616) - 1)) < 0)
 		return err;
+
+	/* There is already a misnamed master switch.  Rename it.  */
+	kctl = snd_ac97_find_mixer_ctl(ac97, "Master Playback Volume");
+	if (!kctl)
+		return -EINVAL;
+
+	snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback");
+
+	err = snd_ac97_add_vmaster(ac97, "Master Playback Volume",
+				   kctl->tlv.p, slave_vols_vt1616);
+	if (err < 0)
+		return err;
+
+	err = snd_ac97_add_vmaster(ac97, "Master Playback Switch",
+				   NULL, slave_sws_vt1616);
+	if (err < 0)
+		return err;
+
 	return 0;
 }
 
@@ -3633,7 +3710,7 @@ static int patch_ucb1400(struct snd_ac97 * ac97)
 {
 	ac97->build_ops = &patch_ucb1400_ops;
 	/* enable headphone driver and smart low power mode by default */
-	snd_ac97_write(ac97, 0x6a, 0x0050);
-	snd_ac97_write(ac97, 0x6c, 0x0030);
+	snd_ac97_write_cache(ac97, 0x6a, 0x0050);
+	snd_ac97_write_cache(ac97, 0x6c, 0x0030);
 	return 0;
 }

sound/pci/ac97/ak4531_codec.c → sound/pci/ak4531_codec.c

@@ -28,9 +28,11 @@
 #include <sound/ak4531_codec.h>
 #include <sound/tlv.h>
 
+/*
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Universal routines for AK4531 codec");
 MODULE_LICENSE("GPL");
+*/
 
 #ifdef CONFIG_PROC_FS
 static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531);
@@ -270,7 +272,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0);
 static const DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0);
 static const DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0);
 
-static struct snd_kcontrol_new snd_ak4531_controls[] = {
+static struct snd_kcontrol_new snd_ak4531_controls[] __devinitdata = {
 
 AK4531_DOUBLE_TLV("Master Playback Switch", 0,
 		  AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1,
@@ -379,8 +381,9 @@ static u8 snd_ak4531_initial_map[0x19 + 1] = {
 	0x01		/* 19: Mic Amp Setup */
 };
 
-int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531,
-		     struct snd_ak4531 **rak4531)
+int __devinit snd_ak4531_mixer(struct snd_card *card,
+			       struct snd_ak4531 *_ak4531,
+			       struct snd_ak4531 **rak4531)
 {
 	unsigned int idx;
 	int err;
@@ -476,7 +479,8 @@ static void snd_ak4531_proc_read(struct snd_info_entry *entry,
 		    ak4531->regs[AK4531_MIC_GAIN] & 1 ? "+30dB" : "+0dB");
 }
 
-static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531)
+static void __devinit
+snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531)
 {
 	struct snd_info_entry *entry;
 
@@ -484,25 +488,3 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453
 		snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read);
 }
 #endif
-
-EXPORT_SYMBOL(snd_ak4531_mixer);
-#ifdef CONFIG_PM
-EXPORT_SYMBOL(snd_ak4531_suspend);
-EXPORT_SYMBOL(snd_ak4531_resume);
-#endif
-
-/*
- *  INIT part
- */
-
-static int __init alsa_ak4531_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_ak4531_exit(void)
-{
-}
-
-module_init(alsa_ak4531_init)
-module_exit(alsa_ak4531_exit)

sound/pci/au88x0/au88x0_game.c

@@ -1,6 +1,4 @@
 /*
- * $Id: au88x0_game.c,v 1.9 2003/09/22 03:51:28 mjander Exp $
- *
  *  Manuel Jander.
  *
  *  Based on the work of:

sound/pci/azt3328.h

@@ -1,7 +1,8 @@
 #ifndef __SOUND_AZT3328_H
 #define __SOUND_AZT3328_H
 
-/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 */
+/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10
+ * "WRITE_ONLY"  == register does not indicate actual bit values */
 
 /*** main I/O area port indices ***/
 /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */
@@ -54,7 +55,10 @@
   #define SOUNDFORMAT_XTAL1		0x00
   #define SOUNDFORMAT_XTAL2		0x01
     /* all _SUSPECTED_ values are not used by Windows drivers, so we don't
-     * have any hard facts, only rough measurements */
+     * have any hard facts, only rough measurements.
+     * All we know is that the crystal used on the board has 24.576MHz,
+     * like many soundcards (which results in the frequencies below when
+     * using certain divider values selected by the values below) */
     #define SOUNDFORMAT_FREQ_SUSPECTED_4000	0x0c | SOUNDFORMAT_XTAL1
     #define SOUNDFORMAT_FREQ_SUSPECTED_4800	0x0a | SOUNDFORMAT_XTAL1
     #define SOUNDFORMAT_FREQ_5510		0x0c | SOUNDFORMAT_XTAL2
@@ -72,6 +76,26 @@
   #define SOUNDFORMAT_FLAG_16BIT	0x0010
   #define SOUNDFORMAT_FLAG_2CHANNELS	0x0020
 
+/* define frequency helpers, for maximum value safety */
+enum azf_freq_t {
+#define AZF_FREQ(rate) AZF_FREQ_##rate = rate
+  AZF_FREQ(4000),
+  AZF_FREQ(4800),
+  AZF_FREQ(5512),
+  AZF_FREQ(6620),
+  AZF_FREQ(8000),
+  AZF_FREQ(9600),
+  AZF_FREQ(11025),
+  AZF_FREQ(13240),
+  AZF_FREQ(16000),
+  AZF_FREQ(22050),
+  AZF_FREQ(32000),
+  AZF_FREQ(44100),
+  AZF_FREQ(48000),
+  AZF_FREQ(66200),
+#undef AZF_FREQ
+} AZF_FREQUENCIES;
+
 /** recording area (see also: playback bit flag definitions) **/
 #define IDX_IO_REC_FLAGS	0x20 /* ??, PU:0x0000 */
 #define IDX_IO_REC_IRQTYPE	0x22 /* ??, PU:0x0000 */
@@ -97,40 +121,171 @@
 
 /** DirectX timer, main interrupt area (FIXME: and something else?) **/ 
 #define IDX_IO_TIMER_VALUE	0x60 /* found this timer area by pure luck :-) */
-  #define TIMER_VALUE_MASK		0x000fffffUL /* timer countdown value; triggers IRQ when timer is finished */
-  #define TIMER_ENABLE_COUNTDOWN	0x01000000UL /* activate the timer countdown */
-  #define TIMER_ENABLE_IRQ		0x02000000UL /* trigger timer IRQ on zero transition */
-  #define TIMER_ACK_IRQ			0x04000000UL /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) had 0x0020 set upon IRQ handler */
+  /* timer countdown value; triggers IRQ when timer is finished */
+  #define TIMER_VALUE_MASK		0x000fffffUL
+  /* activate timer countdown */
+  #define TIMER_COUNTDOWN_ENABLE	0x01000000UL
+  /* trigger timer IRQ on zero transition */
+  #define TIMER_IRQ_ENABLE		0x02000000UL
+  /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?)
+   * had 0x0020 set upon IRQ handler */
+  #define TIMER_IRQ_ACK			0x04000000UL
 #define IDX_IO_IRQSTATUS        0x64
-  #define IRQ_PLAYBACK			0x0001
-  #define IRQ_RECORDING			0x0002
-  #define IRQ_MPU401			0x0010
-  #define IRQ_TIMER			0x0020 /* DirectX timer */
-  #define IRQ_UNKNOWN1			0x0040 /* probably unused, or possibly I2S port? or gameport IRQ? */
-  #define IRQ_UNKNOWN2			0x0080 /* probably unused, or possibly I2S port? or gameport IRQ? */
+  /* some IRQ bit in here might also be used to signal a power-management timer
+   * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing).
+   * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which
+   * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */
+
+  #define IRQ_PLAYBACK	0x0001
+  #define IRQ_RECORDING	0x0002
+  #define IRQ_UNKNOWN1	0x0004 /* most probably I2S port */
+  #define IRQ_GAMEPORT	0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */
+  #define IRQ_MPU401	0x0010
+  #define IRQ_TIMER	0x0020 /* DirectX timer */
+  #define IRQ_UNKNOWN2	0x0040 /* probably unused, or possibly I2S port? */
+  #define IRQ_UNKNOWN3	0x0080 /* probably unused, or possibly I2S port? */
 #define IDX_IO_66H		0x66    /* writing 0xffff returns 0x0000 */
-#define IDX_IO_SOME_VALUE	0x68	/* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */
-#define IDX_IO_6AH		0x6A	/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */
-  #define IO_6A_PAUSE_PLAYBACK		0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */
-#define IDX_IO_6CH		0x6C
-#define IDX_IO_6EH		0x6E	/* writing 0xffff returns 0x83fe */
-/* further I/O indices not saved/restored, so probably not used */
+  /* this is set to e.g. 0x3ff or 0x300, and writable;
+   * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */
+#define IDX_IO_SOME_VALUE	0x68
+  #define IO_68_RANDOM_TOGGLE1	0x0100	/* toggles randomly */
+  #define IO_68_RANDOM_TOGGLE2	0x0200	/* toggles randomly */
+  /* umm, nope, behaviour of these bits changes depending on what we wrote
+   * to 0x6b!!
+   * And they change upon playback/stop, too:
+   * Writing a value to 0x68 will display this exact value during playback,
+   * too but when stopped it can fall back to a rather different
+   * seemingly random value). Hmm, possibly this is a register which
+   * has a remote shadow which needs proper device supply which only exists
+   * in case playback is active? Or is this driver-induced?
+   */
+
+/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??);
+ * actually inhibits PCM playback!!! maybe power management??: */
+#define IDX_IO_6AH		0x6A /* WRITE_ONLY! */
+  /* bit 5: enabling this will activate permanent counting of bytes 2/3
+   * at gameport I/O (0xb402/3) (equal values each) and cause
+   * gameport legacy I/O at 0x0200 to be _DISABLED_!
+   * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode
+   * for Enhanced Digital Gameport (see 4D Wave DX card): */
+  #define IO_6A_SOMETHING1_GAMEPORT	0x0020
+  /* bit 8; sure, this _pauses_ playback (later resumes at same spot!),
+   * but what the heck is this really about??: */
+  #define IO_6A_PAUSE_PLAYBACK_BIT8	0x0100
+  /* bit 9; sure, this _pauses_ playback (later resumes at same spot!),
+   * but what the heck is this really about??: */
+  #define IO_6A_PAUSE_PLAYBACK_BIT9	0x0200
+	/* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback,
+	 * thus it suggests influence on PCM only!!
+	 * However OTOH there seems to be no bit anywhere around here
+	 * which is able to disable OPL3... */
+  /* bit 10: enabling this actually changes values at legacy gameport
+   * I/O address (0x200); is this enabling of the Digital Enhanced Game Port???
+   * Or maybe this simply switches off the NE558 circuit, since enabling this
+   * still lets us evaluate button states, but not axis states */
+  #define IO_6A_SOMETHING2_GAMEPORT      0x0400
+	/* writing 0x0300: causes quite some crackling during
+	 * PC activity such as switching windows (PCI traffic??
+	 * --> FIFO/timing settings???) */
+	/* writing 0x0100 plus/or 0x0200 inhibits playback */
+	/* since the Windows .INF file has Flag_Enable_JoyStick and
+	 * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason
+	 * that some other bit in this same register might be responsible
+	 * for SB DOS Emulation activation (note that the file did NOT define
+	 * a switch for OPL3!) */
+#define IDX_IO_6CH		0x6C	/* unknown; fully read-writable */
+#define IDX_IO_6EH		0x6E
+	/* writing 0xffff returns 0x83fe (or 0x03fe only).
+	 * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch
+	 * from 0000 to ffff. */
 
+/* further I/O indices not saved/restored and not readable after writing,
+ * so probably not used */
 
-/*** I/O 2 area port indices ***/
+
+/*** Gameport area port indices ***/
 /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ 
-#define AZF_IO_SIZE_IO2		0x08
-#define AZF_IO_SIZE_IO2_PM	0x06
+#define AZF_IO_SIZE_GAME		0x08
+#define AZF_IO_SIZE_GAME_PM	0x06
+
+enum {
+	AZF_GAME_LEGACY_IO_PORT = 0x200
+} AZF_GAME_CONFIGS;
+
+#define IDX_GAME_LEGACY_COMPATIBLE	0x00
+	/* in some operation mode, writing anything to this port
+	 * triggers an interrupt:
+	 * yup, that's in case IDX_GAME_01H has one of the
+	 * axis measurement bits enabled
+	 * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */
+
+#define IDX_GAME_AXES_CONFIG            0x01
+	/* NOTE: layout of this register awfully similar (read: "identical??")
+	 * to AD1815JS.pdf (p.29) */
+
+  /* enables axis 1 (X axis) measurement: */
+  #define GAME_AXES_ENABLE_1		0x01
+  /* enables axis 2 (Y axis) measurement: */
+  #define GAME_AXES_ENABLE_2		0x02
+  /* enables axis 3 (X axis) measurement: */
+  #define GAME_AXES_ENABLE_3		0x04
+  /* enables axis 4 (Y axis) measurement: */
+  #define GAME_AXES_ENABLE_4		0x08
+  /* selects the current axis to read the measured value of
+   * (at IDX_GAME_AXIS_VALUE):
+   * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */
+  #define GAME_AXES_READ_MASK		0x30
+  /* enable to have the latch continuously accept ADC values
+   * (and continuously cause interrupts in case interrupts are enabled);
+   * AD1815JS.pdf says it's ~16ms interval there: */
+  #define GAME_AXES_LATCH_ENABLE	0x40
+  /* joystick data (measured axes) ready for reading: */
+  #define GAME_AXES_SAMPLING_READY	0x80
+
+  /* NOTE: other card specs (SiS960 and others!) state that the
+   * game position latches should be frozen when reading and be freed
+   * (== reset?) after reading!!!
+   * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE),
+   *  but how to free the value? */
+  /* An internet search for "gameport latch ADC" should provide some insight
+   * into how to program such a gameport system. */
+
+  /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!?
+   * yup, in case 6AH 0x20 is not enabled
+   * (and 0x40 is sufficient, 0xf0 is not needed) */
+
+#define IDX_GAME_AXIS_VALUE	0x02
+	/* R: value of currently configured axis (word value!);
+	 * W: trigger axis measurement */
+
+#define IDX_GAME_HWCONFIG	0x04
+	/* note: bits 4 to 7 are never set (== 0) when reading!
+	 * --> reserved bits? */
+  /* enables IRQ notification upon axes measurement ready: */
+  #define GAME_HWCFG_IRQ_ENABLE			0x01
+  /* these bits choose a different frequency for the
+   *  internal ADC counter increment.
+   * hmm, seems to be a combo of bits:
+   * 00 --> standard frequency
+   * 10 --> 1/2
+   * 01 --> 1/20
+   * 11 --> 1/200: */
+  #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK	0x06
 
-#define IDX_IO2_LEGACY_ADDR	0x04
-  #define LEGACY_SOMETHING		0x01 /* OPL3?? */
-  #define LEGACY_JOY			0x08
+  /* enable gameport legacy I/O address (0x200)
+   * I was unable to locate any configurability for a different address: */
+  #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE	0x08
 
+/*** MPU401 ***/
 #define AZF_IO_SIZE_MPU		0x04
 #define AZF_IO_SIZE_MPU_PM	0x04
 
-#define AZF_IO_SIZE_SYNTH	0x08
-#define AZF_IO_SIZE_SYNTH_PM	0x06
+/*** OPL3 synth ***/
+#define AZF_IO_SIZE_OPL3	0x08
+#define AZF_IO_SIZE_OPL3_PM	0x06
+/* hmm, given that a standard OPL3 has 4 registers only,
+ * there might be some enhanced functionality lurking at the end
+ * (especially since register 0x04 has a "non-empty" value 0xfe) */
 
 /*** mixer I/O area port indices ***/
 /* (only 0x22 of 0x40 bytes saved/restored by Windows driver)

sound/pci/ca0106/ca0106_main.c

@@ -249,6 +249,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	   .name   = "MSI K8N Diamond MB [SB0438]",
 	   .gpio_type = 2,
 	   .i2c_adc = 1 } ,
+	 /* Another MSI K8N Diamond MB, which has apprently a different SSID */
+	 { .serial = 0x10091102,
+	   .name   = "MSI K8N Diamond MB",
+	   .gpio_type = 2,
+	   .i2c_adc = 1 } ,
 	 /* Shuttle XPC SD31P which has an onboard Creative Labs
 	  * Sound Blaster Live! 24-bit EAX
 	  * high-definition 7.1 audio processor".

