From b0af0de5cb57c96b0c3d739005172152b7de0ce8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 21 Jun 2005 14:49:19 +0200
Subject: [ALSA] hda-codec - Fix oops with ALC880

HDA Codec driver
- Fixed oops with ALC880 auto-config mode
- Fixed a wrong config table entry for ALC880

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bab89843d85..ed16ce817dd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -119,6 +119,7 @@ struct alc_spec {
 	unsigned int num_kctl_alloc, num_kctl_used;
 	snd_kcontrol_new_t *kctl_alloc;
 	struct hda_input_mux private_imux;
+	hda_nid_t private_dac_nids[4];
 };
 
 
@@ -1549,7 +1550,8 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 	{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG },
 	{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG },
 	{ .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG },
-	{ .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG },
+	/* note subvendor = 0 below */
+	/* { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG }, */
 
 	{ .modelname = "w810", .config = ALC880_W810 },
 	{ .pci_subvendor = 0x161f, .pci_subdevice = 0x203d, .config = ALC880_W810 },
@@ -1656,7 +1658,8 @@ static struct alc_config_preset alc880_presets[] = {
 	},
 	[ALC880_W810] = {
 		.mixers = { alc880_w810_base_mixer },
-		.init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs },
+		.init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs,
+				alc880_gpio2_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
 		.dac_nids = alc880_w810_dac_nids,
 		.dig_out_nid = ALC880_DIGOUT_NID,
@@ -1666,8 +1669,7 @@ static struct alc_config_preset alc880_presets[] = {
 	},
 	[ALC880_Z71V] = {
 		.mixers = { alc880_z71v_mixer },
-		.init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs,
-				alc880_gpio2_init_verbs },
+		.init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
 		.dac_nids = alc880_z71v_dac_nids,
 		.dig_out_nid = ALC880_DIGOUT_NID,
@@ -1809,6 +1811,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pi
 	int i, j;
 
 	memset(assigned, 0, sizeof(assigned));
+	spec->multiout.dac_nids = spec->private_dac_nids;
 
 	/* check the pins hardwired to audio widget */
 	for (i = 0; i < cfg->line_outs; i++) {
-- 
cgit v1.2.3


From c7d4b2fa3169a1206450bc445d1997a17479644f Mon Sep 17 00:00:00 2001
From: Matt <matt@embeddedalley.com>
Date: Mon, 27 Jun 2005 14:59:41 +0200
Subject: [ALSA] hda-codec - SigmaTel HDA multichannel support

HDA Codec driver
Adds 6/8 channel support to the SigmaTel HDA patch. Please apply.

Signed-off-by: Matt <matt@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 733 ++++++++++++++++++++++++++++-------------
 1 file changed, 496 insertions(+), 237 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 013be2ea513..fad825677e7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -30,32 +30,33 @@
 #include <linux/slab.h>
 #include <linux/pci.h>
 #include <sound/core.h>
+#include <sound/asoundef.h>
 #include "hda_codec.h"
 #include "hda_local.h"
 
 #undef STAC_TEST
 
 struct sigmatel_spec {
+	snd_kcontrol_new_t *mixers[4];
+	unsigned int num_mixers;
+
+	unsigned int surr_switch: 1;
+
 	/* playback */
 	struct hda_multi_out multiout;
-	hda_nid_t playback_nid;
+	hda_nid_t dac_nids[4];
 
 	/* capture */
 	hda_nid_t *adc_nids;
 	unsigned int num_adcs;
 	hda_nid_t *mux_nids;
 	unsigned int num_muxes;
-	hda_nid_t capture_nid;
 	hda_nid_t dig_in_nid;
 
-	/* power management*/
-	hda_nid_t *pstate_nids;
-	unsigned int num_pstates;
-
+#ifdef STAC_TEST
 	/* pin widgets */
 	hda_nid_t *pin_nids;
 	unsigned int num_pins;
-#ifdef STAC_TEST
 	unsigned int *pin_configs;
 #endif
 
@@ -64,16 +65,20 @@ struct sigmatel_spec {
 	snd_kcontrol_new_t *mixer;
 
 	/* capture source */
-	struct hda_input_mux input_mux;
-	char input_labels[HDA_MAX_NUM_INPUTS][16];
+	struct hda_input_mux *input_mux;
 	unsigned int cur_mux[2];
 
 	/* channel mode */
 	unsigned int num_ch_modes;
 	unsigned int cur_ch_mode;
-	const struct sigmatel_channel_mode *channel_modes;
 
-	struct hda_pcm pcm_rec[1];	/* PCM information */
+	struct hda_pcm pcm_rec[2];	/* PCM information */
+
+	/* dynamic controls and input_mux */
+	struct auto_pin_cfg autocfg;
+	unsigned int num_kctl_alloc, num_kctl_used;
+	snd_kcontrol_new_t *kctl_alloc;
+	struct hda_input_mux private_imux;
 };
 
 static hda_nid_t stac9200_adc_nids[1] = {
@@ -88,14 +93,6 @@ static hda_nid_t stac9200_dac_nids[1] = {
         0x02,
 };
 
-static hda_nid_t stac9200_pstate_nids[3] = {
-	0x01, 0x02, 0x03,
-};
-
-static hda_nid_t stac9200_pin_nids[8] = {
-	0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12,
-};
-
 static hda_nid_t stac922x_adc_nids[2] = {
         0x06, 0x07,
 };
@@ -104,24 +101,22 @@ static hda_nid_t stac922x_mux_nids[2] = {
         0x12, 0x13,
 };
 
-static hda_nid_t stac922x_dac_nids[4] = {
-        0x02, 0x03, 0x04, 0x05,
-};
-
-static hda_nid_t stac922x_pstate_nids[8] = {
-	0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x11,
+#ifdef STAC_TEST
+static hda_nid_t stac9200_pin_nids[8] = {
+	0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12,
 };
 
 static hda_nid_t stac922x_pin_nids[10] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x10, 0x11, 0x15, 0x1b,
 };
+#endif
 
 static int stac92xx_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct sigmatel_spec *spec = codec->spec;
-	return snd_hda_input_mux_info(&spec->input_mux, uinfo);
+	return snd_hda_input_mux_info(spec->input_mux, uinfo);
 }
 
 static int stac92xx_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
@@ -140,26 +135,64 @@ static int stac92xx_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t
 	struct sigmatel_spec *spec = codec->spec;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
 
-	return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol,
+	return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
 				     spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);
 }
 
-static struct hda_verb stac9200_ch2_init[] = {
+static struct hda_verb stac9200_core_init[] = {
 	/* set dac0mux for dac converter */
-	{ 0x07, 0x701, 0x00},
+	{ 0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{}
 };
 
-static struct hda_verb stac922x_ch2_init[] = {
+static struct hda_verb stac922x_core_init[] = {
 	/* set master volume and direct control */	
-	{ 0x16, 0x70f, 0xff},
+	{ 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	{}
 };
 
-struct sigmatel_channel_mode {
-	unsigned int channels;
-	const struct hda_verb *sequence;
-};
+static int stac922x_channel_modes[3] = {2, 6, 8};
+
+static int stac922x_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = spec->num_ch_modes;
+	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+	sprintf(uinfo->value.enumerated.name, "%dch",
+		stac922x_channel_modes[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int stac922x_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_ch_mode;
+	return 0;
+}
+
+static int stac922x_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (ucontrol->value.enumerated.item[0] >= spec->num_ch_modes)
+		ucontrol->value.enumerated.item[0] = spec->num_ch_modes;
+	if (ucontrol->value.enumerated.item[0] == spec->cur_ch_mode &&
+	    ! codec->in_resume)
+		return 0;
+
+	spec->cur_ch_mode = ucontrol->value.enumerated.item[0];
+	spec->multiout.max_channels = stac922x_channel_modes[spec->cur_ch_mode];
+
+	return 1;
+}
 
 static snd_kcontrol_new_t stac9200_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
@@ -174,13 +207,12 @@ static snd_kcontrol_new_t stac9200_mixer[] = {
 	},
 	HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Input Mux Volume", 0x0c, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
+/* This needs to be generated dynamically based on sequence */
 static snd_kcontrol_new_t stac922x_mixer[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x2, 0x0, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Input Source",
@@ -195,14 +227,38 @@ static snd_kcontrol_new_t stac922x_mixer[] = {
 	{ } /* end */
 };
 
+static snd_kcontrol_new_t stac922x_ch_mode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = stac922x_ch_mode_info,
+		.get = stac922x_ch_mode_get,
+		.put = stac922x_ch_mode_put,
+	},
+	{ } /* end */
+};
+
 static int stac92xx_build_controls(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	int err;
+	int i;
 
 	err = snd_hda_add_new_ctls(codec, spec->mixer);
 	if (err < 0)
 		return err;
+
+	for (i = 0; i < spec->num_mixers; i++) {
+		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->surr_switch) {
+		err = snd_hda_add_new_ctls(codec, stac922x_ch_mode_mixer);
+		if (err < 0)
+			return err;
+	}
 	if (spec->multiout.dig_out_nid) {
 		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
 		if (err < 0)
@@ -222,9 +278,9 @@ static unsigned int stac9200_pin_configs[8] = {
 	0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
 };
 
-static unsigned int stac922x_pin_configs[14] = {
-	0x40000100, 0x40000100, 0x40000100, 0x01114010,
-	0x01813122, 0x40000100, 0x01447010, 0x01c47010,
+static unsigned int stac922x_pin_configs[10] = {
+	0x01014010, 0x01014011, 0x01014012, 0x0221401f,
+	0x01813122, 0x01014014, 0x01441030, 0x01c41030,
 	0x40000100, 0x40000100,
 };
 
@@ -255,180 +311,66 @@ static void stac92xx_set_config_regs(struct hda_codec *codec)
 }
 #endif
 
-static int stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int value)
-{
-	unsigned int pin_ctl;
-
-	pin_ctl = snd_hda_codec_read(codec, nid, 0,
-				     AC_VERB_GET_PIN_WIDGET_CONTROL,
-				     0x00);
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pin_ctl | value);
-
-	return 0;
-}
-
-static int stac92xx_set_vref(struct hda_codec *codec, hda_nid_t nid)
-{
-	unsigned int vref_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) >> AC_PINCAP_VREF_SHIFT;
-	unsigned int vref_ctl = AC_PINCTL_VREF_HIZ;
-
-	if (vref_caps & AC_PINCAP_VREF_100)
-		vref_ctl = AC_PINCTL_VREF_100;
-	else if (vref_caps & AC_PINCAP_VREF_80)
-		vref_ctl = AC_PINCTL_VREF_80;
-	else if (vref_caps & AC_PINCAP_VREF_50)
-		vref_ctl = AC_PINCTL_VREF_50;
-	else if (vref_caps & AC_PINCAP_VREF_GRD)
-		vref_ctl = AC_PINCTL_VREF_GRD;
-
-	stac92xx_set_pinctl(codec, nid, vref_ctl);
-	
-	return 0;
-}
-
 /*
- * retrieve the default device type from the default config value
+ * Analog playback callbacks
  */
-#define get_defcfg_type(cfg) ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
-#define get_defcfg_location(cfg) ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
-
-static int stac92xx_config_pin(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_cfg)
+static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      snd_pcm_substream_t *substream)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	u32 location = get_defcfg_location(pin_cfg);
-	char *label;
-	const char *type = NULL;
-	int ainput = 0;
-
-	switch(get_defcfg_type(pin_cfg)) {
-		case AC_JACK_HP_OUT:
-			/* Enable HP amp */
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_HP_EN);
-			/* Fall through */
-		case AC_JACK_SPDIF_OUT:
-		case AC_JACK_LINE_OUT:
-		case AC_JACK_SPEAKER:
-			/* Enable output */
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
-			break;
-		case AC_JACK_SPDIF_IN:
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
-			break;
-		case AC_JACK_MIC_IN:
-			if ((location & 0x0f) == AC_JACK_LOC_FRONT)
-				type = "Front Mic";
-			else
-				type = "Mic";
-			ainput = 1;
-			/* Set vref */
-			stac92xx_set_vref(codec, nid);
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
-			break;
-		case AC_JACK_CD:
-			type = "CD";
-			ainput = 1;
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
-			break;
-		case AC_JACK_LINE_IN:
-			if ((location & 0x0f) == AC_JACK_LOC_FRONT)
-				type = "Front Line";
-			else
-				type = "Line";
-			ainput = 1;
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
-			break;
-		case AC_JACK_AUX:
-			if ((location & 0x0f) == AC_JACK_LOC_FRONT)
-				type = "Front Aux";
-			else
-				type = "Aux";
-			ainput = 1;
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
-			break;
-	}
-
-	if (ainput) {
-		hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
-		int i, j, num_cons, index = -1;
-		if (!type)
-			type = "Input";
-		label = spec->input_labels[spec->input_mux.num_items];
-		strcpy(label, type);
-		spec->input_mux.items[spec->input_mux.num_items].label = label;
-		for (i=0; i<spec->num_muxes; i++) {
-			num_cons = snd_hda_get_connections(codec, spec->mux_nids[i], con_lst, HDA_MAX_NUM_INPUTS);
-			for (j=0; j<num_cons; j++)
-				if (con_lst[j] == nid) {
-					index = j;
-					break;
-				}
-			if (index >= 0)
-				break;
-		}
-		spec->input_mux.items[spec->input_mux.num_items].index = index;
-		spec->input_mux.num_items++;
-	}
-
-	return 0;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
 }
 
