From 8fc6d4186e0a60b3755a6b88bf67a3ac3214dcc3 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela
Date: Wed, 2 Jun 2010 13:29:17 +0200
Subject: ALSA: hda-intel - fix wallclk variable update and condition

This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.

It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/hda_intel.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index dc79564fea30..af701a894687 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1913,11 +1913,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 	if (WARN_ONCE(!azx_dev->period_bytes,
 		      "hda-intel: zero azx_dev->period_bytes"))
 		return -1; /* this shouldn't happen! */
-	if (wallclk <= azx_dev->period_wallclk &&
+	if (wallclk < (azx_dev->period_wallclk * 5) / 4 &&
 	    pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		/* NG - it's below the first next period boundary */
 		return bdl_pos_adj[chip->dev_index] ? 0 : -1;
-	azx_dev->start_wallclk = wallclk;
+	azx_dev->start_wallclk += wallclk;
 	return 1; /* OK, it's fine */
 }
 
-- 
cgit v1.2.3


From c9ff921abecda352e987a6aae169118a3fc9aa5d Mon Sep 17 00:00:00 2001
From: Andy Shevchenko
Date: Tue, 15 Jun 2010 17:26:28 +0300
Subject: ALSA: alsa: riptide: don't use own hex_to_bin() method

Signed-off-by: Andy Shevchenko <ext-andriy.shevchenko@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/riptide/riptide.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index ad4462677615..59d79962f236 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -97,6 +97,7 @@
 #include <linux/gameport.h>
 #include <linux/device.h>
 #include <linux/firmware.h>
+#include <linux/kernel.h>
 #include <asm/io.h>
 #include <sound/core.h>
 #include <sound/info.h>
@@ -667,13 +668,12 @@ static u32 atoh(const unsigned char *in, unsigned int len)
 	unsigned char c;
 
 	while (len) {
+		int value;
+
 		c = in[len - 1];
-		if ((c >= '0') && (c <= '9'))
-			sum += mult * (c - '0');
-		else if ((c >= 'A') && (c <= 'F'))
-			sum += mult * (c - ('A' - 10));
-		else if ((c >= 'a') && (c <= 'f'))
-			sum += mult * (c - ('a' - 10));
+		value = hex_to_bin(c);
+		if (value >= 0)
+			sum += mult * value;
 		mult *= 16;
 		--len;
 	}
-- 
cgit v1.2.3


From f7154de220f14073ef0d76638f85e254ad2e202f Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski
Date: Thu, 17 Jun 2010 14:15:06 -0300
Subject: ALSA: hda - add ideapad model for Conexant 5051 codec

Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b,
which isn't muted when headphone is plugged in. This adds additional
support to the extra subwoofer via new ideapad model.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/HD-Audio-Models.txt |  1 +
 sound/pci/hda/patch_conexant.c               | 20 ++++++++++++++++++++
 2 files changed, 21 insertions(+)

(limited to 'sound/pci')

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 1d38b0dfba95..84e81ad7ba2d 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -282,6 +282,7 @@ Conexant 5051
   hp		HP Spartan laptop
   hp-dv6736	HP dv6736
   hp-f700	HP Compaq Presario F700
+  ideapad	Lenovo IdeaPad laptop
   lenovo-x200	Lenovo X200 laptop
   toshiba	Toshiba Satellite M300
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2bf2cb5da956..54f74191ebca 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec)
 	pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
 	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			    pinctl);
+	/* on ideapad there is an aditional speaker (subwoofer) to mute */
+	if (spec->ideapad)
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+				    pinctl);
 }
 
 /* turn on/off EAPD (+ mute HP) as a master switch */
@@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid,
 #endif
 }
 
+static struct hda_verb cxt5051_ideapad_init_verbs[] = {
+	/* Subwoofer */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ } /* end */
+};
+
 /* initialize jack-sensing, too */
 static int cxt5051_init(struct hda_codec *codec)
 {
@@ -1917,6 +1929,7 @@ enum {
 	CXT5051_LENOVO_X200,	/* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */
 	CXT5051_F700,       /* HP Compaq Presario F700 */
 	CXT5051_TOSHIBA,	/* Toshiba M300 & co */
+	CXT5051_IDEAPAD,	/* Lenovo IdeaPad Y430 */
 	CXT5051_MODELS
 };
 
@@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_LENOVO_X200]	= "lenovo-x200",
 	[CXT5051_F700]          = "hp-700",
 	[CXT5051_TOSHIBA]	= "toshiba",
+	[CXT5051_IDEAPAD]	= "ideapad",
 };
 
 static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
@@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
+	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD),
 	{}
 };
 