sound/pci/emu10k1/emu10k1_main.c

@@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
+	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .adc_1361t = 1,  /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .revision = 0x04,

sound/pci/emu10k1/emumixer.c

@@ -1578,6 +1578,10 @@ static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol,
 		ucontrol->value.integer.value[0] = inl(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0;
 	else
 		ucontrol->value.integer.value[0] = inl(emu->port + HCFG) & HCFG_GPOUT0 ? 1 : 0;
+	if (emu->card_capabilities->invert_shared_spdif)
+		ucontrol->value.integer.value[0] =
+			!ucontrol->value.integer.value[0];
+		
 	return 0;
 }
 
@@ -1586,15 +1590,18 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol,
 {
 	unsigned long flags;
 	struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
-	unsigned int reg, val;
+	unsigned int reg, val, sw;
 	int change = 0;
 
+	sw = ucontrol->value.integer.value[0];
+	if (emu->card_capabilities->invert_shared_spdif)
+		sw = !sw;
 	spin_lock_irqsave(&emu->reg_lock, flags);
 	if ( emu->card_capabilities->i2c_adc) {
 		/* Do nothing for Audigy 2 ZS Notebook */
 	} else if (emu->audigy) {
 		reg = inl(emu->port + A_IOCFG);
-		val = ucontrol->value.integer.value[0] ? A_IOCFG_GPOUT0 : 0;
+		val = sw ? A_IOCFG_GPOUT0 : 0;
 		change = (reg & A_IOCFG_GPOUT0) != val;
 		if (change) {
 			reg &= ~A_IOCFG_GPOUT0;
@@ -1603,7 +1610,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol,
 		}
 	}
 	reg = inl(emu->port + HCFG);
-	val = ucontrol->value.integer.value[0] ? HCFG_GPOUT0 : 0;
+	val = sw ? HCFG_GPOUT0 : 0;
 	change |= (reg & HCFG_GPOUT0) != val;
 	if (change) {
 		reg &= ~HCFG_GPOUT0;

sound/pci/emu10k1/memory.c

@@ -437,43 +437,49 @@ static void get_single_page_range(struct snd_util_memhdr *hdr,
 	*last_page_ret = last_page;
 }
 
+/* release allocated pages */
+static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page,
+			       int last_page)
+{
+	int page;
+
+	for (page = first_page; page <= last_page; page++) {
+		free_page((unsigned long)emu->page_ptr_table[page]);
+		emu->page_addr_table[page] = 0;
+		emu->page_ptr_table[page] = NULL;
+	}
+}
+
 /*
  * allocate kernel pages
  */
 static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
 {
 	int page, first_page, last_page;
-	struct snd_dma_buffer dmab;
 
 	emu10k1_memblk_init(blk);
 	get_single_page_range(emu->memhdr, blk, &first_page, &last_page);
 	/* allocate kernel pages */
 	for (page = first_page; page <= last_page; page++) {
-		if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci),
-					PAGE_SIZE, &dmab) < 0)
-			goto __fail;
-		if (! is_valid_page(emu, dmab.addr)) {
-			snd_dma_free_pages(&dmab);
-			goto __fail;
+		/* first try to allocate from <4GB zone */
+		struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 |
+					    __GFP_NOWARN);
+		if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) {
+			if (p)
+				__free_page(p);
+			/* try to allocate from <16MB zone */
+			p = alloc_page(GFP_ATOMIC | GFP_DMA |
+				       __GFP_NORETRY | /* no OOM-killer */
+				       __GFP_NOWARN);
+		}
+		if (!p) {
+			__synth_free_pages(emu, first_page, page - 1);
+			return -ENOMEM;
 		}
-		emu->page_addr_table[page] = dmab.addr;
-		emu->page_ptr_table[page] = dmab.area;
+		emu->page_addr_table[page] = page_to_phys(p);
+		emu->page_ptr_table[page] = page_address(p);
 	}
 	return 0;
-
-__fail:
-	/* release allocated pages */
-	last_page = page - 1;
-	for (page = first_page; page <= last_page; page++) {
-		dmab.area = emu->page_ptr_table[page];
-		dmab.addr = emu->page_addr_table[page];
-		dmab.bytes = PAGE_SIZE;
-		snd_dma_free_pages(&dmab);
-		emu->page_addr_table[page] = 0;
-		emu->page_ptr_table[page] = NULL;
-	}
-
-	return -ENOMEM;
 }
 
 /*
@@ -481,23 +487,10 @@ __fail:
  */
 static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
 {
-	int page, first_page, last_page;
-	struct snd_dma_buffer dmab;
+	int first_page, last_page;
 
 	get_single_page_range(emu->memhdr, blk, &first_page, &last_page);
-	dmab.dev.type = SNDRV_DMA_TYPE_DEV;
-	dmab.dev.dev = snd_dma_pci_data(emu->pci);
-	for (page = first_page; page <= last_page; page++) {
-		if (emu->page_ptr_table[page] == NULL)
-			continue;
-		dmab.area = emu->page_ptr_table[page];
-		dmab.addr = emu->page_addr_table[page];
-		dmab.bytes = PAGE_SIZE;
-		snd_dma_free_pages(&dmab);
-		emu->page_addr_table[page] = 0;
-		emu->page_ptr_table[page] = NULL;
-	}
-
+	__synth_free_pages(emu, first_page, last_page);
 	return 0;
 }
 

sound/pci/hda/hda_codec.c

@@ -2335,7 +2335,7 @@ int snd_hda_check_board_config(struct hda_codec *codec,
 	if (!tbl)
 		return -1;
 	if (tbl->value >= 0 && tbl->value < num_configs) {
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 		char tmp[10];
 		const char *model = NULL;
 		if (models)

sound/pci/hda/hda_codec.h

@@ -78,7 +78,7 @@ enum {
 #define AC_VERB_GET_BEEP_CONTROL		0x0f0a
 #define AC_VERB_GET_EAPD_BTLENABLE		0x0f0c
 #define AC_VERB_GET_DIGI_CONVERT_1		0x0f0d
-#define AC_VERB_GET_DIGI_CONVERT_2		0x0f0e
+#define AC_VERB_GET_DIGI_CONVERT_2		0x0f0e /* unused */
 #define AC_VERB_GET_VOLUME_KNOB_CONTROL		0x0f0f
 /* f10-f1a: GPIO */
 #define AC_VERB_GET_GPIO_DATA			0x0f15

sound/pci/hda/hda_hwdep.c

@@ -88,7 +88,7 @@ static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file,
 
 static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
 {
-#ifndef CONFIG_SND_DEBUG_DETECT
+#ifndef CONFIG_SND_DEBUG_VERBOSE
 	if (!capable(CAP_SYS_RAWIO))
 		return -EACCES;
 #endif

sound/pci/hda/hda_intel.c

@@ -55,6 +55,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
 static char *model[SNDRV_CARDS];
 static int position_fix[SNDRV_CARDS];
+static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
 static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
 static int single_cmd;
 static int enable_msi;
@@ -69,7 +70,9 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "Fix DMA pointer "
-		 "(0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size).");
+		 "(0 = auto, 1 = none, 2 = POSBUF).");
+module_param_array(bdl_pos_adj, int, NULL, 0644);
+MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
 MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1).");
 module_param(single_cmd, bool, 0444);
@@ -197,6 +200,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define ATIHDMI_NUM_CAPTURE	0
 #define ATIHDMI_NUM_PLAYBACK	1
 
+/* TERA has 4 playback and 3 capture */
+#define TERA_NUM_CAPTURE	3
+#define TERA_NUM_PLAYBACK	4
+
 /* this number is statically defined for simplicity */
 #define MAX_AZX_DEV		16
 
@@ -259,9 +266,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 /* position fix mode */
 enum {
 	POS_FIX_AUTO,
-	POS_FIX_NONE,
+	POS_FIX_LPIB,
 	POS_FIX_POSBUF,
-	POS_FIX_FIFO,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -285,6 +291,7 @@ struct azx_dev {
 	u32 *posbuf;		/* position buffer pointer */
 
 	unsigned int bufsize;	/* size of the play buffer in bytes */
+	unsigned int period_bytes; /* size of the period in bytes */
 	unsigned int frags;	/* number for period in the play buffer */
 	unsigned int fifo_size;	/* FIFO size */
 
@@ -301,11 +308,11 @@ struct azx_dev {
 					 */
 	unsigned char stream_tag;	/* assigned stream */
 	unsigned char index;		/* stream index */
-	/* for sanity check of position buffer */
-	unsigned int period_intr;
 
 	unsigned int opened :1;
 	unsigned int running :1;
+	unsigned int irq_pending :1;
+	unsigned int irq_ignore :1;
 };
 
 /* CORB/RIRB */
@@ -323,6 +330,7 @@ struct azx_rb {
 struct azx {
 	struct snd_card *card;
 	struct pci_dev *pci;
+	int dev_index;
 
 	/* chip type specific */
 	int driver_type;
@@ -366,9 +374,13 @@ struct azx {
 	unsigned int single_cmd :1;
 	unsigned int polling_mode :1;
 	unsigned int msi :1;
+	unsigned int irq_pending_warned :1;
 
 	/* for debugging */
 	unsigned int last_cmd;	/* last issued command (to sync) */
+
+	/* for pending irqs */
+	struct work_struct irq_pending_work;
 };
 
 /* driver types */
@@ -381,6 +393,7 @@ enum {
 	AZX_DRIVER_SIS,
 	AZX_DRIVER_ULI,
 	AZX_DRIVER_NVIDIA,
+	AZX_DRIVER_TERA,
 };
 
 static char *driver_short_names[] __devinitdata = {
@@ -392,6 +405,7 @@ static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_SIS] = "HDA SIS966",
 	[AZX_DRIVER_ULI] = "HDA ULI M5461",
 	[AZX_DRIVER_NVIDIA] = "HDA NVidia",
+	[AZX_DRIVER_TERA] = "HDA Teradici", 
 };
 
 /*
@@ -426,11 +440,6 @@ static char *driver_short_names[] __devinitdata = {
 /* for pcm support */
 #define get_azx_dev(substream) (substream->runtime->private_data)
 
-/* Get the upper 32bit of the given dma_addr_t
- * Compiler should optimize and eliminate the code if dma_addr_t is 32bit
- */
-#define upper_32bit(addr) (sizeof(addr) > 4 ? (u32)((addr) >> 32) : (u32)0)
-
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
 
 /*
@@ -461,7 +470,7 @@ static void azx_init_cmd_io(struct azx *chip)
 	chip->corb.addr = chip->rb.addr;
 	chip->corb.buf = (u32 *)chip->rb.area;
 	azx_writel(chip, CORBLBASE, (u32)chip->corb.addr);
-	azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr));
+	azx_writel(chip, CORBUBASE, upper_32_bits(chip->corb.addr));
 
 	/* set the corb size to 256 entries (ULI requires explicitly) */
 	azx_writeb(chip, CORBSIZE, 0x02);
@@ -476,7 +485,7 @@ static void azx_init_cmd_io(struct azx *chip)
 	chip->rirb.addr = chip->rb.addr + 2048;
 	chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
 	azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr);
-	azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr));
+	azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr));
 
 	/* set the rirb size to 256 entries (ULI requires explicitly) */
 	azx_writeb(chip, RIRBSIZE, 0x02);
@@ -847,7 +856,7 @@ static void azx_init_chip(struct azx *chip)
 
 	/* program the position buffer */
 	azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
-	azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
+	azx_writel(chip, DPUBASE, upper_32_bits(chip->posbuf.addr));
 
 	chip->initialized = 1;
 }
@@ -908,6 +917,8 @@ static void azx_init_pci(struct azx *chip)
 }
 
 
+static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev);
+
 /*
  * interrupt handler
  */
@@ -930,11 +941,23 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 		azx_dev = &chip->azx_dev[i];
 		if (status & azx_dev->sd_int_sta_mask) {
 			azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
-			if (azx_dev->substream && azx_dev->running) {
-				azx_dev->period_intr++;
+			if (!azx_dev->substream || !azx_dev->running)
+				continue;
+			/* ignore the first dummy IRQ (due to pos_adj) */
+			if (azx_dev->irq_ignore) {
+				azx_dev->irq_ignore = 0;
+				continue;
+			}
+			/* check whether this IRQ is really acceptable */
+			if (azx_position_ok(chip, azx_dev)) {
+				azx_dev->irq_pending = 0;
 				spin_unlock(&chip->reg_lock);
 				snd_pcm_period_elapsed(azx_dev->substream);
 				spin_lock(&chip->reg_lock);
+			} else {
+				/* bogus IRQ, process it later */
+				azx_dev->irq_pending = 1;
+				schedule_work(&chip->irq_pending_work);
 			}
 		}
 	}
@@ -958,60 +981,108 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 }
 
 
+/*
+ * set up a BDL entry
+ */
+static int setup_bdle(struct snd_pcm_substream *substream,
+		      struct azx_dev *azx_dev, u32 **bdlp,
+		      int ofs, int size, int with_ioc)
+{
+	struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream);
+	u32 *bdl = *bdlp;
+
+	while (size > 0) {
+		dma_addr_t addr;
+		int chunk;
+
+		if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES)
+			return -EINVAL;
+
+		addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs);
+		/* program the address field of the BDL entry */
+		bdl[0] = cpu_to_le32((u32)addr);
+		bdl[1] = cpu_to_le32(upper_32_bits(addr));
+		/* program the size field of the BDL entry */
+		chunk = PAGE_SIZE - (ofs % PAGE_SIZE);
+		if (size < chunk)
+			chunk = size;
+		bdl[2] = cpu_to_le32(chunk);
+		/* program the IOC to enable interrupt
+		 * only when the whole fragment is processed
+		 */
+		size -= chunk;
+		bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01);
+		bdl += 4;
+		azx_dev->frags++;
+		ofs += chunk;
+	}
+	*bdlp = bdl;
+	return ofs;
+}
+
 /*
  * set up BDL entries
  */
-static int azx_setup_periods(struct snd_pcm_substream *substream,
+static int azx_setup_periods(struct azx *chip,
+			     struct snd_pcm_substream *substream,
 			     struct azx_dev *azx_dev)
 {
-	struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream);
 	u32 *bdl;
 	int i, ofs, periods, period_bytes;
+	int pos_adj;
 
 	/* reset BDL address */
 	azx_sd_writel(azx_dev, SD_BDLPL, 0);
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 
 	period_bytes = snd_pcm_lib_period_bytes(substream);
+	azx_dev->period_bytes = period_bytes;
 	periods = azx_dev->bufsize / period_bytes;
 
 	/* program the initial BDL entries */
 	bdl = (u32 *)azx_dev->bdl.area;
 	ofs = 0;
 	azx_dev->frags = 0;
-	for (i = 0; i < periods; i++) {
-		int size, rest;
-		if (i >= AZX_MAX_BDL_ENTRIES) {
-			snd_printk(KERN_ERR "Too many BDL entries: "
-				   "buffer=%d, period=%d\n",
-				   azx_dev->bufsize, period_bytes);
-			/* reset */
-			azx_sd_writel(azx_dev, SD_BDLPL, 0);
-			azx_sd_writel(azx_dev, SD_BDLPU, 0);
-			return -EINVAL;
+	azx_dev->irq_ignore = 0;
+	pos_adj = bdl_pos_adj[chip->dev_index];
+	if (pos_adj > 0) {
+		struct snd_pcm_runtime *runtime = substream->runtime;
+		pos_adj = (pos_adj * runtime->rate + 47999) / 48000;
+		if (!pos_adj)
+			pos_adj = 1;
+		pos_adj = frames_to_bytes(runtime, pos_adj);
+		if (pos_adj >= period_bytes) {
+			snd_printk(KERN_WARNING "Too big adjustment %d\n",
+				   bdl_pos_adj[chip->dev_index]);
+			pos_adj = 0;
+		} else {
+			ofs = setup_bdle(substream, azx_dev,
+					 &bdl, ofs, pos_adj, 1);
+			if (ofs < 0)
+				goto error;
+			azx_dev->irq_ignore = 1;
 		}
-		rest = period_bytes;
-		do {
-			dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs);
-			/* program the address field of the BDL entry */
-			bdl[0] = cpu_to_le32((u32)addr);
-			bdl[1] = cpu_to_le32(upper_32bit(addr));
-			/* program the size field of the BDL entry */
-			size = PAGE_SIZE - (ofs % PAGE_SIZE);
-			if (rest < size)
-				size = rest;
-			bdl[2] = cpu_to_le32(size);
-			/* program the IOC to enable interrupt
-			 * only when the whole fragment is processed
-			 */
-			rest -= size;
-			bdl[3] = rest ? 0 : cpu_to_le32(0x01);
-			bdl += 4;
-			azx_dev->frags++;
-			ofs += size;
-		} while (rest > 0);
+	} else
+		pos_adj = 0;
+	for (i = 0; i < periods; i++) {
+		if (i == periods - 1 && pos_adj)
+			ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+					 period_bytes - pos_adj, 0);
+		else
+			ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+					 period_bytes, 1);
+		if (ofs < 0)
+			goto error;
 	}
 	return 0;
+
+ error:
+	snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n",
+		   azx_dev->bufsize, period_bytes);
+	/* reset */
+	azx_sd_writel(azx_dev, SD_BDLPL, 0);
+	azx_sd_writel(azx_dev, SD_BDLPU, 0);
+	return -EINVAL;
 }
 