-static int stac92xx_config_pins(struct hda_codec *codec)
+/*
+ * set up the i/o for analog out
+ * when the digital out is available, copy the front out to digital out, too.
+ */
+static int stac92xx_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout,
+				     unsigned int stream_tag,
+				     unsigned int format,
+				     snd_pcm_substream_t *substream)
 {
-	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t *nids = mout->dac_nids;
+	int chs = substream->runtime->channels;
 	int i;
-	unsigned int pin_cfg;
-
-	for (i=0; i < spec->num_pins; i++) {
-		/* Default to disabled */
-		snd_hda_codec_write(codec, spec->pin_nids[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    0x00);
-
-		pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0,
-					     AC_VERB_GET_CONFIG_DEFAULT,
-					     0x00);
-		if (((pin_cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) == AC_JACK_PORT_NONE)
-			continue;	/* Move on */
 
-		stac92xx_config_pin(codec, spec->pin_nids[i], pin_cfg);
+	down(&codec->spdif_mutex);
+	if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
+		if (chs == 2 &&
+		    snd_hda_is_supported_format(codec, mout->dig_out_nid, format) &&
+		    ! (codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+			mout->dig_out_used = HDA_DIG_ANALOG_DUP;
+			/* setup digital receiver */
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+						   stream_tag, 0, format);
+		} else {
+			mout->dig_out_used = 0;
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0);
+		}
 	}
-
-	return 0;
-}
-
-static int stac92xx_init(struct hda_codec *codec)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	int i;
-
-	for (i=0; i < spec->num_pstates; i++)
-		snd_hda_codec_write(codec, spec->pstate_nids[i], 0,
-				    AC_VERB_SET_POWER_STATE, 0x00);
-
-	mdelay(100);
-
-	snd_hda_sequence_write(codec, spec->init);
-
-#ifdef STAC_TEST
-	stac92xx_set_config_regs(codec);
-#endif
-
-	stac92xx_config_pins(codec);
-
+	up(&codec->spdif_mutex);
+
+	/* front */
+	snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format);
+	if (mout->hp_nid)
+		/* headphone out will just decode front left/right (stereo) */
+		snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format);
+	/* surrounds */
+	if (mout->max_channels > 2)
+		for (i = 1; i < mout->num_dacs; i++) {
+			if ((mout->max_channels == 6) && (i == 3))
+				break;
+			if (chs >= (i + 1) * 2) /* independent out */
+				snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2,
+						format);
+			else /* copy front */
+				snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0,
+						format);
+		}
 	return 0;
 }
 
-/*
- * Analog playback callbacks
- */
-static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      snd_pcm_substream_t *substream)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
-}
 
 static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					 struct hda_codec *codec,
@@ -437,7 +379,7 @@ static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					 snd_pcm_substream_t *substream)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+	return stac92xx_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
 						format, substream);
 }
 
@@ -516,7 +458,7 @@ static struct hda_pcm_stream stac92xx_pcm_digital_capture = {
 static struct hda_pcm_stream stac92xx_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
-	.channels_max = 2,
+	.channels_max = 8,
 	.nid = 0x02, /* NID to query formats and rates */
 	.ops = {
 		.open = stac92xx_playback_pcm_open,
@@ -544,11 +486,9 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 	codec->num_pcms = 1;
 	codec->pcm_info = info;
 
-	info->name = "STAC92xx";
+	info->name = "STAC92xx Analog";
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->playback_nid;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->capture_nid;
 
 	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
 		codec->num_pcms++;
@@ -567,9 +507,325 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+#define NUM_CONTROL_ALLOC	32
+
+enum {
+	STAC_CTL_WIDGET_VOL,
+	STAC_CTL_WIDGET_MUTE,
+};
+
+static snd_kcontrol_new_t stac92xx_control_templates[] = {
+	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+	HDA_CODEC_MUTE(NULL, 0, 0, 0),
+};
+
+/* add dynamic controls */
+static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val)
+{
+	snd_kcontrol_new_t *knew;
+
+	if (spec->num_kctl_used >= spec->num_kctl_alloc) {
+		int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
+
+		knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
+		if (! knew)
+			return -ENOMEM;
+		if (spec->kctl_alloc) {
+			memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
+			kfree(spec->kctl_alloc);
+		}
+		spec->kctl_alloc = knew;
+		spec->num_kctl_alloc = num;
+	}
+
+	knew = &spec->kctl_alloc[spec->num_kctl_used];
+	*knew = stac92xx_control_templates[type];
+	knew->name = snd_kmalloc_strdup(name, GFP_KERNEL);
+	if (! knew->name)
+		return -ENOMEM;
+	knew->private_value = val;
+	spec->num_kctl_used++;
+	return 0;
+}
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t nid;
+	int i;
+
+	/* check the pins hardwired to audio widget */
+	for (i = 0; i < cfg->line_outs; i++) {
+		nid = cfg->line_out_pins[i];
+		spec->multiout.dac_nids[i] = snd_hda_codec_read(codec, nid, 0,
+					AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+	}
+
+	spec->multiout.num_dacs = cfg->line_outs;
+
+	return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const struct auto_pin_cfg *cfg)
+{
+	char name[32];
+	static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
+	hda_nid_t nid;
+	int i, err;
+
+	for (i = 0; i < cfg->line_outs; i++) {
+		if (! spec->multiout.dac_nids[i])
+			continue;
+
+		nid = spec->multiout.dac_nids[i];
+
+		if (i == 2) {
+			/* Center/LFE */
+			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Center Playback Volume",
+					       HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
+				return err;
+			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "LFE Playback Volume",
+					       HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
+				return err;
+			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Center Playback Switch",
+					       HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
+				return err;
+			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "LFE Playback Switch",
+					       HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
+				return err;
+		} else {
+			sprintf(name, "%s Playback Volume", chname[i]);
+			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
+					       HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+				return err;
+			sprintf(name, "%s Playback Switch", chname[i]);
+			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
+					       HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+				return err;
+		}
+	}
+
+	return 0;
+}
+
+/* add playback controls for HP output */
+static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t pin = cfg->hp_pin;
+	hda_nid_t nid;
+	int i, err;
+
+	if (! pin)
+		return 0;
+
+	nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+	for (i = 0; i < cfg->line_outs; i++) {
+		if (! spec->multiout.dac_nids[i])
+			continue;
+		if (spec->multiout.dac_nids[i] == nid)
+			return 0;
+	}
+
+	spec->multiout.hp_nid = nid;
+
+	/* control HP volume/switch on the output mixer amp */
+	if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Headphone Playback Volume",
+					HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+		return err;
+	if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
+					HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+		return err;
+
+	return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	static char *labels[AUTO_PIN_LAST] = {
+		"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux"
+	};
+	struct hda_input_mux *imux = &spec->private_imux;
+	hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
+	int i, j, k;
+
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		int index = -1;
+		if (cfg->input_pins[i]) {
+			imux->items[imux->num_items].label = labels[i];
+
+			for (j=0; j<spec->num_muxes; j++) {
+				int num_cons = snd_hda_get_connections(codec, spec->mux_nids[j], con_lst, HDA_MAX_NUM_INPUTS);
+				for (k=0; k<num_cons; k++)
+					if (con_lst[k] == cfg->input_pins[i]) {
+						index = k;
+					 	break;
+					}
+				if (index >= 0)
+					break;
+			}
+			imux->items[imux->num_items].index = index;
+			imux->num_items++;
+		}
+	}
+
+	return 0;
+}
+
+static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
+
+{
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+}
+
+static void stac92xx_auto_init_multi_out(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->autocfg.line_outs; i++) {
+		hda_nid_t nid = spec->autocfg.line_out_pins[i];
+		stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+	}
+}
+
+static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t pin;
+
+	pin = spec->autocfg.hp_pin;
+	if (pin) /* connect to front */
+		stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+}
+
+static int stac922x_parse_auto_config(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int err;
+
+	if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0)
+		return err;
+	if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
+		return err;
+	if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin)
+		return 0; /* can't find valid pin config */
+
+	if ((err = stac92xx_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
+	    (err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg)) < 0 ||
+	    (err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0)
+		return err;
+
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+	if (spec->multiout.max_channels > 2) {
+		spec->surr_switch = 1;
+		spec->cur_ch_mode = 1;
+		spec->num_ch_modes = 2;
+		if (spec->multiout.max_channels == 8) {
+			spec->cur_ch_mode++;
+			spec->num_ch_modes++;
+		}
+	}
+
+	if (spec->autocfg.dig_out_pin) {
+		spec->multiout.dig_out_nid = 0x08;
+		stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN);
+	}
+	if (spec->autocfg.dig_in_pin) {
+		spec->dig_in_nid = 0x09;
+		stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN);
+	}
+
+	if (spec->kctl_alloc)
+		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+	spec->input_mux = &spec->private_imux;
+
+	return 1;
+}
+
+static int stac9200_parse_auto_config(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int err;
+
+	if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0)
+		return err;
+
+	if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0)
+		return err;
+
+	if (spec->autocfg.dig_out_pin) {
+		spec->multiout.dig_out_nid = 0x05;
+		stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN);
+	}
+	if (spec->autocfg.dig_in_pin) {
+		spec->dig_in_nid = 0x04;
+		stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN);
+	}
+
+	if (spec->kctl_alloc)
+		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+	spec->input_mux = &spec->private_imux;
+
+	return 1;
+}
+
+static int stac92xx_init_pstate(struct hda_codec *codec)
+{
+       hda_nid_t nid, nid_start;
+       int nodes;
+
+	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_POWER_STATE, 0x00);
+
+       nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start);
+       for (nid = nid_start; nid < nodes + nid_start; nid++) {
+               unsigned int wid_caps = snd_hda_param_read(codec, nid,
+                                                  AC_PAR_AUDIO_WIDGET_CAP);
+		if (wid_caps & AC_WCAP_POWER)
+			snd_hda_codec_write(codec, nid, 0,
+                                    AC_VERB_SET_POWER_STATE, 0x00);
+	}
+
+	mdelay(100);
+
+	return 0;
+}
+
+static int stac92xx_init(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	stac92xx_init_pstate(codec);
+
+	snd_hda_sequence_write(codec, spec->init);
+
+	stac92xx_auto_init_multi_out(codec);
+	stac92xx_auto_init_hp_out(codec);
+
+	return 0;
+}
+
 static void stac92xx_free(struct hda_codec *codec)
 {
-	kfree(codec->spec);
+	struct sigmatel_spec *spec = codec->spec;
+	int i;
+
+	if (! spec)
+		return;
+
+	if (spec->kctl_alloc) {
+		for (i = 0; i < spec->num_kctl_used; i++)
+			kfree(spec->kctl_alloc[i].name);
+		kfree(spec->kctl_alloc);
+	}
+
+	kfree(spec);
 }
 
