Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: Add missing comma to SND_SOC_DAPM_SWITCH_E in soc-dapm.h
  ALSA: hda: Add STAC92HD83XXX_PWR_REF quirk
  ALSA: hda: revert change to 92hd83xxx power mapping
  ALSA: hda - Add model entry for HP dv4
  ALSA: hda: 83xxx port 0xe DAC selection
  ASoC: fix registration of the SoC card in the Freescale MPC8610 drivers
  sound: virtuoso: document HDAV1.3 driver status
  sound: virtuoso: add newline
  sound: virtuoso: enable UART on Xonar HDAV1.3
  sound: Remove removed OSS kernel parameters from doc
  ALSA: hda: fix invalid power mapping masks
  ASoC: atmel_pcm: Remove non-existant header
  ALSA: hda - add quirks for some 82801H variants to use ALC883_MITAC
  ALSA: hda - Fix (yet more) STAC925x issues

9 files changed, 48 insertions, 45 deletions

diff --git a/Documentation/kernel-parameters.txt b/Documentation/kernel-parameters.txt
index 8511d3532c2..d8362cf9909 100644
--- a/Documentation/kernel-parameters.txt
+++ b/Documentation/kernel-parameters.txt
@@ -577,9 +577,6 @@ and is between 256 and 4096 characters. It is defined in the file
 			a memory unit (amount[KMG]). See also
 			Documentation/kdump/kdump.txt for a example.
 
-	cs4232=		[HW,OSS]
-			Format: <io>,<irq>,<dma>,<dma2>,<mpuio>,<mpuirq>
-
 	cs89x0_dma=	[HW,NET]
 			Format: <dma>
 
@@ -732,10 +729,6 @@ and is between 256 and 4096 characters. It is defined in the file
 			Default value is 0.
 			Value can be changed at runtime via /selinux/enforce.
 
-	es1371=		[HW,OSS]
-			Format: <spdif>,[<nomix>,[<amplifier>]]
-			See also header of sound/oss/es1371.c.
-
 	ether=		[HW,NET] Ethernet cards parameters
 			This option is obsoleted by the "netdev=" option, which
 			has equivalent usage. See its documentation for details.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 64eb1100eec..0f5d26bea80 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -349,6 +349,7 @@ STAC92HD73*
 STAC92HD83*
 ===========
   ref		Reference board
+  mic-ref	Reference board with power managment for ports
 
 STAC9872
 ========
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 93a4edb148b..dfa80495882 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -108,7 +108,7 @@
 #define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
 {	.id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1 \
+	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 6e3a1848447..82b9bddcdcd 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -744,8 +744,8 @@ config SND_VIRTUOSO
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
-	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and
-	  HDAV1.3 (Deluxe).
+	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
+	  Support for the HDAV1.3 (Deluxe) is very experimental.
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-virtuoso.
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 82dd0843197..5d249a547fb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8478,6 +8478,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
 	SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
@@ -8526,6 +8527,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	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, 0x2503, "82801H", ALC883_MITAC),
 	SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
 	{}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c39deebb588..3dd4eee70b7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -81,6 +81,7 @@ enum {
 
 enum {
 	STAC_92HD83XXX_REF,
+	STAC_92HD83XXX_PWR_REF,
 	STAC_92HD83XXX_MODELS
 };
 
@@ -334,7 +335,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
 };
 
 static unsigned int stac92hd83xxx_pwr_mapping[4] = {
-	0x03, 0x0c, 0x10, 0x40,
+	0x03, 0x0c, 0x20, 0x40,
 };
 
 static hda_nid_t stac92hd83xxx_amp_nids[1] = {
@@ -841,10 +842,6 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 };
 
 static struct hda_verb stac92hd83xxx_core_init[] = {
-	/* start of config #1 */
-	{ 0xe, AC_VERB_SET_CONNECT_SEL, 0x3},
-
-	/* start of config #2 */
 	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
 	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
 	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
@@ -885,8 +882,8 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 static struct hda_verb stac925x_core_init[] = {
 	/* set dac0mux for dac converter */
 	{ 0x06, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* unmute and set max the selector */
-	{ 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f },
+	/* mute the master volume */
+	{ 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{}
 };
 
@@ -1138,6 +1135,8 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac925x_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
 	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
@@ -1736,10 +1735,12 @@ static unsigned int ref92hd83xxx_pin_configs[14] = {
 
 static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
+	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
 };
 
 static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = "ref",
+	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
 };
 
 static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1799,6 +1800,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP dv5", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
 		      "HP dv7", STAC_HP_M4),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
+		      "HP dv4", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
 		      "HP dv7", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603,
@@ -3573,13 +3576,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		err = stac92xx_auto_fill_dac_nids(codec);
 		if (err < 0)
 			return err;
+		err = stac92xx_auto_create_multi_out_ctls(codec,
+							  &spec->autocfg);
+		if (err < 0)
+			return err;
 	}
 