 /*
@@ -1062,7 +1133,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 	/* lower BDL address */
 	azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr);
 	/* upper BDL address */
-	azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr));
+	azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr));
 
 	/* enable the position buffer */
 	if (chip->position_fix == POS_FIX_POSBUF ||
@@ -1085,7 +1156,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
  */
 
 static unsigned int azx_max_codecs[] __devinitdata = {
-	[AZX_DRIVER_ICH] = 3,
+	[AZX_DRIVER_ICH] = 4,		/* Some ICH9 boards use SD3 */
 	[AZX_DRIVER_SCH] = 3,
 	[AZX_DRIVER_ATI] = 4,
 	[AZX_DRIVER_ATIHDMI] = 4,
@@ -1093,6 +1164,7 @@ static unsigned int azx_max_codecs[] __devinitdata = {
 	[AZX_DRIVER_SIS] = 3,		/* FIXME: correct? */
 	[AZX_DRIVER_ULI] = 3,		/* FIXME: correct? */
 	[AZX_DRIVER_NVIDIA] = 3,	/* FIXME: correct? */
+	[AZX_DRIVER_TERA] = 1,
 };
 
 static int __devinit azx_codec_create(struct azx *chip, const char *model,
@@ -1316,7 +1388,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 
 	snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
 		    azx_dev->bufsize, azx_dev->format_val);
-	if (azx_setup_periods(substream, azx_dev) < 0)
+	if (azx_setup_periods(chip, substream, azx_dev) < 0)
 		return -EINVAL;
 	azx_setup_controller(chip, azx_dev);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1421,35 +1493,113 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	return 0;
 }
 
-static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
+static unsigned int azx_get_position(struct azx *chip,
+				     struct azx_dev *azx_dev)
 {
-	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
-	struct azx *chip = apcm->chip;
-	struct azx_dev *azx_dev = get_azx_dev(substream);
 	unsigned int pos;
 
 	if (chip->position_fix == POS_FIX_POSBUF ||
 	    chip->position_fix == POS_FIX_AUTO) {
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
-		if (chip->position_fix == POS_FIX_AUTO &&
-		    azx_dev->period_intr == 1 && !pos) {
-			printk(KERN_WARNING
-			       "hda-intel: Invalid position buffer, "
-			       "using LPIB read method instead.\n");
-			chip->position_fix = POS_FIX_NONE;
-			goto read_lpib;
-		}
 	} else {
-	read_lpib:
 		/* read LPIB */
 		pos = azx_sd_readl(azx_dev, SD_LPIB);
-		if (chip->position_fix == POS_FIX_FIFO)
-			pos += azx_dev->fifo_size;
 	}
 	if (pos >= azx_dev->bufsize)
 		pos = 0;
-	return bytes_to_frames(substream->runtime, pos);
+	return pos;
+}
+
+static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(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));
+}
+
+/*
+ * Check whether the current DMA position is acceptable for updating
+ * periods.  Returns non-zero if it's OK.
+ *
+ * Many HD-audio controllers appear pretty inaccurate about
+ * the update-IRQ timing.  The IRQ is issued before actually the
+ * data is processed.  So, we need to process it afterwords in a
+ * workqueue.
+ */
+static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
+{
+	unsigned int pos;
+
+	pos = azx_get_position(chip, azx_dev);
+	if (chip->position_fix == POS_FIX_AUTO) {
+		if (!pos) {
+			printk(KERN_WARNING
+			       "hda-intel: Invalid position buffer, "
+			       "using LPIB read method instead.\n");
+			chip->position_fix = POS_FIX_LPIB;
+			pos = azx_get_position(chip, azx_dev);
+		} else
+			chip->position_fix = POS_FIX_POSBUF;
+	}
+
+	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
+		return 0; /* NG - it's below the period boundary */
+	return 1; /* OK, it's fine */
+}
+
+/*
+ * The work for pending PCM period updates.
+ */
+static void azx_irq_pending_work(struct work_struct *work)
+{
+	struct azx *chip = container_of(work, struct azx, irq_pending_work);
+	int i, pending;
+
+	if (!chip->irq_pending_warned) {
+		printk(KERN_WARNING
+		       "hda-intel: IRQ timing workaround is activated "
+		       "for card #%d. Suggest a bigger bdl_pos_adj.\n",
+		       chip->card->number);
+		chip->irq_pending_warned = 1;
+	}
+
+	for (;;) {
+		pending = 0;
+		spin_lock_irq(&chip->reg_lock);
+		for (i = 0; i < chip->num_streams; i++) {
+			struct azx_dev *azx_dev = &chip->azx_dev[i];
+			if (!azx_dev->irq_pending ||
+			    !azx_dev->substream ||
+			    !azx_dev->running)
+				continue;
+			if (azx_position_ok(chip, azx_dev)) {
+				azx_dev->irq_pending = 0;
+				spin_unlock(&chip->reg_lock);
+				snd_pcm_period_elapsed(azx_dev->substream);
+				spin_lock(&chip->reg_lock);
+			} else
+				pending++;
+		}
+		spin_unlock_irq(&chip->reg_lock);
+		if (!pending)
+			return;
+		cond_resched();
+	}
+}
+
+/* clear irq_pending flags and assure no on-going workq */
+static void azx_clear_irq_pending(struct azx *chip)
+{
+	int i;
+
+	spin_lock_irq(&chip->reg_lock);
+	for (i = 0; i < chip->num_streams; i++)
+		chip->azx_dev[i].irq_pending = 0;
+	spin_unlock_irq(&chip->reg_lock);
+	flush_scheduled_work();
 }
 
 static struct snd_pcm_ops azx_pcm_ops = {
@@ -1676,6 +1826,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 	int i;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	azx_clear_irq_pending(chip);
 	for (i = 0; i < AZX_MAX_PCMS; i++)
 		snd_pcm_suspend_all(chip->pcm[i]);
 	if (chip->initialized)
@@ -1732,6 +1883,7 @@ static int azx_free(struct azx *chip)
 	int i;
 
 	if (chip->initialized) {
+		azx_clear_irq_pending(chip);
 		for (i = 0; i < chip->num_streams; i++)
 			azx_stream_stop(chip, &chip->azx_dev[i]);
 		azx_stop_chip(chip);
@@ -1770,9 +1922,9 @@ static int azx_dev_free(struct snd_device *device)
  * white/black-listing for position_fix
  */
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
-	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE),
-	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE),
-	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE),
+	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	{}
 };
 
@@ -1857,12 +2009,25 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	chip->irq = -1;
 	chip->driver_type = driver_type;
 	chip->msi = enable_msi;
+	chip->dev_index = dev;
+	INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
 
 	chip->position_fix = check_position_fix(chip, position_fix[dev]);
 	check_probe_mask(chip, dev);
 
 	chip->single_cmd = single_cmd;
 
+	if (bdl_pos_adj[dev] < 0) {
+		switch (chip->driver_type) {
+		case AZX_DRIVER_ICH:
+			bdl_pos_adj[dev] = 1;
+			break;
+		default:
+			bdl_pos_adj[dev] = 32;
+			break;
+		}
+	}
+
 #if BITS_PER_LONG != 64
 	/* Fix up base address on ULI M5461 */
 	if (chip->driver_type == AZX_DRIVER_ULI) {
@@ -2089,6 +2254,7 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH },
+	{ PCI_DEVICE(0x8086, 0x2911), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH },
@@ -2141,6 +2307,8 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
+	/* Teradici */
+	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);

sound/pci/hda/hda_proc.c

@@ -366,8 +366,6 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 {
 	unsigned int digi1 = snd_hda_codec_read(codec, nid, 0,
 						AC_VERB_GET_DIGI_CONVERT_1, 0);
-	unsigned int digi2 = snd_hda_codec_read(codec, nid, 0,
-						AC_VERB_GET_DIGI_CONVERT_2, 0);
 	snd_iprintf(buffer, "  Digital:");
 	if (digi1 & AC_DIG1_ENABLE)
 		snd_iprintf(buffer, " Enabled");
@@ -386,7 +384,8 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 	if (digi1 & AC_DIG1_LEVEL)
 		snd_iprintf(buffer, " GenLevel");
 	snd_iprintf(buffer, "\n");
-	snd_iprintf(buffer, "  Digital category: 0x%x\n", digi2 & AC_DIG2_CC);
+	snd_iprintf(buffer, "  Digital category: 0x%x\n",
+		    (digi1 >> 8) & AC_DIG2_CC);
 }
 
 static const char *get_pwr_state(u32 state)

sound/pci/hda/patch_analog.c

@@ -23,7 +23,6 @@
 #include <linux/delay.h>
 #include <linux/slab.h>
 #include <linux/pci.h>
-#include <linux/mutex.h>
 
 #include <sound/core.h>
 #include "hda_codec.h"
@@ -64,7 +63,6 @@ struct ad198x_spec {
 	/* PCM information */
 	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */
 
-	struct mutex amp_mutex;	/* PCM volume/mute control mutex */
 	unsigned int spdif_route;
 
 	/* dynamic controls, init_verbs and input_mux */
@@ -1618,6 +1616,7 @@ static const char *ad1981_models[AD1981_MODELS] = {
 
 static struct snd_pci_quirk ad1981_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
+	SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
 	/* All HP models */
 	SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
@@ -2623,7 +2622,7 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 {
 	struct ad198x_spec *spec = codec->spec;
 	hda_nid_t nid;
-	int idx, err;
+	int i, idx, err;
 	char name[32];
 
 	if (! pin)
@@ -2631,16 +2630,26 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 
 	idx = ad1988_pin_idx(pin);
 	nid = ad1988_idx_to_dac(codec, idx);
-	/* specify the DAC as the extra output */
-	if (! spec->multiout.hp_nid)
-		spec->multiout.hp_nid = nid;
-	else
-		spec->multiout.extra_out_nid[0] = nid;
-	/* control HP volume/switch on the output mixer amp */
-	sprintf(name, "%s Playback Volume", pfx);
-	if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name,
-			       HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
-		return err;
+	/* check whether the corresponding DAC was already taken */
+	for (i = 0; i < spec->autocfg.line_outs; i++) {
+		hda_nid_t pin = spec->autocfg.line_out_pins[i];
+		hda_nid_t dac = ad1988_idx_to_dac(codec, ad1988_pin_idx(pin));
+		if (dac == nid)
+			break;
+	}
+	if (i >= spec->autocfg.line_outs) {
+		/* specify the DAC as the extra output */
+		if (!spec->multiout.hp_nid)
+			spec->multiout.hp_nid = nid;
+		else
+			spec->multiout.extra_out_nid[0] = nid;
+		/* control HP volume/switch on the output mixer amp */
+		sprintf(name, "%s Playback Volume", pfx);
+		err = add_control(spec, AD_CTL_WIDGET_VOL, name,
+				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+		if (err < 0)
+			return err;
+	}
 	nid = ad1988_mixer_nids[idx];
 	sprintf(name, "%s Playback Switch", pfx);
 	if ((err = add_control(spec, AD_CTL_BIND_MUTE, name,
@@ -3177,7 +3186,6 @@ static int patch_ad1884(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
-	mutex_init(&spec->amp_mutex);
 	codec->spec = spec;
 
 	spec->multiout.max_channels = 2;
@@ -3847,7 +3855,6 @@ static int patch_ad1884a(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
-	mutex_init(&spec->amp_mutex);
 	codec->spec = spec;
 
 	spec->multiout.max_channels = 2;
@@ -4152,7 +4159,6 @@ static int patch_ad1882(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
-	mutex_init(&spec->amp_mutex);
 	codec->spec = spec;
 
 	spec->multiout.max_channels = 6;

sound/pci/hda/patch_conexant.c

@@ -82,7 +82,6 @@ struct conexant_spec {
 	/* PCM information */
 	struct hda_pcm pcm_rec[2];	/* used in build_pcms() */
 
-	struct mutex amp_mutex;	/* PCM volume/mute control mutex */
 	unsigned int spdif_route;
 
 	/* dynamic controls, init_verbs and input_mux */
@@ -687,7 +686,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
 
 static struct hda_verb cxt5045_init_verbs[] = {
 	/* Line in, Mic */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
 	/* HP, Amp  */
 	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -907,10 +906,12 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE),
 	SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
 	SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE),
+	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE),
 	SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
 	SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
 	SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE),
-	SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP_HPSENSE),
+	SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505",
+		      CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE),
@@ -928,7 +929,6 @@ static int patch_cxt5045(struct hda_codec *codec)
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
-	mutex_init(&spec->amp_mutex);
 	codec->spec = spec;
 
 	spec->multiout.max_channels = 2;
@@ -963,6 +963,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 		codec->patch_ops.init = cxt5045_init;
 		break;
 	case CXT5045_LAPTOP_MICSENSE:
+		codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
 		spec->input_mux = &cxt5045_capture_source;
 		spec->num_init_verbs = 2;
 		spec->init_verbs[1] = cxt5045_mic_sense_init_verbs;
@@ -1007,15 +1008,19 @@ static int patch_cxt5045(struct hda_codec *codec)
 #endif	
 	}
 
-	/*
-	 * Fix max PCM level to 0 dB
-	 * (originall it has 0x2b steps with 0dB offset 0x14)
-	 */
-	snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
-				  (0x14 << AC_AMPCAP_OFFSET_SHIFT) |
-				  (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-				  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-				  (1 << AC_AMPCAP_MUTE_SHIFT));
+	switch (codec->subsystem_id >> 16) {
+	case 0x103c:
+		/* HP laptop has a really bad sound over 0dB on NID 0x17.
+		 * Fix max PCM level to 0 dB
+		 * (originall it has 0x2b steps with 0dB offset 0x14)
+		 */
+		snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
+					  (0x14 << AC_AMPCAP_OFFSET_SHIFT) |
+					  (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+					  (1 << AC_AMPCAP_MUTE_SHIFT));
+		break;
+	}
 
 	return 0;
 }
@@ -1477,7 +1482,6 @@ static int patch_cxt5047(struct hda_codec *codec)
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
-	mutex_init(&spec->amp_mutex);
 	codec->spec = spec;
 
 	spec->multiout.max_channels = 2;
@@ -1736,7 +1740,6 @@ static int patch_cxt5051(struct hda_codec *codec)
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
-	mutex_init(&spec->amp_mutex);
 	codec->spec = spec;
 
 	codec->patch_ops = conexant_patch_ops;

sound/pci/hda/patch_realtek.c

@@ -163,6 +163,10 @@ enum {
 	ALC662_LENOVO_101E,
 	ALC662_ASUS_EEEPC_P701,
 	ALC662_ASUS_EEEPC_EP20,
+	ALC663_ASUS_M51VA,
+	ALC663_ASUS_G71V,
+	ALC663_ASUS_H13,
+	ALC663_ASUS_G50V,
 	ALC662_AUTO,
 	ALC662_MODEL_LAST,
 };
@@ -205,6 +209,7 @@ enum {
 	ALC883_MITAC,
 	ALC883_CLEVO_M720,
 	ALC883_FUJITSU_PI2515,
+	ALC883_3ST_6ch_INTEL,
 	ALC883_AUTO,
 	ALC883_MODEL_LAST,
 };
@@ -280,6 +285,10 @@ struct alc_spec {
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	struct hda_loopback_check loopback;
 #endif
+
+	/* for PLL fix */
+	hda_nid_t pll_nid;
+	unsigned int pll_coef_idx, pll_coef_bit;
 };
 