 static struct hda_codec_ops stac92xx_patch_ops = {
@@ -582,6 +838,7 @@ static struct hda_codec_ops stac92xx_patch_ops = {
 static int patch_stac9200(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	int err;
 
 	spec  = kcalloc(1, sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -589,26 +846,27 @@ static int patch_stac9200(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+#ifdef STAC_TEST
+	spec->pin_nids = stac9200_pin_nids;
+	spec->num_pins = 8;
+	spec->pin_configs = stac9200_pin_configs;
+	stac92xx_set_config_regs(codec);
+#endif
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = stac9200_dac_nids;
-	spec->multiout.dig_out_nid = 0x05;
-	spec->dig_in_nid = 0x04;
 	spec->adc_nids = stac9200_adc_nids;
 	spec->mux_nids = stac9200_mux_nids;
 	spec->num_muxes = 1;
-	spec->input_mux.num_items = 0;
-	spec->pstate_nids = stac9200_pstate_nids;
-	spec->num_pstates = 3;
-	spec->pin_nids = stac9200_pin_nids;
-#ifdef STAC_TEST
-	spec->pin_configs = stac9200_pin_configs;
-#endif
-	spec->num_pins = 8;
-	spec->init = stac9200_ch2_init;
+
+	spec->init = stac9200_core_init;
 	spec->mixer = stac9200_mixer;
-	spec->playback_nid = 0x02;
-	spec->capture_nid = 0x03;
+
+	err = stac9200_parse_auto_config(codec);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
+	}
 
 	codec->patch_ops = stac92xx_patch_ops;
 
@@ -618,6 +876,7 @@ static int patch_stac9200(struct hda_codec *codec)
 static int patch_stac922x(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	int err;
 
 	spec  = kcalloc(1, sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -625,26 +884,26 @@ static int patch_stac922x(struct hda_codec *codec)
 
 	codec->spec = spec;
 
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = 4;
-	spec->multiout.dac_nids = stac922x_dac_nids;
-	spec->multiout.dig_out_nid = 0x08;
-	spec->dig_in_nid = 0x09;
-	spec->adc_nids = stac922x_adc_nids;
-	spec->mux_nids = stac922x_mux_nids;
-	spec->num_muxes = 2;
-	spec->input_mux.num_items = 0;
-	spec->pstate_nids = stac922x_pstate_nids;
-	spec->num_pstates = 8;
-	spec->pin_nids = stac922x_pin_nids;
 #ifdef STAC_TEST
+	spec->num_pins = 10;
+	spec->pin_nids = stac922x_pin_nids;
 	spec->pin_configs = stac922x_pin_configs;
+	stac92xx_set_config_regs(codec);
 #endif
-	spec->num_pins = 10;
-	spec->init = stac922x_ch2_init;
+	spec->adc_nids = stac922x_adc_nids;
+	spec->mux_nids = stac922x_mux_nids;
+	spec->num_muxes = 2;
+
+	spec->init = stac922x_core_init;
 	spec->mixer = stac922x_mixer;
-	spec->playback_nid = 0x02;
-	spec->capture_nid = 0x06;
+
+	spec->multiout.dac_nids = spec->dac_nids;
+
+	err = stac922x_parse_auto_config(codec);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
+	}
 
 	codec->patch_ops = stac92xx_patch_ops;
 
-- 
cgit v1.2.3


From ff6fdc37fbe66e24ef9ad7c23a278ff757480dda Mon Sep 17 00:00:00 2001
From: Matt <matt@embeddedalley.com>
Date: Mon, 27 Jun 2005 15:06:52 +0200
Subject: [ALSA] hda-codec - SigmaTel HDA resume support

HDA Codec driver
Adds resume support to the SigmaTel HDA patch. Please apply.

Signed-off-by: Matt <matt@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 21 +++++++++++++++++++++
 1 file changed, 21 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index fad825677e7..01cc58a247c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -828,11 +828,32 @@ static void stac92xx_free(struct hda_codec *codec)
 	kfree(spec);
 }
 
+#ifdef CONFIG_PM
+static int stac92xx_resume(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int i;
+
+	stac92xx_init(codec);
+	for (i = 0; i < spec->num_mixers; i++)
+		snd_hda_resume_ctls(codec, spec->mixers[i]);
+	if (spec->multiout.dig_out_nid)
+		snd_hda_resume_spdif_out(codec);
+	if (spec->dig_in_nid)
+		snd_hda_resume_spdif_in(codec);
+
+	return 0;
+}
+#endif
+
 static struct hda_codec_ops stac92xx_patch_ops = {
 	.build_controls = stac92xx_build_controls,
 	.build_pcms = stac92xx_build_pcms,
 	.init = stac92xx_init,
 	.free = stac92xx_free,
+#ifdef CONFIG_PM
+	.resume = stac92xx_resume,
+#endif
 };
 
 static int patch_stac9200(struct hda_codec *codec)
-- 
cgit v1.2.3


From b95eed7cde4a44476fa12e776e090fc494059458 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Jun 2005 15:07:33 +0200
Subject: [ALSA] trident - Shut up compile warnings

Trident driver
Shut up compile warnings about uninitialized variables.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/trident/trident_main.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index a09b0fb49e8..b01c91bb5f6 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -472,6 +472,7 @@ void snd_trident_write_voice_regs(trident_t * trident,
 		break;
 	default:
 		snd_BUG();
+		return;
 	}
 
 	outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
-- 
cgit v1.2.3


From 548e7823bc33b8cde4de59dfafe0fd69d951d3b5 Mon Sep 17 00:00:00 2001
From: Harald Welte <laforge@gnumonks.org>
Date: Mon, 27 Jun 2005 15:10:56 +0200
Subject: [ALSA] Add new pci device id (SB400) to atiixp-modem

ATIIXP-modem driver
I didn't actually test whether the modem works, but at least the driver
loads and initializes fine.  Please consider inclusion.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/atiixp_modem.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index a6b4b8d589f..8d2002951bd 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -265,6 +265,7 @@ struct snd_atiixp {
  */
 static struct pci_device_id snd_atiixp_ids[] = {
 	{ 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */
+	{ 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */
 	{ 0, }
 };
 
-- 
cgit v1.2.3


From b6482d48e536729829025262d6529df09ae20396 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Jun 2005 15:32:43 +0200
Subject: [ALSA] hda-codec - Add 6stack model for ALC880

Documentation,HDA Codec driver
- Added a new '6stack' model for ALC880.
- Fixed the typo in 6stack-digout model name.
- Added description for missing models in ALSA-Configuration.txt.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 14 +++++++++++++-
 1 file changed, 13 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ed16ce817dd..9b4339a004f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -40,6 +40,7 @@ enum {
 	ALC880_W810,
 	ALC880_Z71V,
 	ALC880_AUTO,
+	ALC880_6ST,
 	ALC880_6ST_DIG,
 	ALC880_F1734,
 	ALC880_ASUS,
@@ -1559,7 +1560,9 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 	{ .modelname = "z71v", .config = ALC880_Z71V },
 	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V },
 
-	{ .modelname = "6statack-digout", .config = ALC880_6ST_DIG },
+	{ .modelname = "6stack", .config = ALC880_6ST },
+
+	{ .modelname = "6stack-digout", .config = ALC880_6ST_DIG },
 	{ .pci_subvendor = 0x2668, .pci_subdevice = 0x8086, .config = ALC880_6ST_DIG },
 	{ .pci_subvendor = 0x8086, .pci_subdevice = 0x2668, .config = ALC880_6ST_DIG },
 	{ .pci_subvendor = 0x1462, .pci_subdevice = 0x1150, .config = ALC880_6ST_DIG },
@@ -1646,6 +1649,15 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_fivestack_modes,
 		.input_mux = &alc880_capture_source,
 	},
+	[ALC880_6ST] = {
+		.mixers = { alc880_six_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+		.dac_nids = alc880_6st_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+		.channel_mode = alc880_sixstack_modes,
+		.input_mux = &alc880_6stack_capture_source,
+	},
 	[ALC880_6ST_DIG] = {
 		.mixers = { alc880_six_stack_mixer },
 		.init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
-- 
cgit v1.2.3


From 7a318a70a42057692f191ff49c289cd3e27e21f5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jun 2005 14:16:21 +0200
Subject: [ALSA] hda-codec - Add entry for Acer APFV

HDA Codec driver
Added the model entry for Acer APFV.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9b4339a004f..9b856990078 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1549,7 +1549,7 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 	{ .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG },
 	{ .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG },
 	{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG },
-	{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG },
+	/* { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, */ /* conflict with 6stack */
 	{ .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG },
 	/* note subvendor = 0 below */
 	/* { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG }, */
@@ -1561,6 +1561,7 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V },
 
 	{ .modelname = "6stack", .config = ALC880_6ST },
+	{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_6ST }, /* Acer APFV */
 
 	{ .modelname = "6stack-digout", .config = ALC880_6ST_DIG },
 	{ .pci_subvendor = 0x2668, .pci_subdevice = 0x8086, .config = ALC880_6ST_DIG },
-- 
cgit v1.2.3


From 82fe0c5803f4c77ffeb4c1c2367defb3dcedad45 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2005 10:54:33 +0200
Subject: [ALSA] Use kstrdup

HDA Codec driver
Use the new kstrdup() function instead of in-house one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 01cc58a247c..07d06f7c439 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -540,7 +540,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char
 
 	knew = &spec->kctl_alloc[spec->num_kctl_used];
 	*knew = stac92xx_control_templates[type];
-	knew->name = snd_kmalloc_strdup(name, GFP_KERNEL);
+	knew->name = kstrdup(name, GFP_KERNEL);
 	if (! knew->name)
 		return -ENOMEM;
 	knew->private_value = val;
-- 
cgit v1.2.3


From 2eff7ec81eb586076974cb0918dffc5f4ad763d5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2005 13:45:20 +0200
Subject: [ALSA] cmipci - Add Mic Boost capture switch

CMIPCI driver
Added 'Mic Boost Capture Switch' and 'Phone' switches.
The existing playback switch is renamed as 'Mic Boost Playback Switch'.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/cmipci.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 4725b4a010b..f5a4ac1ceef 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -306,7 +306,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
 #define CM_REG_FM_PCI		0x50
 
 /*
- * for CMI-8338 .. this is not valid for CMI-8738.
+ * access from SB-mixer port
  */
 #define CM_REG_EXTENT_IND	0xf0
 #define CM_VPHONE_MASK		0xe0	/* Phone volume control (0-3) << 5 */
@@ -315,6 +315,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
 #define CM_VSPKM		0x08	/* Speaker mute control, default high */
 #define CM_RLOOPREN		0x04    /* Rec. R-channel enable */
 #define CM_RLOOPLEN		0x02	/* Rec. L-channel enable */
+#define CM_VADMIC3		0x01	/* Mic record boost */
 
 /*
  * CMI-8338 spec ver 0.5 (this is not valid for CMI-8738):
@@ -2135,8 +2136,12 @@ static snd_kcontrol_new_t snd_cmipci_mixers[] __devinitdata = {
 	CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15),
 	CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0),
 	CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0),
-	CMIPCI_MIXER_SW_MONO("Mic Boost", CM_REG_MIXER2, CM_MICGAINZ_SHIFT, 1),
+	CMIPCI_MIXER_SW_MONO("Mic Boost Playback Switch", CM_REG_MIXER2, CM_MICGAINZ_SHIFT, 1),
 	CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7),
+	CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7),
+	CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0),
+	CMIPCI_DOUBLE("PC Speaker Playnack Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
+	CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0),
 };
 
 /*
-- 
cgit v1.2.3


From 52b723888c1a55d34551f9b0b9d9296e0e3e8d3c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2005 13:47:06 +0200
Subject: [ALSA] Fix resume of intel8x0