-	err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg);
-
-	if (err < 0)
-		return err;
-
 	/* setup analog beep controls */
 	if (spec->anabeep_nid > 0) {
 		err = stac92xx_auto_create_beep_ctls(codec,
@@ -4753,7 +4755,9 @@ static struct hda_input_mux stac92hd83xxx_dmux = {
 static int patch_stac92hd83xxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
 	int err;
+	int num_dacs;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4772,15 +4776,16 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
 	spec->multiout.dac_nids = spec->dac_nids;
 
-	spec->init = stac92hd83xxx_core_init;
-	switch (codec->vendor_id) {
-	case 0x111d7605:
-		break;
-	default:
-		spec->num_pwrs--;
-		spec->init++; /* switch to config #2 */
-	}
 
+	/* set port 0xe to select the last DAC
+	 */
+	num_dacs = snd_hda_get_connections(codec, 0x0e,
+		conn, STAC92HD83_DAC_COUNT + 1) - 1;
+
+	snd_hda_codec_write_cache(codec, 0xe, 0,
+		AC_VERB_SET_CONNECT_SEL, num_dacs);
+
+	spec->init = stac92hd83xxx_core_init;
 	spec->mixer = stac92hd83xxx_mixer;
 	spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
@@ -4806,6 +4811,15 @@ again:
 		return err;
 	}
 
+	switch (codec->vendor_id) {
+	case 0x111d7604:
+	case 0x111d7605:
+		if (spec->board_config == STAC_92HD83XXX_PWR_REF)
+			break;
+		spec->num_pwrs = 0;
+		break;
+	}
+
 	err = stac92xx_parse_auto_config(codec, 0x1d, 0);
 	if (!err) {
 		if (spec->board_config < 0) {
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index e9e829e83d7..18c7c91786b 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -683,7 +683,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip)
 	if (chip->uart_input_count >= 2 &&
 	    chip->uart_input[chip->uart_input_count - 2] == 'O' &&
 	    chip->uart_input[chip->uart_input_count - 1] == 'K') {
-		printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:");
+		printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n");
 		print_hex_dump_bytes("", DUMP_PREFIX_OFFSET,
 				     chip->uart_input, chip->uart_input_count);
 		chip->uart_input_count = 0;
@@ -908,6 +908,7 @@ static const struct oxygen_model model_xonar_hdav = {
 	.dac_channels = 8,
 	.dac_volume_min = 0x0f,
 	.dac_volume_max = 0xff,
+	.misc_flags = OXYGEN_MISC_MIDI,
 	.function_flags = OXYGEN_FUNCTION_2WIRE,
 	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 1fac5efd285..3dcdc4e3cfa 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -44,8 +44,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <mach/hardware.h>
-
 #include "atmel-pcm.h"
 
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index bcec3f60bad..acf39a646b2 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 };
 
 /**
- * mpc8610_hpcd_machine: ASoC machine data
- */
-static struct snd_soc_card mpc8610_hpcd_machine = {
-	.probe = mpc8610_hpcd_machine_probe,
-	.remove = mpc8610_hpcd_machine_remove,
-	.name = "MPC8610 HPCD",
-	.num_links = 1,
-};
-
-/**
  * mpc8610_hpcd_probe: OF probe function for the fabric driver
  *
  * This function gets called when an SSI node is found in the device tree.
@@ -455,7 +445,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 	machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */
 	machine_data->dai.ops = &mpc8610_hpcd_ops;
 
-	mpc8610_hpcd_machine.dai_link = &machine_data->dai;
+	machine_data->machine.probe = mpc8610_hpcd_machine_probe;
+	machine_data->machine.remove = mpc8610_hpcd_machine_remove;
+	machine_data->machine.name = "MPC8610 HPCD";
+	machine_data->machine.num_links = 1;
+	machine_data->machine.dai_link = &machine_data->dai;
 
 	/* Allocate a new audio platform device structure */
 	sound_device = platform_device_alloc("soc-audio", -1);
@@ -465,7 +459,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 		goto error;
 	}
 
-	machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
+	machine_data->sound_devdata.card = &machine_data->machine;
 	machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
 	machine_data->machine.platform = &fsl_soc_platform;