 /*
@@ -747,6 +756,38 @@ static struct hda_verb alc_gpio3_init_verbs[] = {
 	{ }
 };
 
+/*
+ * Fix hardware PLL issue
+ * On some codecs, the analog PLL gating control must be off while
+ * the default value is 1.
+ */
+static void alc_fix_pll(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int val;
+
+	if (!spec->pll_nid)
+		return;
+	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
+			    spec->pll_coef_idx);
+	val = snd_hda_codec_read(codec, spec->pll_nid, 0,
+				 AC_VERB_GET_PROC_COEF, 0);
+	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
+			    spec->pll_coef_idx);
+	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
+			    val & ~(1 << spec->pll_coef_bit));
+}
+
+static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid,
+			     unsigned int coef_idx, unsigned int coef_bit)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->pll_nid = nid;
+	spec->pll_coef_idx = coef_idx;
+	spec->pll_coef_bit = coef_bit;
+	alc_fix_pll(codec);
+}
+
 static void alc_sku_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -776,6 +817,24 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 	alc_sku_automute(codec);
 }
 
+/* additional initialization for ALC888 variants */
+static void alc888_coef_init(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0);
+	tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
+	if ((tmp & 0xf0) == 2)
+		/* alc888S-VC */
+		snd_hda_codec_read(codec, 0x20, 0,
+				   AC_VERB_SET_PROC_COEF, 0x830);
+	 else
+		 /* alc888-VB */
+		 snd_hda_codec_read(codec, 0x20, 0,
+				    AC_VERB_SET_PROC_COEF, 0x3030);
+}
+
 /* 32-bit subsystem ID for BIOS loading in HD Audio codec.
  *	31 ~ 16 :	Manufacture ID
  *	15 ~ 8	:	SKU ID
@@ -851,8 +910,10 @@ do_sku:
 		case 0x10ec0267:
 		case 0x10ec0268:
 		case 0x10ec0269:
+		case 0x10ec0660:
+		case 0x10ec0662:
+		case 0x10ec0663:
 		case 0x10ec0862:
-		case 0x10ec0662:	
 		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x14, 0,
 					    AC_VERB_SET_EAPD_BTLENABLE, 2);
@@ -877,7 +938,6 @@ do_sku:
 		case 0x10ec0882:
 		case 0x10ec0883:
 		case 0x10ec0885:
-		case 0x10ec0888:
 		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_COEF_INDEX, 7);
@@ -889,6 +949,9 @@ do_sku:
 					    AC_VERB_SET_PROC_COEF,
 					    tmp | 0x2010);
 			break;
+		case 0x10ec0888:
+			alc888_coef_init(codec);
+			break;
 		case 0x10ec0267:
 		case 0x10ec0268:
 			snd_hda_codec_write(codec, 0x20, 0,
@@ -2373,6 +2436,8 @@ static int alc_init(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	unsigned int i;
 
+	alc_fix_pll(codec);
+
 	for (i = 0; i < spec->num_init_verbs; i++)
 		snd_hda_sequence_write(codec, spec->init_verbs[i]);
 
@@ -3009,6 +3074,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
+	SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
 	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
 	SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
 	SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
@@ -5101,7 +5167,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP),
+	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
@@ -6127,6 +6193,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
+	SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
 	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
@@ -6353,7 +6420,9 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 			continue;
 		vref = PIN_IN;
 		if (1 /*i <= AUTO_PIN_FRONT_MIC*/) {
-			if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) &
+			unsigned int pincap;
+			pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+			if ((pincap >> AC_PINCAP_VREF_SHIFT) &
 			    AC_PINCAP_VREF_80)
 				vref = PIN_VREF80;
 		}
@@ -6450,8 +6519,9 @@ static int patch_alc882(struct hda_codec *codec)
 		case 0x106b1000: /* iMac 24 */
 			board_config = ALC885_IMAC24;
 			break;
-		case 0x106b00a1: /* Macbook */
+		case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
 		case 0x106b2c00: /* Macbook Pro rev3 */
+		case 0x106b3600: /* Macbook 3.1 */
 			board_config = ALC885_MBP3;
 			break;
 		default:
@@ -6485,14 +6555,20 @@ static int patch_alc882(struct hda_codec *codec)
 	if (board_config != ALC882_AUTO)
 		setup_preset(spec, &alc882_presets[board_config]);
 
-	spec->stream_name_analog = "ALC882 Analog";
+	if (codec->vendor_id == 0x10ec0885) {
+		spec->stream_name_analog = "ALC885 Analog";
+		spec->stream_name_digital = "ALC885 Digital";
+	} else {
+		spec->stream_name_analog = "ALC882 Analog";
+		spec->stream_name_digital = "ALC882 Digital";
+	}
+
 	spec->stream_analog_playback = &alc882_pcm_analog_playback;
 	spec->stream_analog_capture = &alc882_pcm_analog_capture;
 	/* FIXME: setup DAC5 */
 	/*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
 	spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
 
-	spec->stream_name_digital = "ALC882 Digital";
 	spec->stream_digital_playback = &alc882_pcm_digital_playback;
 	spec->stream_digital_capture = &alc882_pcm_digital_capture;
 
@@ -6569,6 +6645,16 @@ static struct hda_input_mux alc883_capture_source = {
 	},
 };
 
+static struct hda_input_mux alc883_3stack_6ch_intel = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Front Mic", 0x0 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
 static struct hda_input_mux alc883_lenovo_101e_capture_source = {
 	.num_items = 2,
 	.items = {
@@ -6649,6 +6735,48 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = {
 	{ 6, alc883_3ST_ch6_init },
 };
 
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_intel_init[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x19, 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 */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_intel_init[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 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 */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_intel_init[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x19, 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 struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
+	{ 2, alc883_3ST_ch2_intel_init },
+	{ 4, alc883_3ST_ch4_intel_init },
+	{ 6, alc883_3ST_ch6_intel_init },
+};
+
 /*
  * 6ch mode
  */
@@ -6881,15 +7009,54 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+			      HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 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_MUTE("Headphone Playback Switch", 0x15, 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, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 2,
+		.info = alc883_mux_enum_info,
+		.get = alc883_mux_enum_get,
+		.put = alc883_mux_enum_put,
+	},
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 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_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),
@@ -7729,6 +7896,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 	[ALC883_MITAC]		= "mitac",
 	[ALC883_CLEVO_M720]	= "clevo-m720",
 	[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
+	[ALC883_3ST_6ch_INTEL]	= "3stack-6ch-intel",
 	[ALC883_AUTO]		= "auto",
 };
 
@@ -7786,6 +7954,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+	SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
 	{}
 };
@@ -7824,6 +7994,18 @@ static struct alc_config_preset alc883_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
 	},
+	[ALC883_3ST_6ch_INTEL] = {
+		.mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
+		.channel_mode = alc883_3ST_6ch_intel_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_3stack_6ch_intel,
+	},
 	[ALC883_6ST_DIG] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs },
@@ -8145,6 +8327,8 @@ static int patch_alc883(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+
 	board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST,
 						  alc883_models,
 						  alc883_cfg_tbl);
@@ -8171,12 +8355,25 @@ static int patch_alc883(struct hda_codec *codec)
 	if (board_config != ALC883_AUTO)
 		setup_preset(spec, &alc883_presets[board_config]);
 
-	spec->stream_name_analog = "ALC883 Analog";
+	switch (codec->vendor_id) {
+	case 0x10ec0888:
+		spec->stream_name_analog = "ALC888 Analog";
+		spec->stream_name_digital = "ALC888 Digital";
+		break;
+	case 0x10ec0889:
+		spec->stream_name_analog = "ALC889 Analog";
+		spec->stream_name_digital = "ALC889 Digital";
+		break;
+	default:
+		spec->stream_name_analog = "ALC883 Analog";
+		spec->stream_name_digital = "ALC883 Digital";
+		break;
+	}
+
 	spec->stream_analog_playback = &alc883_pcm_analog_playback;
 	spec->stream_analog_capture = &alc883_pcm_analog_capture;
 	spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture;
 
-	spec->stream_name_digital = "ALC883 Digital";
 	spec->stream_digital_playback = &alc883_pcm_digital_playback;
 	spec->stream_digital_capture = &alc883_pcm_digital_capture;
 
@@ -8189,6 +8386,9 @@ static int patch_alc883(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC883_AUTO)
 		spec->init_hook = alc883_auto_init;
+	else if (codec->vendor_id == 0x10ec0888)
+		spec->init_hook = alc888_coef_init;
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc883_loopbacks;
@@ -9522,6 +9722,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
+		      ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
@@ -9729,6 +9931,8 @@ static int patch_alc262(struct hda_codec *codec)
 	}
 #endif
 
+	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+
 	board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST,
 						  alc262_models,
 						  alc262_cfg_tbl);
@@ -10674,12 +10878,18 @@ static int patch_alc268(struct hda_codec *codec)
 	if (board_config != ALC268_AUTO)
 		setup_preset(spec, &alc268_presets[board_config]);
 
-	spec->stream_name_analog = "ALC268 Analog";
+	if (codec->vendor_id == 0x10ec0267) {
+		spec->stream_name_analog = "ALC267 Analog";
+		spec->stream_name_digital = "ALC267 Digital";
+	} else {
+		spec->stream_name_analog = "ALC268 Analog";
+		spec->stream_name_digital = "ALC268 Digital";
+	}
+
 	spec->stream_analog_playback = &alc268_pcm_analog_playback;
 	spec->stream_analog_capture = &alc268_pcm_analog_capture;
 	spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture;
 
-	spec->stream_name_digital = "ALC268 Digital";
 	spec->stream_digital_playback = &alc268_pcm_digital_playback;
 
 	if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
@@ -11033,6 +11243,8 @@ static int patch_alc269(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	alc_fix_pll_init(codec, 0x20, 0x04, 15);
+
 	board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST,
 						  alc269_models,
 						  alc269_cfg_tbl);
@@ -12631,6 +12843,12 @@ static struct hda_verb alc861vd_eapd_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc660vd_eapd_verbs[] = {
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
 static 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)},
@@ -12786,6 +13004,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
 	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
+	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
 	{}
 };
@@ -13168,11 +13387,19 @@ static int patch_alc861vd(struct hda_codec *codec)
 	if (board_config != ALC861VD_AUTO)
 		setup_preset(spec, &alc861vd_presets[board_config]);
 
-	spec->stream_name_analog = "ALC861VD Analog";
+	if (codec->vendor_id == 0x10ec0660) {
+		spec->stream_name_analog = "ALC660-VD Analog";
+		spec->stream_name_digital = "ALC660-VD Digital";
+		/* always turn on EAPD */
+		spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs;
+	} else {
+		spec->stream_name_analog = "ALC861VD Analog";
+		spec->stream_name_digital = "ALC861VD Digital";
+	}
+
 	spec->stream_analog_playback = &alc861vd_pcm_analog_playback;
 	spec->stream_analog_capture = &alc861vd_pcm_analog_capture;
 
-	spec->stream_name_digital = "ALC861VD Digital";
 	spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
 
@@ -13251,6 +13478,23 @@ static struct hda_input_mux alc662_eeepc_capture_source = {
 	},
 };
 
+static struct hda_input_mux alc663_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+static struct hda_input_mux alc663_m51va_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Ext-Mic", 0x0 },
+		{ "D-Mic", 0x9 },
+	},
+};
+
 #define alc662_mux_enum_info alc_mux_enum_info
 #define alc662_mux_enum_get alc_mux_enum_get
 #define alc662_mux_enum_put alc882_mux_enum_put
@@ -13431,6 +13675,44 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc663_m51va_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_MUTE("DMic Playback Switch", 0x23, 0x9, HDA_INPUT),
+	{ } /* end */
+};
+
+static 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("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static 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("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("i-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 struct snd_kcontrol_new alc662_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -13501,6 +13783,11 @@ static struct hda_verb alc662_init_verbs[] = {
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+	/* always trun on EAPD */
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+
 	{ }
 };
 
@@ -13571,6 +13858,43 @@ static struct hda_verb alc662_auto_init_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc663_m51va_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 */
+
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
+static 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|ALC880_FRONT_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
+	{}
+};
+
+static 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 | ALC880_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
 /* capture mixer elements */
 static struct snd_kcontrol_new alc662_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
@@ -13692,6 +14016,125 @@ static void alc662_eeepc_ep20_inithook(struct hda_codec *codec)
 	alc662_eeepc_ep20_automute(codec);
 }
 
+static void alc663_m51va_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x21, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc663_m51va_mic_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x18, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+	snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+	snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
+	snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
+}
+
+static void alc663_m51va_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_m51va_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc663_m51va_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_m51va_inithook(struct hda_codec *codec)
+{
+	alc663_m51va_speaker_automute(codec);
+	alc663_m51va_mic_automute(codec);
+}
+
+static void alc663_g71v_hp_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x21, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc663_g71v_front_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x15, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc663_g71v_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_g71v_hp_automute(codec);
+		break;
+	case ALC880_FRONT_EVENT:
+		alc663_g71v_front_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_g71v_inithook(struct hda_codec *codec)
+{
+	alc663_g71v_front_automute(codec);
+	alc663_g71v_hp_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+
+static void alc663_g50v_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_m51va_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_g50v_inithook(struct hda_codec *codec)
+{
+	alc663_m51va_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc662_loopbacks	alc880_loopbacks
 #endif
@@ -13714,14 +14157,24 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
 	[ALC662_LENOVO_101E]	= "lenovo-101e",
 	[ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
 	[ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
+	[ALC663_ASUS_M51VA] = "m51va",
+	[ALC663_ASUS_G71V] = "g71v",
+	[ALC663_ASUS_H13] = "h13",
+	[ALC663_ASUS_G50V] = "g50v",
 	[ALC662_AUTO]		= "auto",
 };
 
 static struct snd_pci_quirk alc662_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS G71V", ALC663_ASUS_G71V),
+	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V),
 	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(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+	SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13),
+	SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13),
+	SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),
 	{}
 };
 
@@ -13809,7 +14262,53 @@ static struct alc_config_preset alc662_presets[] = {
 		.unsol_event = alc662_eeepc_ep20_unsol_event,
 		.init_hook = alc662_eeepc_ep20_inithook,
 	},
-
+	[ALC663_ASUS_M51VA] = {
+		.mixers = { alc663_m51va_mixer, alc662_capture_mixer},
+		.init_verbs = { alc662_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,
+		.input_mux = &alc663_m51va_capture_source,
+		.unsol_event = alc663_m51va_unsol_event,
+		.init_hook = alc663_m51va_inithook,
+	},
+	[ALC663_ASUS_G71V] = {
+		.mixers = { alc663_g71v_mixer, alc662_capture_mixer},
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc663_g71v_unsol_event,
+		.init_hook = alc663_g71v_inithook,
+	},
+	[ALC663_ASUS_H13] = {
+		.mixers = { alc663_m51va_mixer, alc662_capture_mixer},
+		.init_verbs = { alc662_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,
+		.input_mux = &alc663_m51va_capture_source,
+		.unsol_event = alc663_m51va_unsol_event,
+		.init_hook = alc663_m51va_inithook,
+	},
+	[ALC663_ASUS_G50V] = {
+		.mixers = { alc663_g50v_mixer, alc662_capture_mixer},
+		.init_verbs = { alc662_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 = alc663_g50v_unsol_event,
+		.init_hook = alc663_g50v_inithook,
+	},
 };
 
 
@@ -14082,6 +14581,8 @@ static int patch_alc662(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	alc_fix_pll_init(codec, 0x20, 0x04, 15);
+
 	board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST,
 						  alc662_models,
 			  	                  alc662_cfg_tbl);
@@ -14108,11 +14609,17 @@ static int patch_alc662(struct hda_codec *codec)
 	if (board_config != ALC662_AUTO)
 		setup_preset(spec, &alc662_presets[board_config]);
 
-	spec->stream_name_analog = "ALC662 Analog";
+	if (codec->vendor_id == 0x10ec0663) {
+		spec->stream_name_analog = "ALC663 Analog";
+		spec->stream_name_digital = "ALC663 Digital";
+	} else {
+		spec->stream_name_analog = "ALC662 Analog";
+		spec->stream_name_digital = "ALC662 Digital";
+	}
+
 	spec->stream_analog_playback = &alc662_pcm_analog_playback;
 	spec->stream_analog_capture = &alc662_pcm_analog_capture;
 
-	spec->stream_name_digital = "ALC662 Digital";
 	spec->stream_digital_playback = &alc662_pcm_digital_playback;
 	spec->stream_digital_capture = &alc662_pcm_digital_capture;
 