Intel8x0 driver,AC97 Codec
Fix resume of intel8x0 driver.  The ac97 codec didn't restore some registers
properly, and the restore of ICH4 SPDIF and SDIN settings was missing.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c |  2 ++
 sound/pci/intel8x0.c        | 13 +++++++++++++
 2 files changed, 15 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index a4b72cd2eea..0677d41239a 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -367,6 +367,7 @@ int snd_ac97_update(ac97_t *ac97, unsigned short reg, unsigned short value)
 		ac97->regs[reg] = value;
 		ac97->bus->ops->write(ac97, reg, value);
 	}
+	set_bit(reg, ac97->reg_accessed);
 	up(&ac97->reg_mutex);
 	return change;
 }
@@ -410,6 +411,7 @@ int snd_ac97_update_bits_nolock(ac97_t *ac97, unsigned short reg,
 		ac97->regs[reg] = new;
 		ac97->bus->ops->write(ac97, reg, new);
 	}
+	set_bit(reg, ac97->reg_accessed);
 	return change;
 }
 
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index cc16f95f9ce..c3c3b68b454 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -424,6 +424,7 @@ struct _snd_intel8x0 {
 	unsigned xbox: 1;		/* workaround for Xbox AC'97 detection */
 
 	int spdif_idx;	/* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */
+	unsigned int sdm_saved;	/* SDM reg value */
 
 	ac97_bus_t *ac97_bus;
 	ac97_t *ac97[3];
@@ -2373,6 +2374,8 @@ static int intel8x0_suspend(snd_card_t *card, pm_message_t state)
 	for (i = 0; i < 3; i++)
 		if (chip->ac97[i])
 			snd_ac97_suspend(chip->ac97[i]);
+	if (chip->device_type == DEVICE_INTEL_ICH4)
+		chip->sdm_saved = igetbyte(chip, ICHREG(SDM));
 	pci_disable_device(chip->pci);
 	return 0;
 }
@@ -2386,6 +2389,16 @@ static int intel8x0_resume(snd_card_t *card)
 	pci_set_master(chip->pci);
 	snd_intel8x0_chip_init(chip, 0);
 
+	/* re-initialize mixer stuff */
+	if (chip->device_type == DEVICE_INTEL_ICH4) {
+		/* enable separate SDINs for ICH4 */
+		iputbyte(chip, ICHREG(SDM), chip->sdm_saved);
+		/* use slot 10/11 for SPDIF */
+		iputdword(chip, ICHREG(GLOB_CNT),
+			  (igetdword(chip, ICHREG(GLOB_CNT)) & ~ICH_PCM_SPDIF_MASK) |
+			  ICH_PCM_SPDIF_1011);
+	}
+
 	/* refill nocache */
 	if (chip->fix_nocache)
 		fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
-- 
cgit v1.2.3


From 5ba1e7b594db4d0e1f88ace87c1cb295761ca5c9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2005 13:47:58 +0200
Subject: [ALSA] maestro3 - Clean up

Maestro3 driver
- Clean up maestro3 code
- Use msleep()
- Don't enable hw-vol irq when not defined

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/maestro3.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 52c585901c5..39b5e7db154 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -1050,11 +1050,6 @@ static struct m3_hv_quirk m3_hv_quirk_list[] = {
  * lowlevel functions
  */
 
-#define big_mdelay(msec) do {\
-	set_current_state(TASK_UNINTERRUPTIBLE);\
-	schedule_timeout(((msec) * HZ) / 1000);\
-} while (0)
-	
 static inline void snd_m3_outw(m3_t *chip, u16 value, unsigned long reg)
 {
 	outw(value, chip->iobase + reg);
@@ -1096,7 +1091,7 @@ static void snd_m3_assp_write(m3_t *chip, u16 region, u16 index, u16 data)
 static void snd_m3_assp_halt(m3_t *chip)
 {
 	chip->reset_state = snd_m3_inb(chip, DSP_PORT_CONTROL_REG_B) & ~REGB_STOP_CLOCK;
-	big_mdelay(10);
+	msleep(10);
 	snd_m3_outb(chip, chip->reset_state & ~REGB_ENABLE_RESET, DSP_PORT_CONTROL_REG_B);
 }
 
@@ -2108,9 +2103,9 @@ static void snd_m3_ac97_reset(m3_t *chip)
 	 */
 	tmp = inw(io + RING_BUS_CTRL_A);
 	outw(RAC_SDFS_ENABLE|LAC_SDFS_ENABLE, io + RING_BUS_CTRL_A);
-	big_mdelay(20);
+	msleep(20);
 	outw(tmp, io + RING_BUS_CTRL_A);
-	big_mdelay(50);
+	msleep(50);
 #endif
 }
 
@@ -2525,9 +2520,13 @@ static void
 snd_m3_enable_ints(m3_t *chip)
 {
 	unsigned long io = chip->iobase;
+	unsigned short val;
 
 	/* TODO: MPU401 not supported yet */
-	outw(ASSP_INT_ENABLE | HV_INT_ENABLE /*| MPU401_INT_ENABLE*/, io + HOST_INT_CTRL);
+	val = ASSP_INT_ENABLE /*| MPU401_INT_ENABLE*/;
+	if (chip->hv_quirk && (chip->hv_quirk->config & HV_CTRL_ENABLE))
+		val |= HV_INT_ENABLE;
+	outw(val, io + HOST_INT_CTRL);
 	outb(inb(io + ASSP_CONTROL_C) | ASSP_HOST_INT_ENABLE,
 	     io + ASSP_CONTROL_C);
 }
@@ -2589,7 +2588,7 @@ static int m3_suspend(snd_card_t *card, pm_message_t state)
 	snd_pcm_suspend_all(chip->pcm);
 	snd_ac97_suspend(chip->ac97);
 
-	big_mdelay(10); /* give the assp a chance to idle.. */
+	msleep(10); /* give the assp a chance to idle.. */
 
 	snd_m3_assp_halt(chip);
 
@@ -2697,6 +2696,8 @@ snd_m3_create(snd_card_t *card, struct pci_dev *pci,
 	}
 
 	spin_lock_init(&chip->reg_lock);
+	spin_lock_init(&chip->ac97_lock);
+
 	switch (pci->device) {
 	case PCI_DEVICE_ID_ESS_ALLEGRO:
 	case PCI_DEVICE_ID_ESS_ALLEGRO_1:
@@ -2765,6 +2766,8 @@ snd_m3_create(snd_card_t *card, struct pci_dev *pci,
 	snd_m3_assp_init(chip);
 	snd_m3_amp_enable(chip, 1);
 
+	tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip);
+
 	if (request_irq(pci->irq, snd_m3_interrupt, SA_INTERRUPT|SA_SHIRQ,
 			card->driver, (void *)chip)) {
 		snd_printk("unable to grab IRQ %d\n", pci->irq);
@@ -2786,9 +2789,6 @@ snd_m3_create(snd_card_t *card, struct pci_dev *pci,
 		return err;
 	}
 
-	spin_lock_init(&chip->ac97_lock);
-	tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip);
-
 	if ((err = snd_m3_mixer(chip)) < 0)
 		return err;
 
-- 
cgit v1.2.3


From 7bc71ecd6477db90221efc08fb742b3df4f49b46 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@suse.cz>
Date: Sat, 2 Jul 2005 15:20:57 +0200
Subject: [ALSA] via82xx - added MSI K7T266 Pro2 - 4005:4710 to white list (DXS
 enable)

VIA82xx driver
Reporter: Marko Kohtala <marko.kohtala@gmail.com>

Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/via82xx.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 42c48f0ce8e..1a286e1a60d 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2187,6 +2187,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci)
 		{ .subvendor = 0x1695, .subdevice = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */
 		{ .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */
 		{ .subvendor = 0x1919, .subdevice = 0x200a, .action = VIA_DXS_NO_VRA }, /* Soltek SL-K8Tpro-939 */
+		{ .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_ENABLE }, /* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */
 		{ } /* terminator */
 	};
 	struct dxs_whitelist *w;
-- 
cgit v1.2.3


From e0474e53985c5fac97a5bb85d66ec0d017b5faf3 Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Sat, 2 Jul 2005 16:33:34 +0200
Subject: [ALSA] snd-emu10k1: Card capabilities tidy up.

EMU10K1/EMU10K2 driver

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/emu10k1.c      | 4 ++--
 sound/pci/emu10k1/emu10k1_main.c | 8 +++++---
 2 files changed, 7 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 2085a998eae..2f96b2fc74f 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -140,7 +140,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
 		return err;
 	}
 	/* This stores the periods table. */
-	if (emu->audigy && emu->revision == 4) { /* P16V */	
+	if (emu->card_capabilities->ca0151_chip) { /* P16V */	
 		if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &emu->p16v_buffer) < 0) {
 			snd_p16v_free(emu);
 			return -ENOMEM;
@@ -161,7 +161,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
 		snd_card_free(card);
 		return err;
 	}
-	if (emu->audigy && emu->revision == 4) { /* P16V */	
+	if (emu->card_capabilities->ca0151_chip) { /* P16V */	
 		if ((err = snd_p16v_pcm(emu, 4, NULL)) < 0) {
 			snd_card_free(card);
 			return err;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index a341e758acd..9478a95e410 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -191,7 +191,7 @@ static int __devinit snd_emu10k1_init(emu10k1_t * emu, int enable_ir)
 		/* Set playback routing. */
 		snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, 0x78e4);
 	}
-	if (emu->audigy && (emu->serial == 0x10011102) ) { /* audigy2 Value */
+	if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */
 		/* Hacks for Alice3 to work independent of haP16V driver */
 		u32 tmp;
 
@@ -253,6 +253,8 @@ static int __devinit snd_emu10k1_init(emu10k1_t * emu, int enable_ir)
 			     HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG);
 		else
 			outl(HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG);
+	/* FIXME: Remove all these emu->model and replace it with a card recognition parameter,
+	 * e.g. card_capabilities->joystick */
 	} else if (emu->model == 0x20 ||
 	    emu->model == 0xc400 ||
 	    (emu->model == 0x21 && emu->revision < 6))
@@ -299,12 +301,12 @@ static int __devinit snd_emu10k1_init(emu10k1_t * emu, int enable_ir)
 	if (emu->audigy) {
 		outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG);
  
-		if (emu->revision == 4) { /* audigy2 */
+		if (emu->card_capabilities->ca0151_chip) { /* audigy2 */
 			/* Unmute Analog now.  Set GPO6 to 1 for Apollo.
 			 * This has to be done after init ALice3 I2SOut beyond 48KHz.
 			 * So, sequence is important. */
 			outl(inl(emu->port + A_IOCFG) | 0x0040, emu->port + A_IOCFG);
-		} else if (emu->serial == 0x10011102) { /* audigy2 value */
+		} else if (emu->card_capabilities->ca0108_chip) { /* audigy2 value */
 			/* Unmute Analog now. */
 			outl(inl(emu->port + A_IOCFG) | 0x0060, emu->port + A_IOCFG);
 		} else {
-- 
cgit v1.2.3


From 3818152e64866b54020b5656ff5fdd0f5e085183 Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Sat, 2 Jul 2005 18:03:37 +0200
Subject: [ALSA] snd-emu10k1: Tidy mixer controls.

EMU10K1/EMU10K2 driver

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/p16v.c | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 98f98018989..a1691330d3b 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -822,7 +822,7 @@ static int snd_p16v_volume_put_analog_unknown(snd_kcontrol_t * kcontrol,
 static snd_kcontrol_new_t snd_p16v_volume_control_analog_front =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD Analog Front Volume",
+        .name =         "HD Analog Front Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_analog_front,
         .put =          snd_p16v_volume_put_analog_front
@@ -831,7 +831,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_front =
 static snd_kcontrol_new_t snd_p16v_volume_control_analog_center_lfe =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD Analog Center/LFE Volume",
+        .name =         "HD Analog Center/LFE Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_analog_center_lfe,
         .put =          snd_p16v_volume_put_analog_center_lfe
@@ -840,7 +840,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_center_lfe =
 static snd_kcontrol_new_t snd_p16v_volume_control_analog_unknown =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD Analog Unknown Volume",
+        .name =         "HD Analog Unknown Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_analog_unknown,
         .put =          snd_p16v_volume_put_analog_unknown
@@ -849,7 +849,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_unknown =
 static snd_kcontrol_new_t snd_p16v_volume_control_analog_rear =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD Analog Rear Volume",
+        .name =         "HD Analog Rear Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_analog_rear,
         .put =          snd_p16v_volume_put_analog_rear
@@ -858,7 +858,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_rear =
 static snd_kcontrol_new_t snd_p16v_volume_control_spdif_front =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD SPDIF Front Volume",
+        .name =         "HD SPDIF Front Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_spdif_front,
         .put =          snd_p16v_volume_put_spdif_front
@@ -867,7 +867,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_spdif_front =
 static snd_kcontrol_new_t snd_p16v_volume_control_spdif_center_lfe =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD SPDIF Center/LFE Volume",
+        .name =         "HD SPDIF Center/LFE Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_spdif_center_lfe,
         .put =          snd_p16v_volume_put_spdif_center_lfe
@@ -876,7 +876,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_spdif_center_lfe =
 static snd_kcontrol_new_t snd_p16v_volume_control_spdif_unknown =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD SPDIF Unknown Volume",
+        .name =         "HD SPDIF Unknown Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_spdif_unknown,
         .put =          snd_p16v_volume_put_spdif_unknown
@@ -885,7 +885,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_spdif_unknown =
 static snd_kcontrol_new_t snd_p16v_volume_control_spdif_rear =
 {
         .iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
-        .name =         "HD SPDIF Rear Volume",
+        .name =         "HD SPDIF Rear Playback Volume",
         .info =         snd_p16v_volume_info,
         .get =          snd_p16v_volume_get_spdif_rear,
         .put =          snd_p16v_volume_put_spdif_rear
@@ -936,7 +936,7 @@ static int snd_p16v_capture_source_put(snd_kcontrol_t * kcontrol,
 static snd_kcontrol_new_t snd_p16v_capture_source __devinitdata =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name =		"HD Capture source",
+	.name =		"HD source Capture",
 	.info =		snd_p16v_capture_source_info,
 	.get =		snd_p16v_capture_source_get,
 	.put =		snd_p16v_capture_source_put
@@ -985,7 +985,7 @@ static int snd_p16v_capture_channel_put(snd_kcontrol_t * kcontrol,
 static snd_kcontrol_new_t snd_p16v_capture_channel __devinitdata =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name =		"HD Capture channel",
+	.name =		"HD channel Capture",
 	.info =		snd_p16v_capture_channel_info,
 	.get =		snd_p16v_capture_channel_get,
 	.put =		snd_p16v_capture_channel_put
-- 
cgit v1.2.3


From a6f6192bb38a76c4ad44c894144b1fbf3d14606b Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Sun, 3 Jul 2005 12:32:40 +0200
Subject: [ALSA] emu10k1: Sort by card id.

EMU10K1/EMU10K2 driver

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/emu10k1_main.c | 80 +++++++++++++++++++---------------------
 1 file changed, 37 insertions(+), 43 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 9478a95e410..6dca38120b9 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -714,54 +714,49 @@ static emu_chip_details_t emu_chip_details[] = {
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
 	 .ac97_chip = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102,
-	 .driver = "EMU10K1", .name = "E-mu APS [4001]", 
-	 .id = "APS",
-	 .emu10k1_chip = 1,
-	 .ecard = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
-	 .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102,
+	 .driver = "EMU10K1", .name = "SBLive! [SB0105]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102,
-	 .driver = "EMU10K1", .name = "SB Live 5.1", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102,
-	 .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", 
-	 .id = "Live",
-	 .emu10k1_chip = 1,
-	 .ac97_chip = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102,
-	 .driver = "EMU10K1", .name = "SBLive! [CT4620]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102,
+	 .driver = "EMU10K1", .name = "SB Live 5.1", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
+	 .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102,
-	 .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102,
+	 .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", 
+	 .id = "Live",
+	 .emu10k1_chip = 1,
+	 .ac97_chip = 1} ,
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
@@ -772,50 +767,49 @@ static emu_chip_details_t emu_chip_details[] = {
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102,
+	 .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [SB0060]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", 
-	 .id = "Live",
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102,
+	 .driver = "EMU10K1", .name = "E-mu APS [4001]", 
+	 .id = "APS",
 	 .emu10k1_chip = 1,
-	 .ac97_chip = 1,
-	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102,
-	 .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", 
+	 .ecard = 1} ,
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102,
+	 .driver = "EMU10K1", .name = "SBLive! [CT4620]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
-	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102,
-	 .driver = "EMU10K1", .name = "SBLive! [SB0105]", 
+	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102,
+	 .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", 
 	 .id = "Live",
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
-- 
cgit v1.2.3


From 88dc0e5dadf9b0cb529c89b12cd10f75d5b1bce4 Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Sun, 3 Jul 2005 12:54:29 +0200
Subject: [ALSA] emu10k1: Added tested status comments.

EMU10K1/EMU10K2 driver

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/emu10k1_main.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 6dca38120b9..8bc9bc18c74 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -616,6 +616,7 @@ static int snd_emu10k1_dev_free(snd_device_t *device)
 
 static emu_chip_details_t emu_chip_details[] = {
 	/* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/
+	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102,
 	 .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", 
 	 .id = "Audigy2",
@@ -629,6 +630,7 @@ static emu_chip_details_t emu_chip_details[] = {
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
 	 .ac97_chip = 1} ,
+	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102,
 	 .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", 
 	 .id = "Audigy2",
@@ -773,6 +775,7 @@ static emu_chip_details_t emu_chip_details[] = {
 	 .emu10k1_chip = 1,
 	 .ac97_chip = 1,
 	 .sblive51 = 1} ,
+	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102,
 	 .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", 
 	 .id = "Live",
-- 
cgit v1.2.3


From 41e2fce431070cb2d91391808077378582d3e6b1 Mon Sep 17 00:00:00 2001
From: Matt <matt@embeddedalley.com>
Date: Mon, 4 Jul 2005 17:49:55 +0200
Subject: [ALSA] hda: enable unsolicited responses

HDA Intel driver
Patch enables unsolicited responses on the HDA controller. Without
the UREN bit set, the controller will not place unsolicited responses
in a RIRB.

Signed-off-by: Matt <matt@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 5e0cca36ed5..288ab076483 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -178,6 +178,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define ICH6_INT_CTRL_EN	0x40000000	/* controller interrupt enable bit */
 #define ICH6_INT_GLOBAL_EN	0x80000000	/* global interrupt enable bit */
 
+/* GCTL unsolicited response enable bit */
+#define ICH6_GCTL_UREN		(1<<8)
+
 /* GCTL reset bit */
 #define ICH6_GCTL_RESET		(1<<0)
 
@@ -562,6 +565,9 @@ static int azx_reset(azx_t *chip)
 		return -EBUSY;
 	}
 
+	/* Accept unsolicited responses */
+	azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN);
+
 	/* detect codecs */
 	if (! chip->codec_mask) {
 		chip->codec_mask = azx_readw(chip, STATESTS);
-- 
cgit v1.2.3


From 4e55096e27d745908e44c6abd2cc0c5b615854a4 Mon Sep 17 00:00:00 2001
From: Matt <matt@embeddedalley.com>
Date: Mon, 4 Jul 2005 17:51:39 +0200
Subject: [ALSA] hda: add sigmatel hp detect support

HDA Codec driver
Adds support for detecting hp insertion/removal and enable/disable of
lineouts based on unsolicited events.

Signed-off-by: Matt <matt@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.h      |  3 ++
 sound/pci/hda/patch_sigmatel.c | 62 ++++++++++++++++++++++++++++++++++++++++--
 2 files changed, 63 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 59991560d49..dd0d99d2ad2 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -262,6 +262,9 @@ enum {
 #define AC_PINCTL_OUT_EN		(1<<6)
 #define AC_PINCTL_HP_EN			(1<<7)
 
+/* Unsolicited response - 8bit */
+#define AC_USRSP_EN			(1<<7)
+
 /* configuration default - 32bit */
 #define AC_DEFCFG_SEQUENCE		(0xf<<0)
 #define AC_DEFCFG_DEF_ASSOC		(0xf<<4)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 07d06f7c439..9d503da7320 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,6 +36,10 @@
 
 #undef STAC_TEST
 
+#define NUM_CONTROL_ALLOC	32
+#define STAC_HP_EVENT		0x37
+#define STAC_UNSOL_ENABLE 	(AC_USRSP_EN | STAC_HP_EVENT)
+
 struct sigmatel_spec {
 	snd_kcontrol_new_t *mixers[4];
 	unsigned int num_mixers;
@@ -507,8 +511,6 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
-#define NUM_CONTROL_ALLOC	32
-
 enum {
 	STAC_CTL_WIDGET_VOL,
 	STAC_CTL_WIDGET_MUTE,
@@ -617,10 +619,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin
 	hda_nid_t pin = cfg->hp_pin;
 	hda_nid_t nid;
 	int i, err;
+	unsigned int wid_caps;
 
 	if (! pin)
 		return 0;
 
+	wid_caps = snd_hda_param_read(codec, pin, AC_PAR_AUDIO_WIDGET_CAP);
+	if (wid_caps & AC_WCAP_UNSOL_CAP)
+		/* Enable unsolicited responses on the HP widget */
+		snd_hda_codec_write(codec, pin, 0,
+				AC_VERB_SET_UNSOLICITED_ENABLE,
+				STAC_UNSOL_ENABLE);
+
 	nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
 	for (i = 0; i < cfg->line_outs; i++) {
 		if (! spec->multiout.dac_nids[i])
@@ -828,6 +838,53 @@ static void stac92xx_free(struct hda_codec *codec)
 	kfree(spec);
 }
 
+static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
+				unsigned int flag)
+{
+	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
+			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
+	snd_hda_codec_write(codec, nid, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL,
+			pin_ctl | flag);
+}
+
+static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
+				  unsigned int flag)
+{
+	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
+			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
+	snd_hda_codec_write(codec, nid, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL,
+			pin_ctl & ~flag);
+}
+
+static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, presence;
+
+	if ((res >> 26) != STAC_HP_EVENT)
+		return;
+
+	presence = snd_hda_codec_read(codec, cfg->hp_pin, 0,
+			AC_VERB_GET_PIN_SENSE, 0x00) >> 31;
+
+	if (presence) {
+		/* disable lineouts, enable hp */
+		for (i = 0; i < cfg->line_outs; i++)
+			stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
+						AC_PINCTL_OUT_EN);
+		stac92xx_set_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN);
+	} else {
+		/* enable lineouts, disable hp */
+		for (i = 0; i < cfg->line_outs; i++)
+			stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
+						AC_PINCTL_OUT_EN);
+		stac92xx_reset_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN);
+	}
+} 
+
 #ifdef CONFIG_PM
 static int stac92xx_resume(struct hda_codec *codec)
 {
@@ -851,6 +908,7 @@ static struct hda_codec_ops stac92xx_patch_ops = {
 	.build_pcms = stac92xx_build_pcms,
 	.init = stac92xx_init,
 	.free = stac92xx_free,
+	.unsol_event = stac92xx_unsol_event,
 #ifdef CONFIG_PM
 	.resume = stac92xx_resume,
 #endif
-- 
cgit v1.2.3


From e3ea4d896109edd64dc549ecaeeff8d89025fb57 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 4 Jul 2005 18:12:39 +0200
Subject: [ALSA] hdsp - Add 'Sample Clock Source Locking' control