@@ -14151,6 +14658,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
 	  .patch = patch_alc883 },
 	{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
 	  .patch = patch_alc662 },
+	{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
 	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
 	{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },

sound/pci/hda/patch_sigmatel.c

@@ -636,21 +636,28 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 };
 
+#define HD_DISABLE_PORTF 3
 static struct hda_verb stac92hd71bxx_analog_core_init[] = {
+	/* start of config #1 */
+
+	/* connect port 0f to audio mixer */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
+	/* unmute right and left channels for node 0x0f */
+	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* start of config #2 */
+
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* connect ports 0d and 0f to audio mixer */
+	/* connect port 0d to audio mixer */
 	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute dac0 input in audio mixer */
 	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
-	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
+	/* unmute right and left channels for nodes 0x0a, 0xd */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{}
 };
 
@@ -818,6 +825,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT),
 
+	HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
+
 	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
 	{ } /* end */
@@ -1317,13 +1327,13 @@ static unsigned int ref92hd71bxx_pin_configs[10] = {
 	0x90a000f0, 0x01452050,
 };
 
-static unsigned int dell_m4_1_pin_configs[13] = {
+static unsigned int dell_m4_1_pin_configs[10] = {
 	0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
 	0x40f000f0, 0x4f0000f0,
 };
 
-static unsigned int dell_m4_2_pin_configs[13] = {
+static unsigned int dell_m4_2_pin_configs[10] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
 	0x40f000f0, 0x044413b0,
@@ -1754,12 +1764,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 		      "unknown Dell", STAC_9205_DELL_M42),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f8,
 		      "Dell Precision", STAC_9205_DELL_M43),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c,
-			  "Dell Precision", STAC_9205_DELL_M43),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9,
 		      "Dell Precision", STAC_9205_DELL_M43),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b,
-		      "Dell Precision", STAC_9205_DELL_M43),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa,
 		      "Dell Precision", STAC_9205_DELL_M43),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc,
@@ -1770,18 +1776,14 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 		      "Dell Precision", STAC_9205_DELL_M43),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ff,
 		      "Dell Precision M4300", STAC_9205_DELL_M43),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206,
-		      "Dell Precision", STAC_9205_DELL_M43),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
-		      "Dell Inspiron", STAC_9205_DELL_M44),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,
-		      "Dell Inspiron", STAC_9205_DELL_M44),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc,
-		      "Dell Inspiron", STAC_9205_DELL_M44),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd,
-		      "Dell Inspiron", STAC_9205_DELL_M44),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0204,
 		      "unknown Dell", STAC_9205_DELL_M42),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206,
+		      "Dell Precision", STAC_9205_DELL_M43),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b,
+		      "Dell Precision", STAC_9205_DELL_M43),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c,
+		      "Dell Precision", STAC_9205_DELL_M43),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f,
 		      "Dell Inspiron", STAC_9205_DELL_M44),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
@@ -3103,13 +3105,16 @@ static int stac92xx_init(struct hda_codec *codec)
 					0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i],
 					0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = get_defcfg_connect(def_conf);
 		/* outputs are only ports capable of power management
 		 * any attempts on powering down a input port cause the
 		 * referenced VREF to act quirky.
 		 */
 		if (pinctl & AC_PINCTL_IN_EN)
 			continue;
-		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED)
+		/* skip any ports that don't have jacks since presence
+ 		 * detection is useless */
+		if (def_conf && def_conf != AC_JACK_PORT_FIXED)
 			continue;
 		enable_pin_detect(codec, spec->pwr_nids[i], event | i);
 		codec->patch_ops.unsol_event(codec, (event | i) << 26);
@@ -3614,6 +3619,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 
 	codec->spec = spec;
 	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
+	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
 	spec->pin_nids = stac92hd71bxx_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD71BXX_MODELS,
@@ -3642,6 +3648,19 @@ again:
 		spec->mixer = stac92hd71bxx_mixer;
 		spec->init = stac92hd71bxx_core_init;
 		break;
+	case 0x111d7608: /* 5 Port with Analog Mixer */
+		/* no output amps */
+		spec->num_pwrs = 0;
+		spec->mixer = stac92hd71bxx_analog_mixer;
+
+		/* disable VSW */
+		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
+		stac92xx_set_config_reg(codec, 0xf, 0x40f000f0);
+		break;
+	case 0x111d7603: /* 6 Port with Analog Mixer */
+		/* no output amps */
+		spec->num_pwrs = 0;
+		/* fallthru */
 	default:
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
@@ -3653,22 +3672,19 @@ again:
 	/* GPIO0 High = EAPD */
 	spec->gpio_mask = 0x01;
 	spec->gpio_dir = 0x01;
-	spec->gpio_mask = 0x01;
 	spec->gpio_data = 0x01;
 
 	spec->mux_nids = stac92hd71bxx_mux_nids;
 	spec->adc_nids = stac92hd71bxx_adc_nids;
 	spec->dmic_nids = stac92hd71bxx_dmic_nids;
 	spec->dmux_nids = stac92hd71bxx_dmux_nids;
+	spec->pwr_nids = stac92hd71bxx_pwr_nids;
 
 	spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
 	spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 
-	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
-	spec->pwr_nids = stac92hd71bxx_pwr_nids;
-
 	spec->multiout.num_dacs = 1;
 	spec->multiout.hp_nid = 0x11;
 	spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
@@ -4306,10 +4322,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
  	{ .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 },
  	{ .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
  	{ .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
+	{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
+	{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx },
 	{ .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
 	{ .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
 	{ .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },

sound/pci/ice1712/envy24ht.h

@@ -93,9 +93,13 @@ enum {
 #define VT1724_REG_MPU_TXFIFO		0x0a	/*byte ro. number of bytes in TX fifo*/
 #define VT1724_REG_MPU_RXFIFO		0x0b	/*byte ro. number of bytes in RX fifo*/
 
-//are these 2 the wrong way around? they don't seem to be used yet anyway
-#define VT1724_REG_MPU_CTRL		0x0c	/* byte */
-#define VT1724_REG_MPU_DATA		0x0d	/* byte */
+#define VT1724_REG_MPU_DATA		0x0c	/* byte */
+#define VT1724_REG_MPU_CTRL		0x0d	/* byte */
+#define   VT1724_MPU_UART	0x01
+#define   VT1724_MPU_TX_EMPTY	0x02
+#define   VT1724_MPU_TX_FULL	0x04
+#define   VT1724_MPU_RX_EMPTY	0x08
+#define   VT1724_MPU_RX_FULL	0x10
 
 #define VT1724_REG_MPU_FIFO_WM	0x0e	/*byte set the high/low watermarks for RX/TX fifos*/
 #define   VT1724_MPU_RX_FIFO	0x20	//1=rx fifo watermark 0=tx fifo watermark

sound/pci/ice1712/ice1712.h

@@ -333,6 +333,8 @@ struct snd_ice1712 {
 	unsigned int has_spdif: 1;	/* VT1720/4 - has SPDIF I/O */
 	unsigned int force_pdma4: 1;	/* VT1720/4 - PDMA4 as non-spdif */
 	unsigned int force_rdma1: 1;	/* VT1720/4 - RDMA1 as non-spdif */
+	unsigned int midi_output: 1;	/* VT1720/4: MIDI output triggered */
+	unsigned int midi_input: 1;	/* VT1720/4: MIDI input triggered */
 	unsigned int num_total_dacs;	/* total DACs */
 	unsigned int num_total_adcs;	/* total ADCs */
 	unsigned int cur_rate;		/* current rate */

sound/pci/ice1712/ice1724.c

@@ -32,7 +32,7 @@
 #include <linux/mutex.h>
 #include <sound/core.h>
 #include <sound/info.h>
-#include <sound/mpu401.h>
+#include <sound/rawmidi.h>
 #include <sound/initval.h>
 
 #include <sound/asoundef.h>
@@ -223,30 +223,153 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice)
 }
 
 /*
- * MPU401 accessor
+ * MIDI
  */
-static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu,
-					    unsigned long addr)
+
+static void vt1724_midi_clear_rx(struct snd_ice1712 *ice)
+{
+	unsigned int count;
+
+	for (count = inb(ICEREG1724(ice, MPU_RXFIFO)); count > 0; --count)
+		inb(ICEREG1724(ice, MPU_DATA));
+}
+
+static inline struct snd_rawmidi_substream *
+get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream)
 {
-	/* fix status bits to the standard position */
-	/* only RX_EMPTY and TX_FULL are checked */
-	if (addr == MPU401C(mpu))
-		return (inb(addr) & 0x0c) << 4;
+	return list_first_entry(&ice->rmidi[0]->streams[stream].substreams,
+				struct snd_rawmidi_substream, list);
+}
+
+static void vt1724_midi_write(struct snd_ice1712 *ice)
+{
+	struct snd_rawmidi_substream *s;
+	int count, i;
+	u8 buffer[32];
+
+	s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_OUTPUT);
+	count = 31 - inb(ICEREG1724(ice, MPU_TXFIFO));
+	if (count > 0) {
+		count = snd_rawmidi_transmit(s, buffer, count);
+		for (i = 0; i < count; ++i)
+			outb(buffer[i], ICEREG1724(ice, MPU_DATA));
+	}
+}
+
+static void vt1724_midi_read(struct snd_ice1712 *ice)
+{
+	struct snd_rawmidi_substream *s;
+	int count, i;
+	u8 buffer[32];
+
+	s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_INPUT);
+	count = inb(ICEREG1724(ice, MPU_RXFIFO));
+	if (count > 0) {
+		count = min(count, 32);
+		for (i = 0; i < count; ++i)
+			buffer[i] = inb(ICEREG1724(ice, MPU_DATA));
+		snd_rawmidi_receive(s, buffer, count);
+	}
+}
+
+static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
+				   u8 flag, int enable)
+{
+	struct snd_ice1712 *ice = substream->rmidi->private_data;
+	u8 mask;
+
+	spin_lock_irq(&ice->reg_lock);
+	mask = inb(ICEREG1724(ice, IRQMASK));
+	if (enable)
+		mask &= ~flag;
 	else
-		return inb(addr);
+		mask |= flag;
+	outb(mask, ICEREG1724(ice, IRQMASK));
+	spin_unlock_irq(&ice->reg_lock);
 }
 
-static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu,
-				    unsigned char data, unsigned long addr)
+static int vt1724_midi_output_open(struct snd_rawmidi_substream *s)
 {
-	if (addr == MPU401C(mpu)) {
-		if (data == MPU401_ENTER_UART)
-			outb(0x01, addr);
-		/* what else? */
-	} else
-		outb(data, addr);
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1);
+	return 0;
+}
+
+static int vt1724_midi_output_close(struct snd_rawmidi_substream *s)
+{
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
+	return 0;
 }
 
+static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up)
+{
+	struct snd_ice1712 *ice = s->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&ice->reg_lock, flags);
+	if (up) {
+		ice->midi_output = 1;
+		vt1724_midi_write(ice);
+	} else {
+		ice->midi_output = 0;
+	}
+	spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s)
+{
+	struct snd_ice1712 *ice = s->rmidi->private_data;
+	unsigned long timeout;
+
+	/* 32 bytes should be transmitted in less than about 12 ms */
+	timeout = jiffies + msecs_to_jiffies(15);
+	do {
+		if (inb(ICEREG1724(ice, MPU_CTRL)) & VT1724_MPU_TX_EMPTY)
+			break;
+		schedule_timeout_uninterruptible(1);
+	} while (time_after(timeout, jiffies));
+}
+
+static struct snd_rawmidi_ops vt1724_midi_output_ops = {
+	.open = vt1724_midi_output_open,
+	.close = vt1724_midi_output_close,
+	.trigger = vt1724_midi_output_trigger,
+	.drain = vt1724_midi_output_drain,
+};
+
+static int vt1724_midi_input_open(struct snd_rawmidi_substream *s)
+{
+	vt1724_midi_clear_rx(s->rmidi->private_data);
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 1);
+	return 0;
+}
+
+static int vt1724_midi_input_close(struct snd_rawmidi_substream *s)
+{
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 0);
+	return 0;
+}
+
+static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up)
+{
+	struct snd_ice1712 *ice = s->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&ice->reg_lock, flags);
+	if (up) {
+		ice->midi_input = 1;
+		vt1724_midi_read(ice);
+	} else {
+		ice->midi_input = 0;
+	}
+	spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops vt1724_midi_input_ops = {
+	.open = vt1724_midi_input_open,
+	.close = vt1724_midi_input_close,
+	.trigger = vt1724_midi_input_trigger,
+};
+
 
 /*
  *  Interrupt handler
@@ -278,13 +401,10 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 #endif
 		handled = 1;		
 		if (status & VT1724_IRQ_MPU_TX) {
-			if (ice->rmidi[0])
-				snd_mpu401_uart_interrupt_tx(irq,
-					ice->rmidi[0]->private_data);
-			else /* disable TX to be sure */
-				outb(inb(ICEREG1724(ice, IRQMASK)) |
-				     VT1724_IRQ_MPU_TX,
-				     ICEREG1724(ice, IRQMASK));
+			spin_lock(&ice->reg_lock);
+			if (ice->midi_output)
+				vt1724_midi_write(ice);
+			spin_unlock(&ice->reg_lock);
 			/* Due to mysterical reasons, MPU_TX is always
 			 * generated (and can't be cleared) when a PCM
 			 * playback is going.  So let's ignore at the
@@ -293,13 +413,12 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 			status_mask &= ~VT1724_IRQ_MPU_TX;
 		}
 		if (status & VT1724_IRQ_MPU_RX) {
-			if (ice->rmidi[0])
-				snd_mpu401_uart_interrupt(irq,
-					ice->rmidi[0]->private_data);
-			else /* disable RX to be sure */
-				outb(inb(ICEREG1724(ice, IRQMASK)) |
-				     VT1724_IRQ_MPU_RX,
-				     ICEREG1724(ice, IRQMASK));
+			spin_lock(&ice->reg_lock);
+			if (ice->midi_input)
+				vt1724_midi_read(ice);
+			else
+				vt1724_midi_clear_rx(ice);
+			spin_unlock(&ice->reg_lock);
 		}
 		/* ack MPU irq */
 		outb(status, ICEREG1724(ice, IRQSTAT));
@@ -2425,28 +2544,30 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 
 	if (! c->no_mpu401) {
 		if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) {
-			struct snd_mpu401 *mpu;
-			if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
-						       ICEREG1724(ice, MPU_CTRL),
-						       (MPU401_INFO_INTEGRATED |
-							MPU401_INFO_NO_ACK |
-							MPU401_INFO_TX_IRQ),
-						       ice->irq, 0,
-						       &ice->rmidi[0])) < 0) {
+			struct snd_rawmidi *rmidi;
+
+			err = snd_rawmidi_new(card, "MIDI", 0, 1, 1, &rmidi);
+			if (err < 0) {
 				snd_card_free(card);
 				return err;
 			}
-			mpu = ice->rmidi[0]->private_data;
-			mpu->read = snd_vt1724_mpu401_read;
-			mpu->write = snd_vt1724_mpu401_write;
-			/* unmask MPU RX/TX irqs */
-			outb(inb(ICEREG1724(ice, IRQMASK)) &
-			     ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX),
-			     ICEREG1724(ice, IRQMASK));
+			ice->rmidi[0] = rmidi;
+			rmidi->private_data = ice;
+			strcpy(rmidi->name, "ICE1724 MIDI");
+			rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+					    SNDRV_RAWMIDI_INFO_INPUT |
+					    SNDRV_RAWMIDI_INFO_DUPLEX;
+			snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+					    &vt1724_midi_output_ops);
+			snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+					    &vt1724_midi_input_ops);
+
 			/* set watermarks */
 			outb(VT1724_MPU_RX_FIFO | 0x1,
 			     ICEREG1724(ice, MPU_FIFO_WM));
 			outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
+			/* set UART mode */
+			outb(VT1724_MPU_UART, ICEREG1724(ice, MPU_CTRL));
 		}
 	}
 

sound/pci/maestro3.c

@@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, int enable)
 	outw(0xffff, io + GPIO_MASK);
 }
 