RME HDSP driver
Added 'Sample Clock Source Locking' control.  If this switch is on,
the clock source can't be changed via PCM hw_params API (as sample rate).
This will fix the problem of OSS-emulation, for example.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 62 +++++++++++++++++++++++++++++++++++++++++-------
 1 file changed, 53 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index a673cc438b9..0db558a9287 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -445,6 +445,7 @@ struct _hdsp {
 	u32                   control2_register;     /* cached value */
 	u32                   creg_spdif;
 	u32                   creg_spdif_stream;
+	int                   clock_source_locked;
 	char                 *card_name;	     /* digiface/multiface */
 	HDSP_IO_Type          io_type;               /* ditto, but for code use */
         unsigned short        firmware_rev;
@@ -2095,6 +2096,34 @@ static int snd_hdsp_put_clock_source(snd_kcontrol_t * kcontrol, snd_ctl_elem_val
 	return change;
 }
 
+static int snd_hdsp_info_clock_source_lock(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+static int snd_hdsp_get_clock_source_lock(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+	hdsp_t *hdsp = snd_kcontrol_chip(kcontrol);
+	
+	ucontrol->value.integer.value[0] = hdsp->clock_source_locked;
+	return 0;
+}
+
+static int snd_hdsp_put_clock_source_lock(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+	hdsp_t *hdsp = snd_kcontrol_chip(kcontrol);
+	int change;
+
+	change = (int)ucontrol->value.integer.value[0] != hdsp->clock_source_locked;
+	if (change)
+		hdsp->clock_source_locked = ucontrol->value.integer.value[0];
+	return change;
+}
+
 #define HDSP_DA_GAIN(xname, xindex) \
 { .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
   .name = xname, \
@@ -3117,6 +3146,15 @@ HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0),
 HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0),
 /* 'Sample Clock Source' complies with the alsa control naming scheme */ 
 HDSP_CLOCK_SOURCE("Sample Clock Source", 0),
+{
+	/* FIXME: should be PCM or MIXER? */
+	/* .iface = SNDRV_CTL_ELEM_IFACE_PCM, */
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Sample Clock Source Locking",
+	.info = snd_hdsp_info_clock_source_lock,
+	.get = snd_hdsp_get_clock_source_lock,
+	.put = snd_hdsp_put_clock_source_lock,
+},
 HDSP_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
 HDSP_PREF_SYNC_REF("Preferred Sync Reference", 0),
 HDSP_AUTOSYNC_REF("AutoSync Reference", 0),
@@ -3349,6 +3387,7 @@ snd_hdsp_proc_read(snd_info_entry_t *entry, snd_info_buffer_t *buffer)
 	snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode);
 
 	snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate);
+	snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No");
 		
 	snd_iprintf(buffer, "\n");
 
@@ -3853,13 +3892,14 @@ static int snd_hdsp_hw_params(snd_pcm_substream_t *substream,
 	 */
 
 	spin_lock_irq(&hdsp->lock);
-	if ((err = hdsp_set_rate(hdsp, params_rate(params), 0)) < 0) {
-		spin_unlock_irq(&hdsp->lock);
-		_snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
-		return err;
-	} else {
-		spin_unlock_irq(&hdsp->lock);
+	if (! hdsp->clock_source_locked) {
+		if ((err = hdsp_set_rate(hdsp, params_rate(params), 0)) < 0) {
+			spin_unlock_irq(&hdsp->lock);
+			_snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
+			return err;
+		}
 	}
+	spin_unlock_irq(&hdsp->lock);
 
 	if ((err = hdsp_set_interrupt_interval(hdsp, params_period_size(params))) < 0) {
 		_snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
@@ -4284,13 +4324,17 @@ static int snd_hdsp_playback_open(snd_pcm_substream_t *substream)
 
 	snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes);
-	if (hdsp->io_type == H9632) {
-		runtime->hw.channels_min = hdsp->qs_out_channels;
-		runtime->hw.channels_max = hdsp->ss_out_channels;
+	if (hdsp->clock_source_locked) {
+		runtime->hw.rate_min = runtime->hw.rate_max = hdsp->system_sample_rate;
+	} else if (hdsp->io_type == H9632) {
 		runtime->hw.rate_max = 192000;
 		runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
 		snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates);
 	}
+	if (hdsp->io_type == H9632) {
+		runtime->hw.channels_min = hdsp->qs_out_channels;
+		runtime->hw.channels_max = hdsp->ss_out_channels;
+	}	
 	
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 			     snd_hdsp_hw_rule_out_channels, hdsp,
-- 
cgit v1.2.3


From 2201987c562f7c810440d399ef7a85fe79be01e7 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@suse.cz>
Date: Tue, 5 Jul 2005 10:27:09 +0200
Subject: [ALSA] via82xx - changed MSI K7T266 Pro2 - 4005:4710 in white list
 (SRC enable)

VIA82xx driver

Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/via82xx.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 1a286e1a60d..064972b14d0 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2187,7 +2187,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci)
 		{ .subvendor = 0x1695, .subdevice = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */
 		{ .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */
 		{ .subvendor = 0x1919, .subdevice = 0x200a, .action = VIA_DXS_NO_VRA }, /* Soltek SL-K8Tpro-939 */
-		{ .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_ENABLE }, /* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */
+		{ .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_SRC },	/* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */
 		{ } /* terminator */
 	};
 	struct dxs_whitelist *w;
-- 
cgit v1.2.3


From e66bc8b2a7d85166935a2da651b94efb9e7a2f11 Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Wed, 6 Jul 2005 22:21:51 +0200
Subject: [ALSA] emu10k1: Add module option uint subsystem.

EMU10K1/EMU10K2 driver
It allows the user to force the snd-emu10k1 module to think the user
has a particular sound card. Useful if their particular sound card
is not yet recognised.

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/emu10k1.c      |  6 ++++--
 sound/pci/emu10k1/emu10k1_main.c | 25 ++++++++++++++++++-------
 2 files changed, 22 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 2f96b2fc74f..b17142cabea 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -52,6 +52,7 @@ static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4};
 static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
 static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
 static int enable_ir[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
+static uint subsystem[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; /* Force card subsystem model */
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
@@ -71,7 +72,8 @@ module_param_array(max_buffer_size, int, NULL, 0444);
 MODULE_PARM_DESC(max_buffer_size, "Maximum sample buffer size in MB.");
 module_param_array(enable_ir, bool, NULL, 0444);
 MODULE_PARM_DESC(enable_ir, "Enable IR.");
-
+module_param_array(subsystem, uint, NULL, 0444);
+MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
 /*
  * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value  Model:SB0400
  */
@@ -122,7 +124,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
 		max_buffer_size[dev] = 1024;
 	if ((err = snd_emu10k1_create(card, pci, extin[dev], extout[dev],
 				      (long)max_buffer_size[dev] * 1024 * 1024,
-				      enable_ir[dev],
+				      enable_ir[dev], subsystem[dev],
 				      &emu)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 8bc9bc18c74..4ced4b09253 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -832,6 +832,7 @@ int __devinit snd_emu10k1_create(snd_card_t * card,
 		       unsigned short extout_mask,
 		       long max_cache_bytes,
 		       int enable_ir,
+		       uint subsystem,
 		       emu10k1_t ** remu)
 {
 	emu10k1_t *emu;
@@ -877,10 +878,16 @@ int __devinit snd_emu10k1_create(snd_card_t * card,
 
 	for (c = emu_chip_details; c->vendor; c++) {
 		if (c->vendor == pci->vendor && c->device == pci->device) {
-			if (c->subsystem && c->subsystem != emu->serial)
-				continue;
-			if (c->revision && c->revision != emu->revision)
-				continue;
+			if (subsystem) {
+				if (c->subsystem && (c->subsystem == subsystem) ) {
+					break;
+				} else continue;
+			} else {
+				if (c->subsystem && (c->subsystem != emu->serial) )
+					continue;
+				if (c->revision && c->revision != emu->revision)
+					continue;
+			}
 			break;
 		}
 	}
@@ -891,10 +898,14 @@ int __devinit snd_emu10k1_create(snd_card_t * card,
 		return -ENOENT;
 	}
 	emu->card_capabilities = c;
-	if (c->subsystem != 0)
+	if (c->subsystem && !subsystem)
 		snd_printdd("Sound card name=%s\n", c->name);
-	else
-		snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x\n", c->name, pci->vendor, pci->device, emu->serial);
+	else if (subsystem) 
+		snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n",
+		       	c->name, pci->vendor, pci->device, emu->serial, c->subsystem);
+	else 
+		snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n",
+		      	c->name, pci->vendor, pci->device, emu->serial);
 	
 	if (!*card->id && c->id) {
 		int i, n = 0;
-- 
cgit v1.2.3


From ae3a72d8cb4e5b30606c5e3ac9c59b729117579a Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Wed, 6 Jul 2005 22:36:18 +0200
Subject: [ALSA] snd-emu10k1: Fixes recognition of Audigy ES.

EMU10K1/EMU10K2 driver
Fixes ALSA bug #1237.

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/emu10k1_main.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 4ced4b09253..7c31d9b3024 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -691,18 +691,18 @@ static emu_chip_details_t emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spdif_bug = 1,
 	 .ac97_chip = 1} ,
-	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10020052,
-	 .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", 
+	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102,
+	 .driver = "Audigy", .name = "Audigy 1 [SB0090]", 
 	 .id = "Audigy",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
-	 .spdif_bug = 1,
 	 .ac97_chip = 1} ,
-	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102,
-	 .driver = "Audigy", .name = "Audigy 1 [SB0090]", 
+	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102,
+	 .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", 
 	 .id = "Audigy",
 	 .emu10k2_chip = 1,
 	 .ca0102_chip = 1,
+	 .spdif_bug = 1,
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102,
 	 .driver = "Audigy", .name = "Audigy 1 [SB0090]", 
-- 
cgit v1.2.3


From 072c01194df6e4843582d09380b780987f642d6d Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@suse.cz>
Date: Sat, 9 Jul 2005 10:07:55 +0200
Subject: [ALSA] ens1371 - added extra delay for ac97 codec initialization

ENS1370/1+ driver

Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/ens1370.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 4e63498a58b..d4287338c04 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -685,6 +685,15 @@ static unsigned short snd_es1371_codec_read(ac97_t *ac97,
 	return 0;
 }
 
+static void snd_es1371_codec_wait(ac97_t *ac97)
+{
+	msleep(750);
+	snd_es1371_codec_read(ac97, AC97_RESET);
+	snd_es1371_codec_read(ac97, AC97_VENDOR_ID1);
+	snd_es1371_codec_read(ac97, AC97_VENDOR_ID2);
+	msleep(50);
+}
+
 static void snd_es1371_adc_rate(ensoniq_t * ensoniq, unsigned int rate)
 {
 	unsigned int n, truncm, freq, result;
@@ -1585,6 +1594,7 @@ static int snd_ensoniq_1371_mixer(ensoniq_t * ensoniq)
 	static ac97_bus_ops_t ops = {
 		.write = snd_es1371_codec_write,
 		.read = snd_es1371_codec_read,
+		.wait = snd_es1371_codec_wait,
 	};
 
 	if ((err = snd_ac97_bus(card, 0, &ops, NULL, &pbus)) < 0)
-- 
cgit v1.2.3


From ef21ca24faf28df6d06939e77d5032a313490289 Mon Sep 17 00:00:00 2001
From: Nishanth Aravamudan <nacc@us.ibm.com>
Date: Sat, 9 Jul 2005 10:13:22 +0200
Subject: [ALSA] sound/pci: fix-up sleeping paths

ENS1370/1+ driver,ES1968 driver,Intel8x0 driver,VIA82xx driver
VIA82xx-modem driver,AC97 Codec,ALI5451 driver,CS46xx driver
MIXART driver,RME HDSP driver,Trident driver,YMFPCI driver
Description: Fix-up sleeping in sound/pci. These changes fall under the
following two categories:

        1) Replace schedule_timeout() with msleep() to guarantee the
        task delays as expected. This also involved replacing/removing
        custom sleep functions.
        2) Do not assume jiffies will only increment by one if you
        request a 1 jiffy sleep, i.e. use time_after/time_before in
        while loops.

Signed-off-by: Nishanth Aravamudan <nacc@us.ibm.com>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/ac97/ac97_codec.c      | 17 +++++++++--------
 sound/pci/ali5451/ali5451.c      |  4 ++--
 sound/pci/cs46xx/cs46xx_lib.c    | 15 +++++----------
 sound/pci/ens1370.c              | 12 +-----------
 sound/pci/es1968.c               | 14 ++++----------
 sound/pci/intel8x0.c             |  3 +--
 sound/pci/mixart/mixart.c        |  4 ++--
 sound/pci/rme9652/hdsp.c         |  6 ++----
 sound/pci/trident/trident_main.c |  3 +--
 sound/pci/via82xx.c              | 13 ++++++-------
 sound/pci/via82xx_modem.c        | 13 ++++++-------
 sound/pci/ymfpci/ymfpci_main.c   |  6 +++---
 12 files changed, 42 insertions(+), 68 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 0677d41239a..94cd989cff2 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2227,6 +2227,7 @@ void snd_ac97_restore_iec958(ac97_t *ac97)
 void snd_ac97_resume(ac97_t *ac97)
 {
 	int i;
+	unsigned long end_time;
 
 	if (ac97->bus->ops->reset) {
 		ac97->bus->ops->reset(ac97);
@@ -2244,26 +2245,26 @@ void snd_ac97_resume(ac97_t *ac97)
 	snd_ac97_write(ac97, AC97_POWERDOWN, ac97->regs[AC97_POWERDOWN]);
 	if (ac97_is_audio(ac97)) {
 		ac97->bus->ops->write(ac97, AC97_MASTER, 0x8101);
-		for (i = HZ/10; i >= 0; i--) {
+		end_time = jiffies + msecs_to_jiffies(100);
+		do {
 			if (snd_ac97_read(ac97, AC97_MASTER) == 0x8101)
 				break;
 			set_current_state(TASK_UNINTERRUPTIBLE);
 			schedule_timeout(1);
-		}
+		} while (time_after_eq(end_time, jiffies));
 		/* FIXME: extra delay */
 		ac97->bus->ops->write(ac97, AC97_MASTER, 0x8000);
-		if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000) {
-			set_current_state(TASK_UNINTERRUPTIBLE);
-			schedule_timeout(HZ/4);
-		}
+		if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000)
+			msleep(250);
 	} else {
-		for (i = HZ/10; i >= 0; i--) {
+		end_time = jiffies + msecs_to_jiffies(100);
+		do {
 			unsigned short val = snd_ac97_read(ac97, AC97_EXTENDED_MID);
 			if (val != 0xffff && (val & 1) != 0)
 				break;
 			set_current_state(TASK_UNINTERRUPTIBLE);
 			schedule_timeout(1);
-		}
+		} while (time_after_eq(end_time, jiffies));
 	}
 __reset_ready:
 
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index eb5c36d31a5..f08ae71f902 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -399,7 +399,7 @@ static int snd_ali_codec_ready(	ali_t *codec,
 	unsigned long end_time;
 	unsigned int res;
 	
-	end_time = jiffies + 10 * (HZ >> 2);
+	end_time = jiffies + 10 * msecs_to_jiffies(250);
 	do {
 		res = snd_ali_5451_peek(codec,port);
 		if (! (res & 0x8000))
@@ -422,7 +422,7 @@ static int snd_ali_stimer_ready(ali_t *codec, int sched)
 	dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER);
 	dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER);
 
-	end_time = jiffies + 10 * (HZ >> 2);
+	end_time = jiffies + 10 * msecs_to_jiffies(250);
 	do {
 		dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER);
 		if (dwChk2 != dwChk1)
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index fd4c50c88bc..ff28af1f658 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -2400,8 +2400,7 @@ static void snd_cs46xx_codec_reset (ac97_t * ac97)
 		if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05)
 			return;
 
-		set_current_state(TASK_UNINTERRUPTIBLE);
-		schedule_timeout(HZ/100);
+		msleep(10);
 	} while (time_after_eq(end_time, jiffies));
 
 	snd_printk("CS46xx secondary codec dont respond!\n");  
@@ -2435,8 +2434,7 @@ static int __devinit cs46xx_detect_codec(cs46xx_t *chip, int codec)
 			err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97[codec]);
 			return err;
 		}
-		set_current_state(TASK_INTERRUPTIBLE);
-		schedule_timeout(HZ/100);
+		msleep(10);
 	}
 	snd_printdd("snd_cs46xx: codec %d detection timeout\n", codec);
 	return -ENXIO;
@@ -3018,8 +3016,7 @@ static int snd_cs46xx_chip_init(cs46xx_t *chip)
 	/*
          *  Wait until the PLL has stabilized.
 	 */
-	set_current_state(TASK_UNINTERRUPTIBLE);
-	schedule_timeout(HZ/10); /* 100ms */
+	msleep(100);
 
 	/*
 	 *  Turn on clocking of the core so that we can setup the serial ports.
@@ -3072,8 +3069,7 @@ static int snd_cs46xx_chip_init(cs46xx_t *chip)
 		 */
 		if (snd_cs46xx_peekBA0(chip, BA0_ACSTS) & ACSTS_CRDY)
 			goto ok1;
-		set_current_state(TASK_UNINTERRUPTIBLE);
-		schedule_timeout((HZ+99)/100);
+		msleep(10);
 	}
 
 
@@ -3122,8 +3118,7 @@ static int snd_cs46xx_chip_init(cs46xx_t *chip)
 		 */
 		if ((snd_cs46xx_peekBA0(chip, BA0_ACISV) & (ACISV_ISV3 | ACISV_ISV4)) == (ACISV_ISV3 | ACISV_ISV4))
 			goto ok2;
-		set_current_state(TASK_UNINTERRUPTIBLE);
-		schedule_timeout((HZ+99)/100);
+		msleep(10);
 	}
 
 #ifndef CONFIG_SND_CS46XX_NEW_DSP
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index d4287338c04..78a81f3912a 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2018,21 +2018,11 @@ static int __devinit snd_ensoniq_create(snd_card_t * card,
 		if (pci->vendor == es1371_ac97_reset_hack[idx].vid &&
 		    pci->device == es1371_ac97_reset_hack[idx].did &&
 		    ensoniq->rev == es1371_ac97_reset_hack[idx].rev) {
-		        unsigned long tmo;
-			signed long tmo2;
-
 			ensoniq->cssr |= ES_1371_ST_AC97_RST;
 			outl(ensoniq->cssr, ES_REG(ensoniq, STATUS));
 			/* need to delay around 20ms(bleech) to give
 			some CODECs enough time to wakeup */
-			tmo = jiffies + (HZ / 50) + 1;
-			while (1) {
-				tmo2 = tmo - jiffies;
-				if (tmo2 <= 0)
-					break;
-				set_current_state(TASK_UNINTERRUPTIBLE);
-				schedule_timeout(tmo2);
-			}
+			msleep(20);
 			break;
 		}
 	/* AC'97 warm reset to start the bitclk */
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 327a341e276..9d7a2878393 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -664,11 +664,6 @@ static inline u16 maestro_read(es1968_t *chip, u16 reg)
 	return result;
 }
 
-#define big_mdelay(msec) do {\
-	set_current_state(TASK_UNINTERRUPTIBLE);\
-	schedule_timeout(((msec) * HZ + 999) / 1000);\
-} while (0)
-	
 /* Wait for the codec bus to be free */
 static int snd_es1968_ac97_wait(es1968_t *chip)
 {
@@ -1809,8 +1804,7 @@ static void __devinit es1968_measure_clock(es1968_t *chip)
 	snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR);
 	do_gettimeofday(&start_time);
 	spin_unlock_irq(&chip->reg_lock);
-	set_current_state(TASK_UNINTERRUPTIBLE);
-	schedule_timeout(HZ / 20); /* 50 msec */
+	msleep(50);
 	spin_lock_irq(&chip->reg_lock);
 	offset = __apu_get_register(chip, apu, 5);
 	do_gettimeofday(&stop_time);
@@ -2093,7 +2087,7 @@ static void snd_es1968_ac97_reset(es1968_t *chip)
 	outw(0x0000, ioaddr + 0x60);	/* write 0 to gpio 0 */
 	udelay(20);
 	outw(0x0001, ioaddr + 0x60);	/* write 1 to gpio 1 */
-	big_mdelay(20);
+	msleep(20);
 
 	outw(save_68 | 0x1, ioaddr + 0x68);	/* now restore .. */
 	outw((inw(ioaddr + 0x38) & 0xfffc) | 0x1, ioaddr + 0x38);
@@ -2109,7 +2103,7 @@ static void snd_es1968_ac97_reset(es1968_t *chip)
 	outw(0x0001, ioaddr + 0x60);	/* write 1 to gpio */
 	udelay(20);
 	outw(0x0009, ioaddr + 0x60);	/* write 9 to gpio */
-	big_mdelay(500);
+	msleep(500);
 	//outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38);
 	outw(inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a);
 	outw(inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c);
@@ -2135,7 +2129,7 @@ static void snd_es1968_ac97_reset(es1968_t *chip)
 
 		if (w > 10000) {
 			outb(inb(ioaddr + 0x37) | 0x08, ioaddr + 0x37);	/* do a software reset */
-			big_mdelay(500);	/* oh my.. */
+			msleep(500);	/* oh my.. */
 			outb(inb(ioaddr + 0x37) & ~0x08,
 				ioaddr + 0x37);
 			udelay(1);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index c3c3b68b454..7c806bd9cc9 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2464,8 +2464,7 @@ static void __devinit intel8x0_measure_ac97_clock(intel8x0_t *chip)
 	}
 	do_gettimeofday(&start_time);
 	spin_unlock_irq(&chip->reg_lock);
-	set_current_state(TASK_UNINTERRUPTIBLE);
-	schedule_timeout(HZ / 20);
+	msleep(50);
 	spin_lock_irq(&chip->reg_lock);
 	/* check the position */
 	pos = ichdev->fragsize1;
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 082c0d0f73d..6c868d91363 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -445,9 +445,9 @@ static int snd_mixart_trigger(snd_pcm_substream_t *subs, int cmd)
 
 static int mixart_sync_nonblock_events(mixart_mgr_t *mgr)
 {
-	int timeout = HZ;
+	unsigned long timeout = jiffies + HZ;
 	while (atomic_read(&mgr->msg_processed) > 0) {
-		if (! timeout--) {
+		if (time_after(jiffies, timeout)) {
 			snd_printk(KERN_ERR "mixart: cannot process nonblock events!\n");
 			return -EBUSY;
 		}
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 0db558a9287..796621de500 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -679,8 +679,7 @@ static int snd_hdsp_load_firmware_from_cache(hdsp_t *hdsp) {
 		}
 
 		if ((1000 / HZ) < 3000) {
-			set_current_state(TASK_UNINTERRUPTIBLE);
-			schedule_timeout((3000 * HZ + 999) / 1000);
+			ssleep(3);
 		} else {
 			mdelay(3000);
 		}
@@ -5080,8 +5079,7 @@ static int __devinit snd_hdsp_create(snd_card_t *card,
 	if (!is_9652 && !is_9632) {
 		/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
  		if ((1000 / HZ) < 2000) {
-			set_current_state(TASK_UNINTERRUPTIBLE);
-			schedule_timeout((2000 * HZ + 999) / 1000);
+			ssleep(2);
 		} else {
 			mdelay(2000);
 		}
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index b01c91bb5f6..29d89bfba0a 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3153,8 +3153,7 @@ static int snd_trident_gameport_open(struct gameport *gameport, int mode)
 	switch (mode) {
 		case GAMEPORT_MODE_COOKED:
 			outb(GAMEPORT_MODE_ADC, TRID_REG(chip, GAMEPORT_GCR));
-			set_current_state(TASK_UNINTERRUPTIBLE);
-			schedule_timeout(1 + 20 * HZ / 1000); /* 20msec */
+			msleep(20);
 			return 0;
 		case GAMEPORT_MODE_RAW:
 			outb(0, TRID_REG(chip, GAMEPORT_GCR));
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 064972b14d0..890582ce874 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -547,8 +547,7 @@ static void snd_via82xx_codec_wait(ac97_t *ac97)
 	int err;
 	err = snd_via82xx_codec_ready(chip, ac97->num);
 	/* here we need to wait fairly for long time.. */
-	set_current_state(TASK_UNINTERRUPTIBLE);
-	schedule_timeout(HZ/2);
+	msleep(500);
 }
 
 static void snd_via82xx_codec_write(ac97_t *ac97,
@@ -1847,7 +1846,7 @@ static void __devinit snd_via82xx_proc_init(via82xx_t *chip)
 static int snd_via82xx_chip_init(via82xx_t *chip)
 {
 	unsigned int val;
-	int max_count;
+	unsigned long end_time;
 	unsigned char pval;
 