+static void
+snd_m3_hv_init(struct snd_m3 *chip)
+{
+	unsigned long io = chip->iobase;
+	u16 val = GPI_VOL_DOWN | GPI_VOL_UP;
+
+	if (!chip->is_omnibook)
+		return;
+
+	/*
+	 * Volume buttons on some HP OmniBook laptops
+	 * require some GPIO magic to work correctly.
+	 */
+	outw(0xffff, io + GPIO_MASK);
+	outw(0x0000, io + GPIO_DATA);
+
+	outw(~val, io + GPIO_MASK);
+	outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION);
+	outw(val, io + GPIO_MASK);
+
+	outw(0xffff, io + GPIO_MASK);
+}
+
 static int
 snd_m3_chip_init(struct snd_m3 *chip)
 {
@@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip)
 	       DISABLE_LEGACY);
 	pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w);
 
-	if (chip->is_omnibook) {
-		/*
-		 * Volume buttons on some HP OmniBook laptops don't work
-		 * correctly. This makes them work for the most part.
-		 *
-		 * Volume up and down buttons on the laptop side work.
-		 * Fn+cursor_up (volme up) works.
-		 * Fn+cursor_down (volume down) doesn't work.
-		 * Fn+F7 (mute) works acts as volume up.
-		 */
-		outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK);
-		outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION);
-		outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA);
-		outw(0xffff, io + GPIO_MASK);
-	}
 	pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n);
 	n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD);
 	n |= chip->hv_config;
@@ -2642,6 +2650,8 @@ static int m3_resume(struct pci_dev *pci)
 	snd_m3_enable_ints(chip);
 	snd_m3_amp_enable(chip, 1);
 
+	snd_m3_hv_init(chip);
+
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
@@ -2781,6 +2791,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci,
 
 	snd_m3_amp_enable(chip, 1);
 
+	snd_m3_hv_init(chip);
+
 	tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip);
 
 	if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED,

sound/pci/nm256/nm256.c

@@ -1302,8 +1302,8 @@ snd_nm256_mixer(struct nm256 *chip)
 		.read = snd_nm256_ac97_read,
 	};
 
-	chip->ac97_regs = kcalloc(sizeof(short),
-				  ARRAY_SIZE(nm256_ac97_init_val), GFP_KERNEL);
+	chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val),
+				  sizeof(short), GFP_KERNEL);
 	if (! chip->ac97_regs)
 		return -ENOMEM;
 

sound/pci/oxygen/hifier.c

@@ -28,7 +28,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("TempoTec HiFier driver");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
@@ -62,16 +62,28 @@ static void ak4396_write(struct oxygen *chip, u8 reg, u8 value)
 			 AK4396_WRITE | (reg << 8) | value);
 }
 
-static void hifier_init(struct oxygen *chip)
+static void update_ak4396_volume(struct oxygen *chip)
+{
+	ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]);
+	ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]);
+}
+
+static void hifier_registers_init(struct oxygen *chip)
 {
 	struct hifier_data *data = chip->model_data;
 
-	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
 	ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
 	ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2);
 	ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM);
-	ak4396_write(chip, AK4396_LCH_ATT, 0);
-	ak4396_write(chip, AK4396_RCH_ATT, 0);
+	update_ak4396_volume(chip);
+}
+
+static void hifier_init(struct oxygen *chip)
+{
+	struct hifier_data *data = chip->model_data;
+
+	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+	hifier_registers_init(chip);
 
 	snd_component_add(chip->card, "AK4396");
 	snd_component_add(chip->card, "CS5340");
@@ -100,12 +112,6 @@ static void set_ak4396_params(struct oxygen *chip,
 	ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
 }
 
-static void update_ak4396_volume(struct oxygen *chip)
-{
-	ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]);
-	ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]);
-}
-
 static void update_ak4396_mute(struct oxygen *chip)
 {
 	struct hifier_data *data = chip->model_data;
@@ -140,6 +146,7 @@ static const struct oxygen_model model_hifier = {
 	.init = hifier_init,
 	.control_filter = hifier_control_filter,
 	.cleanup = hifier_cleanup,
+	.resume = hifier_registers_init,
 	.set_dac_params = set_ak4396_params,
 	.set_adc_params = set_cs5340_params,
 	.update_dac_volume = update_ak4396_volume,
@@ -180,6 +187,10 @@ static struct pci_driver hifier_driver = {
 	.id_table = hifier_ids,
 	.probe = hifier_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+	.suspend = oxygen_pci_suspend,
+	.resume = oxygen_pci_resume,
+#endif
 };
 
 static int __init alsa_card_hifier_init(void)

sound/pci/oxygen/oxygen.c

@@ -43,7 +43,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 driver");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
@@ -80,6 +80,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids);
 
 struct generic_data {
 	u8 ak4396_ctl2;
+	u16 saved_wm8785_registers[2];
 };
 
 static void ak4396_write(struct oxygen *chip, unsigned int codec,
@@ -99,20 +100,35 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec,
 
 static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value)
 {
+	struct generic_data *data = chip->model_data;
+
 	oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER |
 			 OXYGEN_SPI_DATA_LENGTH_2 |
 			 OXYGEN_SPI_CLOCK_160 |
 			 (3 << OXYGEN_SPI_CODEC_SHIFT) |
 			 OXYGEN_SPI_CEN_LATCH_CLOCK_LO,
 			 (reg << 9) | value);
+	if (reg < ARRAY_SIZE(data->saved_wm8785_registers))
+		data->saved_wm8785_registers[reg] = value;
 }
 
-static void ak4396_init(struct oxygen *chip)
+static void update_ak4396_volume(struct oxygen *chip)
+{
+	unsigned int i;
+
+	for (i = 0; i < 4; ++i) {
+		ak4396_write(chip, i,
+			     AK4396_LCH_ATT, chip->dac_volume[i * 2]);
+		ak4396_write(chip, i,
+			     AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]);
+	}
+}
+
+static void ak4396_registers_init(struct oxygen *chip)
 {
 	struct generic_data *data = chip->model_data;
 	unsigned int i;
 
-	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
 	for (i = 0; i < 4; ++i) {
 		ak4396_write(chip, i,
 			     AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
@@ -120,9 +136,16 @@ static void ak4396_init(struct oxygen *chip)
 			     AK4396_CONTROL_2, data->ak4396_ctl2);
 		ak4396_write(chip, i,
 			     AK4396_CONTROL_3, AK4396_PCM);
-		ak4396_write(chip, i, AK4396_LCH_ATT, 0);
-		ak4396_write(chip, i, AK4396_RCH_ATT, 0);
 	}
+	update_ak4396_volume(chip);
+}
+
+static void ak4396_init(struct oxygen *chip)
+{
+	struct generic_data *data = chip->model_data;
+
+	data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+	ak4396_registers_init(chip);
 	snd_component_add(chip->card, "AK4396");
 }
 
@@ -133,12 +156,23 @@ static void ak5385_init(struct oxygen *chip)
 	snd_component_add(chip->card, "AK5385");
 }
 
-static void wm8785_init(struct oxygen *chip)
+static void wm8785_registers_init(struct oxygen *chip)
 {
+	struct generic_data *data = chip->model_data;
+
 	wm8785_write(chip, WM8785_R7, 0);
-	wm8785_write(chip, WM8785_R0, WM8785_MCR_SLAVE |
-		     WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST);
-	wm8785_write(chip, WM8785_R1, WM8785_WL_24);
+	wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]);
+	wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]);
+}
+
+static void wm8785_init(struct oxygen *chip)
+{
+	struct generic_data *data = chip->model_data;
+
+	data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE |
+		WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST;
+	data->saved_wm8785_registers[1] = WM8785_WL_24;
+	wm8785_registers_init(chip);
 	snd_component_add(chip->card, "WM8785");
 }
 
@@ -158,6 +192,12 @@ static void generic_cleanup(struct oxygen *chip)
 {
 }
 
+static void generic_resume(struct oxygen *chip)
+{
+	ak4396_registers_init(chip);
+	wm8785_registers_init(chip);
+}
+
 static void set_ak4396_params(struct oxygen *chip,
 			      struct snd_pcm_hw_params *params)
 {
@@ -183,18 +223,6 @@ static void set_ak4396_params(struct oxygen *chip,
 	}
 }
 
-static void update_ak4396_volume(struct oxygen *chip)
-{
-	unsigned int i;
-
-	for (i = 0; i < 4; ++i) {
-		ak4396_write(chip, i,
-			     AK4396_LCH_ATT, chip->dac_volume[i * 2]);
-		ak4396_write(chip, i,
-			     AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]);
-	}
-}
-
 static void update_ak4396_mute(struct oxygen *chip)
 {
 	struct generic_data *data = chip->model_data;
@@ -256,6 +284,7 @@ static const struct oxygen_model model_generic = {
 	.owner = THIS_MODULE,
 	.init = generic_init,
 	.cleanup = generic_cleanup,
+	.resume = generic_resume,
 	.set_dac_params = set_ak4396_params,
 	.set_adc_params = set_wm8785_params,
 	.update_dac_volume = update_ak4396_volume,
@@ -283,6 +312,7 @@ static const struct oxygen_model model_meridian = {
 	.owner = THIS_MODULE,
 	.init = meridian_init,
 	.cleanup = generic_cleanup,
+	.resume = ak4396_registers_init,
 	.set_dac_params = set_ak4396_params,
 	.set_adc_params = set_ak5385_params,
 	.update_dac_volume = update_ak4396_volume,
@@ -331,6 +361,10 @@ static struct pci_driver oxygen_driver = {
 	.id_table = oxygen_ids,
 	.probe = generic_oxygen_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+	.suspend = oxygen_pci_suspend,
+	.resume = oxygen_pci_resume,
+#endif
 };
 
 static int __init alsa_card_oxygen_init(void)

sound/pci/oxygen/oxygen.h

@@ -16,6 +16,8 @@
 #define PCM_AC97	5
 #define PCM_COUNT	6
 
+#define OXYGEN_IO_SIZE	0x100
+
 /* model-specific configuration of outputs/inputs */
 #define PLAYBACK_0_TO_I2S	0x001
 #define PLAYBACK_1_TO_SPDIF	0x004
@@ -78,6 +80,12 @@ struct oxygen {
 	struct work_struct spdif_input_bits_work;
 	struct work_struct gpio_work;
 	wait_queue_head_t ac97_waitqueue;
+	union {
+		u8 _8[OXYGEN_IO_SIZE];
+		__le16 _16[OXYGEN_IO_SIZE / 2];
+		__le32 _32[OXYGEN_IO_SIZE / 4];
+	} saved_registers;
+	u16 saved_ac97_registers[2][0x40];
 };
 
 struct oxygen_model {
@@ -89,6 +97,8 @@ struct oxygen_model {
 	int (*control_filter)(struct snd_kcontrol_new *template);
 	int (*mixer_init)(struct oxygen *chip);
 	void (*cleanup)(struct oxygen *chip);
+	void (*suspend)(struct oxygen *chip);
+	void (*resume)(struct oxygen *chip);
 	void (*pcm_hardware_filter)(unsigned int channel,
 				    struct snd_pcm_hardware *hardware);
 	void (*set_dac_params)(struct oxygen *chip,
@@ -117,6 +127,10 @@ struct oxygen_model {
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		     const struct oxygen_model *model);
 void oxygen_pci_remove(struct pci_dev *pci);
+#ifdef CONFIG_PM
+int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
+int oxygen_pci_resume(struct pci_dev *pci);
+#endif
 
 /* oxygen_mixer.c */
 

sound/pci/oxygen/oxygen_io.c

@@ -44,18 +44,21 @@ EXPORT_SYMBOL(oxygen_read32);
 void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value)
 {
 	outb(value, chip->addr + reg);
+	chip->saved_registers._8[reg] = value;
 }
 EXPORT_SYMBOL(oxygen_write8);
 
 void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value)
 {
 	outw(value, chip->addr + reg);
+	chip->saved_registers._16[reg / 2] = cpu_to_le16(value);
 }
 EXPORT_SYMBOL(oxygen_write16);
 
 void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value)
 {
 	outl(value, chip->addr + reg);
+	chip->saved_registers._32[reg / 4] = cpu_to_le32(value);
 }
 EXPORT_SYMBOL(oxygen_write32);
 
@@ -63,7 +66,10 @@ void oxygen_write8_masked(struct oxygen *chip, unsigned int reg,
 			  u8 value, u8 mask)
 {
 	u8 tmp = inb(chip->addr + reg);
-	outb((tmp & ~mask) | (value & mask), chip->addr + reg);
+	tmp &= ~mask;
+	tmp |= value & mask;
+	outb(tmp, chip->addr + reg);
+	chip->saved_registers._8[reg] = tmp;
 }
 EXPORT_SYMBOL(oxygen_write8_masked);
 
@@ -71,7 +77,10 @@ void oxygen_write16_masked(struct oxygen *chip, unsigned int reg,
 			   u16 value, u16 mask)
 {
 	u16 tmp = inw(chip->addr + reg);
-	outw((tmp & ~mask) | (value & mask), chip->addr + reg);
+	tmp &= ~mask;
+	tmp |= value & mask;
+	outw(tmp, chip->addr + reg);
+	chip->saved_registers._16[reg / 2] = cpu_to_le16(tmp);
 }
 EXPORT_SYMBOL(oxygen_write16_masked);
 
@@ -79,7 +88,10 @@ void oxygen_write32_masked(struct oxygen *chip, unsigned int reg,
 			   u32 value, u32 mask)
 {
 	u32 tmp = inl(chip->addr + reg);
-	outl((tmp & ~mask) | (value & mask), chip->addr + reg);
+	tmp &= ~mask;
+	tmp |= value & mask;
+	outl(tmp, chip->addr + reg);
+	chip->saved_registers._32[reg / 4] = cpu_to_le32(tmp);
 }
 EXPORT_SYMBOL(oxygen_write32_masked);
 
@@ -128,8 +140,10 @@ void oxygen_write_ac97(struct oxygen *chip, unsigned int codec,
 		oxygen_write32(chip, OXYGEN_AC97_REGS, reg);
 		/* require two "completed" writes, just to be sure */
 		if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 &&
-		    ++succeeded >= 2)
+		    ++succeeded >= 2) {
+			chip->saved_ac97_registers[codec][index / 2] = data;
 			return;
+		}
 	}
 	snd_printk(KERN_ERR "AC'97 write timeout\n");
 }

sound/pci/oxygen/oxygen_lib.c

@@ -32,7 +32,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 helper library");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 
 
 static irqreturn_t oxygen_interrupt(int dummy, void *dev_id)
@@ -173,7 +173,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry,
 	int i, j;
 
 	snd_iprintf(buffer, "CMI8788\n\n");
-	for (i = 0; i < 0x100; i += 0x10) {
+	for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) {
 		snd_iprintf(buffer, "%02x:", i);
 		for (j = 0; j < 0x10; ++j)
 			snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j));
@@ -314,6 +314,10 @@ static void oxygen_init(struct oxygen *chip)
 				    OXYGEN_SPDIF_LOCK_MASK |
 				    OXYGEN_SPDIF_RATE_MASK);
 	oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits);
+	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+		       OXYGEN_2WIRE_LENGTH_8 |
+		       OXYGEN_2WIRE_INTERRUPT_MASK |
+		       OXYGEN_2WIRE_SPEED_STANDARD);
 	oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK);
 	oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0);
 	oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0);
@@ -455,7 +459,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	}
 
 	if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) ||
-	    pci_resource_len(pci, 0) < 0x100) {
+	    pci_resource_len(pci, 0) < OXYGEN_IO_SIZE) {
 		snd_printk(KERN_ERR "invalid PCI I/O range\n");
 		err = -ENXIO;
 		goto err_pci_regions;
@@ -534,3 +538,99 @@ void oxygen_pci_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 EXPORT_SYMBOL(oxygen_pci_remove);
+
+#ifdef CONFIG_PM
+int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct oxygen *chip = card->private_data;
+	unsigned int i, saved_interrupt_mask;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+	for (i = 0; i < PCM_COUNT; ++i)
+		if (chip->streams[i])
+			snd_pcm_suspend(chip->streams[i]);
+
+	if (chip->model->suspend)
+		chip->model->suspend(chip);
+
+	spin_lock_irq(&chip->reg_lock);
+	saved_interrupt_mask = chip->interrupt_mask;
+	chip->interrupt_mask = 0;
+	oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+	oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+	spin_unlock_irq(&chip->reg_lock);
+
+	synchronize_irq(chip->irq);
+	flush_scheduled_work();
+	chip->interrupt_mask = saved_interrupt_mask;
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+	pci_set_power_state(pci, pci_choose_state(pci, state));
+	return 0;
+}
+EXPORT_SYMBOL(oxygen_pci_suspend);
+
+static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = {
+	0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff,
+	0x00300000, 0x00000fe4, 0x0ff7001f, 0x00000000
+};
+static const u32 ac97_registers_to_restore[2][0x40 / 32] = {
+	{ 0x18284fa2, 0x03060000 },
+	{ 0x00007fa6, 0x00200000 }
+};
+
+static inline int is_bit_set(const u32 *bitmap, unsigned int bit)
+{
+	return bitmap[bit / 32] & (1 << (bit & 31));
+}
+
+static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec)
+{
+	unsigned int i;
+
+	oxygen_write_ac97(chip, codec, AC97_RESET, 0);
+	msleep(1);
+	for (i = 1; i < 0x40; ++i)
+		if (is_bit_set(ac97_registers_to_restore[codec], i))
+			oxygen_write_ac97(chip, codec, i * 2,
+					  chip->saved_ac97_registers[codec][i]);
+}
+
+int oxygen_pci_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct oxygen *chip = card->private_data;
+	unsigned int i;
+
+	pci_set_power_state(pci, PCI_D0);
+	pci_restore_state(pci);
+	if (pci_enable_device(pci) < 0) {
+		snd_printk(KERN_ERR "cannot reenable device");
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+	pci_set_master(pci);
+
+	oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+	oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+	for (i = 0; i < OXYGEN_IO_SIZE; ++i)
+		if (is_bit_set(registers_to_restore, i))
+			oxygen_write8(chip, i, chip->saved_registers._8[i]);
+	if (chip->has_ac97_0)
+		oxygen_restore_ac97(chip, 0);
+	if (chip->has_ac97_1)
+		oxygen_restore_ac97(chip, 1);
+
+	if (chip->model->resume)
+		chip->model->resume(chip);
+
+	oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+EXPORT_SYMBOL(oxygen_pci_resume);
+#endif /* CONFIG_PM */

sound/pci/oxygen/oxygen_pcm.c

@@ -24,6 +24,16 @@
 #include <sound/pcm_params.h>
 #include "oxygen.h"
 
+/* most DMA channels have a 16-bit counter for 32-bit words */
+#define BUFFER_BYTES_MAX		((1 << 16) * 4)
+/* the multichannel DMA channel has a 24-bit counter */
+#define BUFFER_BYTES_MAX_MULTICH	((1 << 24) * 4)
+
+#define PERIOD_BYTES_MIN		64
+
+#define DEFAULT_BUFFER_BYTES		(BUFFER_BYTES_MAX / 2)
+#define DEFAULT_BUFFER_BYTES_MULTICH	(1024 * 1024)
+
 static const struct snd_pcm_hardware oxygen_stereo_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_MMAP_VALID |
@@ -44,11 +54,11 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = {
 	.rate_max = 192000,
 	.channels_min = 2,
 	.channels_max = 2,
-	.buffer_bytes_max = 256 * 1024,
-	.period_bytes_min = 128,
-	.period_bytes_max = 128 * 1024,
+	.buffer_bytes_max = BUFFER_BYTES_MAX,
+	.period_bytes_min = PERIOD_BYTES_MIN,
+	.period_bytes_max = BUFFER_BYTES_MAX / 2,
 	.periods_min = 2,
-	.periods_max = 2048,
+	.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
 };
 static const struct snd_pcm_hardware oxygen_multichannel_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP |
@@ -70,11 +80,11 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = {
 	.rate_max = 192000,
 	.channels_min = 2,
 	.channels_max = 8,
-	.buffer_bytes_max = 2048 * 1024,
-	.period_bytes_min = 128,
-	.period_bytes_max = 256 * 1024,
+	.buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH,
+	.period_bytes_min = PERIOD_BYTES_MIN,
+	.period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2,
 	.periods_min = 2,
-	.periods_max = 16384,
+	.periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN,
 };
 static const struct snd_pcm_hardware oxygen_ac97_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP |
@@ -88,11 +98,11 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = {
 	.rate_max = 48000,
 	.channels_min = 2,
 	.channels_max = 2,
-	.buffer_bytes_max = 256 * 1024,
-	.period_bytes_min = 128,
-	.period_bytes_max = 128 * 1024,
+	.buffer_bytes_max = BUFFER_BYTES_MAX,
+	.period_bytes_min = PERIOD_BYTES_MIN,
+	.period_bytes_max = BUFFER_BYTES_MAX / 2,
 	.periods_min = 2,
-	.periods_max = 2048,
+	.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
 };
 
 static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = {
@@ -155,6 +165,12 @@ static int oxygen_open(struct snd_pcm_substream *substream,
 		if (err < 0)
 			return err;
 	}
+	if (channel == PCM_MULTICH) {
+		err = snd_pcm_hw_constraint_minmax
+			(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000);
+		if (err < 0)
+			return err;
+	}
 	snd_pcm_set_sync(substream);
 	chip->streams[channel] = substream;
 
@@ -517,6 +533,7 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 		pausing = 0;
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -663,12 +680,14 @@ int oxygen_pcm_init(struct oxygen *chip)
 			snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
 						      SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      512 * 1024, 2048 * 1024);
+						      DEFAULT_BUFFER_BYTES_MULTICH,
+						      BUFFER_BYTES_MAX_MULTICH);
 		if (ins)
 			snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
 						      SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      128 * 1024, 256 * 1024);
+						      DEFAULT_BUFFER_BYTES,
+						      BUFFER_BYTES_MAX);
 	}
 
 	outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF);
@@ -688,7 +707,8 @@ int oxygen_pcm_init(struct oxygen *chip)
 		strcpy(pcm->name, "Digital");
 		snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      128 * 1024, 256 * 1024);
+						      DEFAULT_BUFFER_BYTES,
+						      BUFFER_BYTES_MAX);
 	}
 
 	if (chip->has_ac97_1) {
@@ -718,7 +738,8 @@ int oxygen_pcm_init(struct oxygen *chip)
 		strcpy(pcm->name, outs ? "Front Panel" : "Analog 2");
 		snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 						      snd_dma_pci_data(chip->pci),
-						      128 * 1024, 256 * 1024);
+						      DEFAULT_BUFFER_BYTES,
+						      BUFFER_BYTES_MAX);
 	}
 	return 0;
 }

sound/pci/oxygen/virtuoso.c

@@ -79,7 +79,7 @@
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("Asus AVx00 driver");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
@@ -132,6 +132,9 @@ struct xonar_data {
 	u8 ext_power_int_reg;
 	u8 ext_power_bit;
 	u8 has_power;
+	u8 pcm1796_oversampling;
+	u8 cs4398_fm;
+	u8 cs4362a_fm;
 };
 
 static void pcm1796_write(struct oxygen *chip, unsigned int codec,
@@ -159,6 +162,14 @@ static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value)
 	oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value);
 }
 
+static void xonar_enable_output(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	msleep(data->anti_pop_delay);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+}
+
 static void xonar_common_init(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -170,32 +181,59 @@ static void xonar_common_init(struct oxygen *chip)
 		data->has_power = !!(oxygen_read8(chip, data->ext_power_reg)
 				     & data->ext_power_bit);
 	}
-	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+			  GPIO_CS53x1_M_MASK | data->output_enable_bit);
 	oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
 			      GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK);
 	oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC);
-	msleep(data->anti_pop_delay);
-	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit);
-	oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+	xonar_enable_output(chip);
 }
 
-static void xonar_d2_init(struct oxygen *chip)
+static void update_pcm1796_volume(struct oxygen *chip)
 {
-	struct xonar_data *data = chip->model_data;
 	unsigned int i;
 
-	data->anti_pop_delay = 300;
-	data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
+	for (i = 0; i < 4; ++i) {
+		pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]);
+		pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]);
+	}
+}
+
+static void update_pcm1796_mute(struct oxygen *chip)
+{
+	unsigned int i;
+	u8 value;
+
+	value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD;
+	if (chip->dac_mute)
+		value |= PCM1796_MUTE;
+	for (i = 0; i < 4; ++i)
+		pcm1796_write(chip, i, 18, value);
+}
+
+static void pcm1796_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+	unsigned int i;
 
 	for (i = 0; i < 4; ++i) {
-		pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED |
-			      PCM1796_FMT_24_LJUST | PCM1796_ATLD);
 		pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1);
-		pcm1796_write(chip, i, 20, PCM1796_OS_64);
+		pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
 		pcm1796_write(chip, i, 21, 0);
-		pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */
-		pcm1796_write(chip, i, 17, 0x0f);
 	}
+	update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */
+	update_pcm1796_volume(chip);
+}
+
+static void xonar_d2_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->anti_pop_delay = 300;
+	data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
+	data->pcm1796_oversampling = PCM1796_OS_64;
+
+	pcm1796_init(chip);
 
 	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT);
 	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT);
@@ -217,31 +255,47 @@ static void xonar_d2x_init(struct oxygen *chip)
 	xonar_d2_init(chip);
 }
 
-static void xonar_dx_init(struct oxygen *chip)
+static void update_cs4362a_volumes(struct oxygen *chip)
 {
-	struct xonar_data *data = chip->model_data;
+	u8 mute;
 
-	data->anti_pop_delay = 800;
-	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
-	data->ext_power_reg = OXYGEN_GPI_DATA;
-	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
-	data->ext_power_bit = GPI_DX_EXT_POWER;
+	mute = chip->dac_mute ? CS4362A_MUTE : 0;
+	cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute);
+	cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute);
+	cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute);
+	cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute);
+	cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute);
+	cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute);
+}
 
-	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
-		       OXYGEN_2WIRE_LENGTH_8 |
-		       OXYGEN_2WIRE_INTERRUPT_MASK |
-		       OXYGEN_2WIRE_SPEED_FAST);
+static void update_cs43xx_volume(struct oxygen *chip)
+{
+	cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2);
+	cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2);
+	update_cs4362a_volumes(chip);
+}
+
+static void update_cs43xx_mute(struct oxygen *chip)
+{
+	u8 reg;
+
+	reg = CS4398_MUTEP_LOW | CS4398_PAMUTE;
+	if (chip->dac_mute)
+		reg |= CS4398_MUTE_B | CS4398_MUTE_A;
+	cs4398_write(chip, 4, reg);
+	update_cs4362a_volumes(chip);
+}
+
+static void cs43xx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
 
 	/* set CPEN (control port mode) and power down */
 	cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN);
 	cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
 	/* configure */
-	cs4398_write(chip, 2, CS4398_FM_SINGLE |
-		     CS4398_DEM_NONE | CS4398_DIF_LJUST);
+	cs4398_write(chip, 2, data->cs4398_fm);
 	cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L);
-	cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE);
-	cs4398_write(chip, 5, 0xfe);
-	cs4398_write(chip, 6, 0xfe);
 	cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP |
 		     CS4398_ZERO_CROSS | CS4398_SOFT_RAMP);
 	cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST);
@@ -249,21 +303,35 @@ static void xonar_dx_init(struct oxygen *chip)
 		      CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP);
 	cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE);
 	cs4362a_write(chip, 0x05, 0);
-	cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE |
-		      CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
-	cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE |
-		      CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
-	cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE |
-		      CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
-	cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE);
-	cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE);
+	cs4362a_write(chip, 0x06, data->cs4362a_fm);
+	cs4362a_write(chip, 0x09, data->cs4362a_fm);
+	cs4362a_write(chip, 0x0c, data->cs4362a_fm);
+	update_cs43xx_volume(chip);
+	update_cs43xx_mute(chip);
 	/* clear power down */
 	cs4398_write(chip, 8, CS4398_CPEN);
 	cs4362a_write(chip, 0x01, CS4362A_CPEN);
+}
+
+static void xonar_dx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->anti_pop_delay = 800;
+	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+	data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
+	data->cs4362a_fm = CS4362A_FM_SINGLE |
+		CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+
+	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+		       OXYGEN_2WIRE_LENGTH_8 |
+		       OXYGEN_2WIRE_INTERRUPT_MASK |
+		       OXYGEN_2WIRE_SPEED_FAST);
+
+	cs43xx_init(chip);
 
 	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
 			  GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
@@ -291,37 +359,28 @@ static void xonar_dx_cleanup(struct oxygen *chip)
 	oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
 }
 
-static void set_pcm1796_params(struct oxygen *chip,
-			       struct snd_pcm_hw_params *params)
+static void xonar_d2_resume(struct oxygen *chip)
 {
-	unsigned int i;
-	u8 value;
-
-	value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
-	for (i = 0; i < 4; ++i)
-		pcm1796_write(chip, i, 20, value);
+	pcm1796_init(chip);
+	xonar_enable_output(chip);
 }
 
-static void update_pcm1796_volume(struct oxygen *chip)
+static void xonar_dx_resume(struct oxygen *chip)
 {
-	unsigned int i;
-
-	for (i = 0; i < 4; ++i) {
-		pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]);
-		pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]);
-	}
+	cs43xx_init(chip);
+	xonar_enable_output(chip);
 }
 
-static void update_pcm1796_mute(struct oxygen *chip)
+static void set_pcm1796_params(struct oxygen *chip,
+			       struct snd_pcm_hw_params *params)
 {
+	struct xonar_data *data = chip->model_data;
 	unsigned int i;
-	u8 value;
 
-	value = PCM1796_FMT_24_LJUST | PCM1796_ATLD;
-	if (chip->dac_mute)
-		value |= PCM1796_MUTE;
+	data->pcm1796_oversampling =
+		params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
 	for (i = 0; i < 4; ++i)
-		pcm1796_write(chip, i, 18, value);
+		pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
 }
 
 static void set_cs53x1_params(struct oxygen *chip,
@@ -342,55 +401,24 @@ static void set_cs53x1_params(struct oxygen *chip,
 static void set_cs43xx_params(struct oxygen *chip,
 			      struct snd_pcm_hw_params *params)
 {
-	u8 fm_cs4398, fm_cs4362a;
+	struct xonar_data *data = chip->model_data;
 
-	fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST;
-	fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+	data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST;
+	data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
 	if (params_rate(params) <= 50000) {
-		fm_cs4398 |= CS4398_FM_SINGLE;
-		fm_cs4362a |= CS4362A_FM_SINGLE;
+		data->cs4398_fm |= CS4398_FM_SINGLE;
+		data->cs4362a_fm |= CS4362A_FM_SINGLE;
 	} else if (params_rate(params) <= 100000) {
-		fm_cs4398 |= CS4398_FM_DOUBLE;
-		fm_cs4362a |= CS4362A_FM_DOUBLE;
+		data->cs4398_fm |= CS4398_FM_DOUBLE;
+		data->cs4362a_fm |= CS4362A_FM_DOUBLE;
 	} else {
-		fm_cs4398 |= CS4398_FM_QUAD;
-		fm_cs4362a |= CS4362A_FM_QUAD;
+		data->cs4398_fm |= CS4398_FM_QUAD;
+		data->cs4362a_fm |= CS4362A_FM_QUAD;
 	}
-	cs4398_write(chip, 2, fm_cs4398);
-	cs4362a_write(chip, 0x06, fm_cs4362a);
-	cs4362a_write(chip, 0x09, fm_cs4362a);
-	cs4362a_write(chip, 0x0c, fm_cs4362a);
-}
-
-static void update_cs4362a_volumes(struct oxygen *chip)
-{
-	u8 mute;
-
-	mute = chip->dac_mute ? CS4362A_MUTE : 0;
-	cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute);
-	cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute);
-	cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute);
-	cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute);
-	cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute);
-	cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute);
-}
-
-static void update_cs43xx_volume(struct oxygen *chip)
-{
-	cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2);
-	cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2);
-	update_cs4362a_volumes(chip);
-}
-
-static void update_cs43xx_mute(struct oxygen *chip)
-{
-	u8 reg;
-
-	reg = CS4398_MUTEP_LOW | CS4398_PAMUTE;
-	if (chip->dac_mute)
-		reg |= CS4398_MUTE_B | CS4398_MUTE_A;
-	cs4398_write(chip, 4, reg);
-	update_cs4362a_volumes(chip);
+	cs4398_write(chip, 2, data->cs4398_fm);
+	cs4362a_write(chip, 0x06, data->cs4362a_fm);
+	cs4362a_write(chip, 0x09, data->cs4362a_fm);
+	cs4362a_write(chip, 0x0c, data->cs4362a_fm);
 }
 