 #if 0 /* broken on K7M? */
@@ -1889,14 +1888,14 @@ static int snd_via82xx_chip_init(via82xx_t *chip)
 	}
 
 	/* wait until codec ready */
-	max_count = ((3 * HZ) / 4) + 1;
+	end_time = jiffies + msecs_to_jiffies(750);
 	do {
 		pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
 		if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */
 			break;
 		set_current_state(TASK_UNINTERRUPTIBLE);
 		schedule_timeout(1);
-	} while (--max_count > 0);
+	} while (time_before(jiffies, end_time));
 
 	if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)
 		snd_printk("AC'97 codec is not ready [0x%x]\n", val);
@@ -1905,7 +1904,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip)
 	snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
 				 VIA_REG_AC97_SECONDARY_VALID |
 				 (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
-	max_count = ((3 * HZ) / 4) + 1;
+	end_time = jiffies + msecs_to_jiffies(750);
 	snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
 				 VIA_REG_AC97_SECONDARY_VALID |
 				 (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
@@ -1916,7 +1915,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip)
 		}
 		set_current_state(TASK_INTERRUPTIBLE);
 		schedule_timeout(1);
-	} while (--max_count > 0);
+	} while (time_before(jiffies, end_time));
 	/* This is ok, the most of motherboards have only one codec */
 
       __ac97_ok2:
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5896d289f9a..4a9779cc973 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -408,8 +408,7 @@ static void snd_via82xx_codec_wait(ac97_t *ac97)
 	int err;
 	err = snd_via82xx_codec_ready(chip, ac97->num);
 	/* here we need to wait fairly for long time.. */
-	set_current_state(TASK_UNINTERRUPTIBLE);
-	schedule_timeout(HZ/2);
+	msleep(500);
 }
 
 static void snd_via82xx_codec_write(ac97_t *ac97,
@@ -923,7 +922,7 @@ static void __devinit snd_via82xx_proc_init(via82xx_t *chip)
 static int snd_via82xx_chip_init(via82xx_t *chip)
 {
 	unsigned int val;
-	int max_count;
+	unsigned long end_time;
 	unsigned char pval;
 
 	pci_read_config_byte(chip->pci, VIA_MC97_CTRL, &pval);
@@ -962,14 +961,14 @@ static int snd_via82xx_chip_init(via82xx_t *chip)
 	}
 
 	/* wait until codec ready */
-	max_count = ((3 * HZ) / 4) + 1;
+	end_time = jiffies + msecs_to_jiffies(750);
 	do {
 		pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
 		if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */
 			break;
 		set_current_state(TASK_UNINTERRUPTIBLE);
 		schedule_timeout(1);
-	} while (--max_count > 0);
+	} while (time_before(jiffies, end_time));
 
 	if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)
 		snd_printk("AC'97 codec is not ready [0x%x]\n", val);
@@ -977,7 +976,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip)
 	snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
 				 VIA_REG_AC97_SECONDARY_VALID |
 				 (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
-	max_count = ((3 * HZ) / 4) + 1;
+	end_time = jiffies + msecs_to_jiffies(750);
 	snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
 				 VIA_REG_AC97_SECONDARY_VALID |
 				 (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
@@ -988,7 +987,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip)
 		}
 		set_current_state(TASK_INTERRUPTIBLE);
 		schedule_timeout(1);
-	} while (--max_count > 0);
+	} while (time_before(jiffies, end_time));
 	/* This is ok, the most of motherboards have only one codec */
 
       __ac97_ok2:
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 2ae79610ecb..d54f88a1b52 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -84,16 +84,16 @@ static inline void snd_ymfpci_writel(ymfpci_t *chip, u32 offset, u32 val)
 
 static int snd_ymfpci_codec_ready(ymfpci_t *chip, int secondary)
 {
-	signed long end_time;
+	unsigned long end_time;
 	u32 reg = secondary ? YDSXGR_SECSTATUSADR : YDSXGR_PRISTATUSADR;
 	
-	end_time = (jiffies + ((3 * HZ) / 4)) + 1;
+	end_time = jiffies + msecs_to_jiffies(750);
 	do {
 		if ((snd_ymfpci_readw(chip, reg) & 0x8000) == 0)
 			return 0;
 		set_current_state(TASK_UNINTERRUPTIBLE);
 		schedule_timeout(1);
-	} while (end_time - (signed long)jiffies >= 0);
+	} while (time_before(jiffies, end_time));
 	snd_printk("codec_ready: codec %i is not ready [0x%x]\n", secondary, snd_ymfpci_readw(chip, reg));
 	return -EBUSY;
 }
-- 
cgit v1.2.3


From 7c1d549aa9b22365fe5405c372f840cdbc6315f5 Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Sun, 10 Jul 2005 11:50:36 +0200
Subject: [ALSA] emu10k1: Add EMU 1212m card entry and document it as not
 supported yet.

EMU10K1/EMU10K2 driver

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/emu10k1/emu10k1_main.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7c31d9b3024..746b51ef396 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -630,6 +630,13 @@ static emu_chip_details_t emu_chip_details[] = {
 	 .emu10k2_chip = 1,
 	 .ca0108_chip = 1,
 	 .ac97_chip = 1} ,
+	/* Tested by James@superbug.co.uk 8th July 2005. No sound available yet. */
+	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102,
+	 .driver = "Audigy2", .name = "E-mu 1212m [4001]", 
+	 .id = "EMU1212m",
+	 .emu10k2_chip = 1,
+	 .ca0102_chip = 1,
+	 .ecard = 1} ,
 	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102,
 	 .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", 
-- 
cgit v1.2.3


From c9eab129fcbcef364b34fb3a70cb2531847e1edf Mon Sep 17 00:00:00 2001
From: James Courtier-Dutton <James@superbug.co.uk>
Date: Sun, 10 Jul 2005 12:04:29 +0200
Subject: [ALSA] ac97: Fix volume control bit size detection for STAC9704.

AC97 Codec

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
---
 sound/pci/ac97/ac97_codec.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 94cd989cff2..1f09653dc0f 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1078,6 +1078,11 @@ static void check_volume_resolution(ac97_t *ac97, int reg, unsigned char *lo_max
 	for (i = 0 ; i < ARRAY_SIZE(cbit); i++) {
 		unsigned short val;
 		snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8));
+		/* Do the read twice due to buffers on some ac97 codecs.
+		 * e.g. The STAC9704 returns exactly what you wrote the the register
+		 * if you read it immediately. This causes the detect routine to fail.
+		 */
+		val = snd_ac97_read(ac97, reg);
 		val = snd_ac97_read(ac97, reg);
 		if (! *lo_max && (val & 0x7f) == cbit[i])
 			*lo_max = max[i];
-- 
cgit v1.2.3


From 7858ffa062886706026cfff3ba80b8400b520501 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 11 Jul 2005 15:37:19 +0200
Subject: [ALSA] ac97 - remove unused variable

AC97 Codec
remove a variable made obsolete by the last change

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/pci/ac97/ac97_codec.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 1f09653dc0f..6983eea226d 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2231,7 +2231,6 @@ void snd_ac97_restore_iec958(ac97_t *ac97)
  */
 void snd_ac97_resume(ac97_t *ac97)
 {
-	int i;
 	unsigned long end_time;
 
 	if (ac97->bus->ops->reset) {
-- 
cgit v1.2.3


From fb92e6f05e84f6c217d786208e2ed5acf633b6ce Mon Sep 17 00:00:00 2001
From: Nicolas Graziano <nicolas.graziano@wanadoo.fr>
Date: Wed, 27 Jul 2005 17:25:08 +0200
Subject: [ALSA] hda driver, correct bug in model 'auto'

HDA Codec driver
- Correct some index variable inversion in patch_cmedia.c

Signed-off-by: Nicolas Graziano <nicolas.graziano@wanadoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cmedia.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 2d6e3e3d0a3..86f195f19ee 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -408,7 +408,7 @@ static int cmi9880_fill_multi_dac_nids(struct hda_codec *codec, const struct aut
 		/* search for an empty channel */
 		for (j = 0; j < cfg->line_outs; j++) {
 			if (! assigned[j]) {
-				spec->dac_nids[i] = i + 0x03;
+				spec->dac_nids[i] = j + 0x03;
 				assigned[j] = 1;
 				break;
 			}
@@ -444,11 +444,10 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
 			len = snd_hda_get_connections(codec, nid, conn, 4);
 			for (k = 0; k < len; k++)
 				if (conn[k] == spec->dac_nids[i]) {
-					spec->multi_init[j].param = j;
+					spec->multi_init[j].param = k;
 					break;
 				}
 			j++;
-			break;
 		}
 	}
 	return 0;
-- 
cgit v1.2.3


From eeacb5457cf5f0802fb29f385befa0b1d166cadb Mon Sep 17 00:00:00 2001
From: Sergey Ulanov <Sergey.Ulanov@asplinux.ru>
Date: Wed, 27 Jul 2005 17:28:58 +0200
Subject: [ALSA] Jack Sense support for AD1980 and AD1888

AC97 Codec
Attached patch adds 'Jack Sense' controls for AD1980 and AD1888 chips.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_patch.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index a15eb8522b7..66edc857d3e 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1528,6 +1528,9 @@ static const snd_kcontrol_new_t snd_ac97_ad1888_controls[] = {
 	},
 	AC97_SURROUND_JACK_MODE_CTL,
 	AC97_CHANNEL_MODE_CTL,
+
+	AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 10, 1, 0),
+	AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0),
 };
 
 static int patch_ad1888_specific(ac97_t *ac97)
-- 
cgit v1.2.3


From 69c3e5f8562c7854d9dd8d7820a89286f9440e41 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 27 Jul 2005 17:30:14 +0200
Subject: [ALSA] via82xx - Fix dxs_support of twinhead laptop

VIA82xx driver
Changed the dxs_support value of twinhead laptop to DXS_SRC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/via82xx.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 890582ce874..4889600387c 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2177,7 +2177,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci)
 		{ .subvendor = 0x147b, .subdevice = 0x1413, .action = VIA_DXS_ENABLE }, /* ABIT KV8 Pro */
 		{ .subvendor = 0x147b, .subdevice = 0x1415, .action = VIA_DXS_NO_VRA }, /* Abit AV8 */
 		{ .subvendor = 0x14ff, .subdevice = 0x0403, .action = VIA_DXS_ENABLE }, /* Twinhead mobo */
-		{ .subvendor = 0x14ff, .subdevice = 0x0408, .action = VIA_DXS_NO_VRA }, /* Twinhead mobo */
+		{ .subvendor = 0x14ff, .subdevice = 0x0408, .action = VIA_DXS_SRC }, /* Twinhead laptop */
 		{ .subvendor = 0x1584, .subdevice = 0x8120, .action = VIA_DXS_ENABLE }, /* Gericom/Targa/Vobis/Uniwill laptop */
 		{ .subvendor = 0x1584, .subdevice = 0x8123, .action = VIA_DXS_NO_VRA }, /* Uniwill (Targa Visionary XP-210) */
 		{ .subvendor = 0x161f, .subdevice = 0x202b, .action = VIA_DXS_NO_VRA }, /* Amira Note book */
-- 
cgit v1.2.3