 static void xonar_gpio_changed(struct oxygen *chip)
@@ -535,6 +563,8 @@ static const struct oxygen_model xonar_models[] = {
 		.control_filter = xonar_d2_control_filter,
 		.mixer_init = xonar_mixer_init,
 		.cleanup = xonar_cleanup,
+		.suspend = xonar_cleanup,
+		.resume = xonar_d2_resume,
 		.set_dac_params = set_pcm1796_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_pcm1796_volume,
@@ -563,6 +593,8 @@ static const struct oxygen_model xonar_models[] = {
 		.control_filter = xonar_d2_control_filter,
 		.mixer_init = xonar_mixer_init,
 		.cleanup = xonar_cleanup,
+		.suspend = xonar_cleanup,
+		.resume = xonar_d2_resume,
 		.set_dac_params = set_pcm1796_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_pcm1796_volume,
@@ -592,6 +624,8 @@ static const struct oxygen_model xonar_models[] = {
 		.control_filter = xonar_dx_control_filter,
 		.mixer_init = xonar_dx_mixer_init,
 		.cleanup = xonar_dx_cleanup,
+		.suspend = xonar_dx_cleanup,
+		.resume = xonar_dx_resume,
 		.set_dac_params = set_cs43xx_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_cs43xx_volume,
@@ -636,6 +670,10 @@ static struct pci_driver xonar_driver = {
 	.id_table = xonar_ids,
 	.probe = xonar_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+	.suspend = oxygen_pci_suspend,
+	.resume = oxygen_pci_resume,
+#endif
 };
 
 static int __init alsa_card_xonar_init(void)

sound/pci/pcxhr/pcxhr.c

@@ -516,7 +516,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 	int capture_mask = 0;
 	int playback_mask = 0;
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	struct timeval my_tv1, my_tv2;
 	do_gettimeofday(&my_tv1);
 #endif
@@ -623,7 +623,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
 
 	mutex_unlock(&mgr->setup_mutex);
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	do_gettimeofday(&my_tv2);
 	snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n",
 		    (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);

sound/pci/pcxhr/pcxhr_core.c

@@ -473,7 +473,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = {
 [CMD_AUDIO_LEVEL_ADJUST] =		{ 0xc22000, 0, RMH_SSIZE_FIXED },
 };
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 static char* cmd_names[] = {
 [CMD_VERSION] =				"CMD_VERSION",
 [CMD_SUPPORTED] =			"CMD_SUPPORTED",
@@ -549,7 +549,7 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 				}
 			}
 		}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (rmh->cmd_idx < CMD_LAST_INDEX)
 			snd_printdd("    stat[%d]=%x\n", i, data);
 #endif
@@ -597,7 +597,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		data |= 0x008000;	/* MASK_MORE_THAN_1_WORD_COMMAND */
 	else
 		data &= 0xff7fff;	/* MASK_1_WORD_COMMAND */
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	if (rmh->cmd_idx < CMD_LAST_INDEX)
 		snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]);
 #endif
@@ -624,7 +624,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
 		for (i=1; i < rmh->cmd_len; i++) {
 			/* send other words */
 			data = rmh->cmd[i];
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 			if (rmh->cmd_idx < CMD_LAST_INDEX)
 				snd_printdd("    cmd[%d]=%x\n", i, data);
 #endif
@@ -847,7 +847,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
 	int state, i, err;
 	int audio_mask;
 
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	struct timeval my_tv1, my_tv2;
 	do_gettimeofday(&my_tv1);
 #endif
@@ -894,7 +894,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
 		if (err)
 			return err;
 	}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	do_gettimeofday(&my_tv2);
 	snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n",
 		    (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
@@ -951,7 +951,7 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err,
 				  enum pcxhr_async_err_src err_src, int pipe,
 				  int is_capture)
 {
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	static char* err_src_name[] = {
 		[PCXHR_ERR_PIPE]	= "Pipe",
 		[PCXHR_ERR_STREAM]	= "Stream",
@@ -1169,7 +1169,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 				    mgr->dsp_time_last, dsp_time_new);
 			mgr->dsp_time_err++;
 		}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (dsp_time_diff == 0)
 			snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new);
 		else if (dsp_time_diff >= (2*PCXHR_GRANULARITY))
@@ -1208,7 +1208,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 		mgr->src_it_dsp = reg;
 		tasklet_hi_schedule(&mgr->msg_taskq);
 	}
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	if (reg & PCXHR_FATAL_DSP_ERR)
 		snd_printdd("FATAL DSP ERROR : %x\n", reg);
 #endif

sound/pci/trident/trident_main.c

@@ -1590,7 +1590,10 @@ static int snd_trident_trigger(struct snd_pcm_substream *substream,
 	if (spdif_flag) {
 		if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
 			outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS));
-			outb(trident->spdif_pcm_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+			val = trident->spdif_pcm_ctrl;
+			if (!go)
+				val &= ~(0x28);
+			outb(val, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
 		} else {
 			outl(trident->spdif_pcm_bits, TRID_REG(trident, SI_SPDIF_CS));
 			val = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) | SPDIF_EN;

sound/pci/trident/trident_memory.c

@@ -310,181 +310,3 @@ int snd_trident_free_pages(struct snd_trident *trident,
 	mutex_unlock(&hdr->block_mutex);
 	return 0;
 }
-
-
-/*----------------------------------------------------------------
- * memory allocation using multiple pages (for synth)
- *----------------------------------------------------------------
- * Unlike the DMA allocation above, non-contiguous pages are
- * assigned to TLB.
- *----------------------------------------------------------------*/
-
-/*
- */
-static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk);
-static int synth_free_pages(struct snd_trident *hw, struct snd_util_memblk *blk);
-
-/*
- * allocate a synth sample area
- */
-struct snd_util_memblk *
-snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size)
-{
-	struct snd_util_memblk *blk;
-	struct snd_util_memhdr *hdr = hw->tlb.memhdr; 
-
-	mutex_lock(&hdr->block_mutex);
-	blk = __snd_util_mem_alloc(hdr, size);
-	if (blk == NULL) {
-		mutex_unlock(&hdr->block_mutex);
-		return NULL;
-	}
-	if (synth_alloc_pages(hw, blk)) {
-		__snd_util_mem_free(hdr, blk);
-		mutex_unlock(&hdr->block_mutex);
-		return NULL;
-	}
-	mutex_unlock(&hdr->block_mutex);
-	return blk;
-}
-
-EXPORT_SYMBOL(snd_trident_synth_alloc);
-
-/*
- * free a synth sample area
- */
-int
-snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk)
-{
-	struct snd_util_memhdr *hdr = hw->tlb.memhdr; 
-
-	mutex_lock(&hdr->block_mutex);
-	synth_free_pages(hw, blk);
-	 __snd_util_mem_free(hdr, blk);
-	mutex_unlock(&hdr->block_mutex);
-	return 0;
-}
-
-EXPORT_SYMBOL(snd_trident_synth_free);
-
-/*
- * reset TLB entry and free kernel page
- */
-static void clear_tlb(struct snd_trident *trident, int page)
-{
-	void *ptr = page_to_ptr(trident, page);
-	dma_addr_t addr = page_to_addr(trident, page);
-	set_silent_tlb(trident, page);
-	if (ptr) {
-		struct snd_dma_buffer dmab;
-		dmab.dev.type = SNDRV_DMA_TYPE_DEV;
-		dmab.dev.dev = snd_dma_pci_data(trident->pci);
-		dmab.area = ptr;
-		dmab.addr = addr;
-		dmab.bytes = ALIGN_PAGE_SIZE;
-		snd_dma_free_pages(&dmab);
-	}
-}
-
-/* check new allocation range */
-static void get_single_page_range(struct snd_util_memhdr *hdr,
-				  struct snd_util_memblk *blk,
-				  int *first_page_ret, int *last_page_ret)
-{
-	struct list_head *p;
-	struct snd_util_memblk *q;
-	int first_page, last_page;
-	first_page = firstpg(blk);
-	if ((p = blk->list.prev) != &hdr->block) {
-		q = list_entry(p, struct snd_util_memblk, list);
-		if (lastpg(q) == first_page)
-			first_page++;  /* first page was already allocated */
-	}
-	last_page = lastpg(blk);
-	if ((p = blk->list.next) != &hdr->block) {
-		q = list_entry(p, struct snd_util_memblk, list);
-		if (firstpg(q) == last_page)
-			last_page--; /* last page was already allocated */
-	}
-	*first_page_ret = first_page;
-	*last_page_ret = last_page;
-}
-
-/*
- * allocate kernel pages and assign them to TLB
- */
-static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk)
-{
-	int page, first_page, last_page;
-	struct snd_dma_buffer dmab;
-
-	firstpg(blk) = get_aligned_page(blk->offset);
-	lastpg(blk) = get_aligned_page(blk->offset + blk->size - 1);
-	get_single_page_range(hw->tlb.memhdr, blk, &first_page, &last_page);
-
-	/* allocate a kernel page for each Trident page -
-	 * fortunately Trident page size and kernel PAGE_SIZE is identical!
-	 */
-	for (page = first_page; page <= last_page; page++) {
-		if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(hw->pci),
-					ALIGN_PAGE_SIZE, &dmab) < 0)
-			goto __fail;
-		if (! is_valid_page(dmab.addr)) {
-			snd_dma_free_pages(&dmab);
-			goto __fail;
-		}
-		set_tlb_bus(hw, page, (unsigned long)dmab.area, dmab.addr);
-	}
-	return 0;
-
-__fail:
-	/* release allocated pages */
-	last_page = page - 1;
-	for (page = first_page; page <= last_page; page++)
-		clear_tlb(hw, page);
-
-	return -ENOMEM;
-}
-
-/*
- * free pages
- */
-static int synth_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk)
-{
-	int page, first_page, last_page;
-
-	get_single_page_range(trident->tlb.memhdr, blk, &first_page, &last_page);
-	for (page = first_page; page <= last_page; page++)
-		clear_tlb(trident, page);
-
-	return 0;
-}
-
-/*
- * copy_from_user(blk + offset, data, size)
- */
-int snd_trident_synth_copy_from_user(struct snd_trident *trident,
-				     struct snd_util_memblk *blk,
-				     int offset, const char __user *data, int size)
-{
-	int page, nextofs, end_offset, temp, temp1;
-
-	offset += blk->offset;
-	end_offset = offset + size;
-	page = get_aligned_page(offset) + 1;
-	do {
-		nextofs = aligned_page_offset(page);
-		temp = nextofs - offset;
-		temp1 = end_offset - offset;
-		if (temp1 < temp)
-			temp = temp1;
-		if (copy_from_user(offset_ptr(trident, offset), data, temp))
-			return -EFAULT;
-		offset = nextofs;
-		data += temp;
-		page++;
-	} while (offset < end_offset);
-	return 0;
-}
-
-EXPORT_SYMBOL(snd_trident_synth_copy_from_user);

sound/pci/via82xx.c

@@ -1756,6 +1756,12 @@ static struct ac97_quirk ac97_quirks[] = {
 		.name = "ECS L7VMM2",
 		.type = AC97_TUNE_HP_ONLY
 	},
+	{
+		.subvendor = 0x1019,
+		.subdevice = 0x1841,
+		.name = "ECS K7VTA3",
+		.type = AC97_TUNE_HP_ONLY
+	},
 	{
 		.subvendor = 0x1849,
 		.subdevice = 0x3059,

sound/pci/ymfpci/ymfpci_main.c

@@ -2205,6 +2205,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip)
 	for (reg = 0x80; reg < 0xc0; reg += 4)
 		snd_ymfpci_writel(chip, reg, 0);
 	snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x3fff3fff);
+	snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0x3fff3fff);
 	snd_ymfpci_writel(chip, YDSXGR_ZVOUTVOL, 0x3fff3fff);
 	snd_ymfpci_writel(chip, YDSXGR_SPDIFOUTVOL, 0x3fff3fff);
 	snd_ymfpci_writel(chip, YDSXGR_NATIVEADCINVOL, 0x3fff3fff);
@@ -2324,6 +2325,7 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
 		chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]);
 	chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE);
 	snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0);
+	snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0);
 	snd_ymfpci_disable_dsp(chip);
 	pci_disable_device(pci);
 	pci_save_state(pci);

sound/pcmcia/Kconfig

@@ -1,11 +1,16 @@
 # ALSA PCMCIA drivers
 
-menu "PCMCIA devices"
-	depends on SND!=n && PCMCIA
+menuconfig SND_PCMCIA
+	bool "PCMCIA sound devices"
+	depends on PCMCIA
+	default y
+	help
+	  Support for sound devices connected via the PCMCIA bus.
+
+if SND_PCMCIA && PCMCIA
 
 config SND_VXPOCKET
 	tristate "Digigram VXpocket"
-	depends on SND && PCMCIA
 	select SND_VX_LIB
 	help
 	  Say Y here to include support for Digigram VXpocket and
@@ -16,7 +21,6 @@ config SND_VXPOCKET
 
 config SND_PDAUDIOCF
 	tristate "Sound Core PDAudioCF"
-	depends on SND && PCMCIA
 	select SND_PCM
 	help
 	  Say Y here to include support for Sound Core PDAudioCF
@@ -25,4 +29,5 @@ config SND_PDAUDIOCF
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-pdaudiocf.
 
-endmenu
+endif	# SND_PCMCIA
+

sound/pcmcia/vx/vxp_ops.c

@@ -151,7 +151,7 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware *
 	unsigned int i;
 	int c;
 	int regCSUER, regRUER;
-	unsigned char *image;
+	const unsigned char *image;
 	unsigned char data;
 
 	/* Switch to programmation mode */

sound/ppc/Kconfig

@@ -1,17 +1,17 @@
 # ALSA PowerMac drivers
 
-menu "ALSA PowerMac devices"
-	depends on SND!=n && PPC
-
-comment "ALSA PowerMac requires I2C"
-	depends on SND && I2C=n
+menuconfig SND_PPC
+	bool "PowerPC sound devices"
+	depends on PPC64 || PPC32
+	default y
+	help
+	  Support for sound devices specific to PowerPC architectures.
 
-comment "ALSA PowerMac requires INPUT"
-	depends on SND && INPUT=n
+if SND_PPC
 
 config SND_POWERMAC
 	tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)"
-	depends on SND && I2C && INPUT && PPC_PMAC
+	depends on I2C && INPUT && PPC_PMAC
 	select SND_PCM
 	help
 	  Say Y here to include support for the integrated sound device.
@@ -32,14 +32,9 @@ config SND_POWERMAC_AUTO_DRC
 	  Note that you can turn on/off DRC manually even without this
 	  option.
 
-endmenu
-
-menu "ALSA PowerPC devices"
-	depends on SND!=n && ( PPC64 || PPC32 )
-
 config SND_PS3
 	tristate "PS3 Audio support"
-	depends on SND && PS3_PS3AV
+	depends on PS3_PS3AV
 	select SND_PCM
 	default m
 	help
@@ -52,4 +47,5 @@ config SND_PS3_DEFAULT_START_DELAY
 	int "Startup delay time in ms"
 	depends on SND_PS3
 	default "2000"
-endmenu
+
+endif	# SND_PPC

sound/ppc/daca.c

@@ -249,9 +249,7 @@ int __init snd_pmac_daca_init(struct snd_pmac *chip)
 	int i, err;
 	struct pmac_daca *mix;
 
-#ifdef CONFIG_KMOD
 	request_module("i2c-powermac");
-#endif /* CONFIG_KMOD */
 
 	mix = kzalloc(sizeof(*mix), GFP_KERNEL);
 	if (! mix)

sound/ppc/tumbler.c

@@ -1350,9 +1350,7 @@ int __init snd_pmac_tumbler_init(struct snd_pmac *chip)
 	struct device_node *tas_node, *np;
 	char *chipname;
 
-#ifdef CONFIG_KMOD
 	request_module("i2c-powermac");
-#endif /* CONFIG_KMOD */
 
 	mix = kzalloc(sizeof(*mix), GFP_KERNEL);
 	if (! mix)

sound/sh/Kconfig

@@ -1,14 +1,22 @@
 # ALSA SH drivers
 
-menu "SUPERH devices"
-	depends on SND!=n && SUPERH
+menuconfig SND_SUPERH
+	bool "SUPERH sound devices"
+	depends on SUPERH
+	default y
+	help
+	  Support for sound devices specific to SUPERH architectures.
+	  Drivers that are implemented on ASoC can be found in
+	  "ALSA for SoC audio support" section.
+
+if SND_SUPERH
 
 config SND_AICA
 	tristate "Dreamcast Yamaha AICA sound"
-	depends on SH_DREAMCAST && SND
+	depends on SH_DREAMCAST
 	select SND_PCM
 	help
 	  ALSA Sound driver for the SEGA Dreamcast console.
 
-endmenu
+endif	# SND_SUPERH