@@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec)
 		spec->mixers[0] = cxt5051_toshiba_mixers;
 		spec->auto_mic = AUTO_MIC_PORTB;
 		break;
+	case CXT5051_IDEAPAD:
+		spec->init_verbs[spec->num_init_verbs++] =
+			cxt5051_ideapad_init_verbs;
+		spec->ideapad = 1;
+		break;
 	}
 
 	return 0;
-- 
cgit v1.2.3


From d1eb57f47b7f524c13112c891e87fb1f51029fd1 Mon Sep 17 00:00:00 2001
From: Kailang Yang
Date: Wed, 23 Jun 2010 16:25:26 +0200
Subject: ALSA: hda - Support ALC680 codec

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 331 +++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 330 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f1ce7d7f5aa3..630e66743e8e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -18612,7 +18612,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 
 	add_verb(spec, alc662_init_verbs);
 	if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
-	    codec->vendor_id == 0x10ec0665)
+	    codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
 		add_verb(spec, alc663_init_verbs);
 
 	if (codec->vendor_id == 0x10ec0272)
@@ -18755,6 +18755,334 @@ static int patch_alc888(struct hda_codec *codec)
 	return patch_alc882(codec);
 }
 
+/*
+ * ALC680 support
+ */
+#define ALC680_DIGOUT_NID	ALC880_DIGOUT_NID
+#define alc680_modes		alc260_modes
+
+static hda_nid_t alc680_dac_nids[3] = {
+	/* Lout1, Lout2, hp */
+	0x02, 0x03, 0x04
+};
+
+static hda_nid_t alc680_adc_nids[3] = {
+	/* ADC0-2 */
+	/* DMIC, MIC, Line-in*/
+	0x07, 0x08, 0x09
+};
+
+static struct snd_kcontrol_new alc680_base_mixer[] = {
+	/* output mixer control */
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	{ }
+};
+
+static struct snd_kcontrol_new alc680_capture_mixer[] = {
+	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc680_init_verbs[] = {
+	/* Unmute DAC0-1 and set vol = 0 */
+	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{ }
+};
+
+/* create input playback/capture controls for the given pin */
+static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
+				    const char *ctlname, int idx)
+{
+	hda_nid_t dac;
+	int err;
+
+	switch (nid) {
+	case 0x14:
+		dac = 0x02;
+		break;
+	case 0x15:
+		dac = 0x03;
+		break;
+	case 0x16:
+		dac = 0x04;
+		break;
+	default:
+		return 0;
+	}
+	if (spec->multiout.dac_nids[0] != dac &&
+	    spec->multiout.dac_nids[1] != dac) {
+		err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname,
+				  HDA_COMPOSE_AMP_VAL(dac, 3, idx,
+						      HDA_OUTPUT));
+		if (err < 0)
+			return err;
+
+		err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
+			  HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
+
+		if (err < 0)
+			return err;
+		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+	}
+
+	return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int alc680_auto_create_multi_out_ctls(struct alc_spec *spec,
+					     const struct auto_pin_cfg *cfg)
+{
+	hda_nid_t nid;
+	int err;
+
+	spec->multiout.dac_nids = spec->private_dac_nids;
+
+	nid = cfg->line_out_pins[0];
+	if (nid) {
+		const char *name;
+		if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+			name = "Speaker";
+		else
+			name = "Front";
+		err = alc680_new_analog_output(spec, nid, name, 0);
+		if (err < 0)
+			return err;
+	}
+
+	nid = cfg->speaker_pins[0];
+	if (nid) {
+		err = alc680_new_analog_output(spec, nid, "Speaker", 0);
+		if (err < 0)
+			return err;
+	}
+	nid = cfg->hp_pins[0];
+	if (nid) {
+		err = alc680_new_analog_output(spec, nid, "Headphone", 0);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static void alc680_auto_set_output_and_unmute(struct hda_codec *codec,
+					      hda_nid_t nid, int pin_type)
+{
+	alc_set_pin_output(codec, nid, pin_type);
+}
+
+static void alc680_auto_init_multi_out(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid = spec->autocfg.line_out_pins[0];
+	if (nid) {
+		int pin_type = get_pin_type(spec->autocfg.line_out_type);
+		alc680_auto_set_output_and_unmute(codec, nid, pin_type);
+	}
+}
+
+static void alc680_auto_init_hp_out(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin;
+
+	pin = spec->autocfg.hp_pins[0];
+	if (pin)
+		alc680_auto_set_output_and_unmute(codec, pin, PIN_HP);
+	pin = spec->autocfg.speaker_pins[0];
+	if (pin)
+		alc680_auto_set_output_and_unmute(codec, pin, PIN_OUT);
+}
+
+/* pcm configuration: identical with ALC880 */
+#define alc680_pcm_analog_playback	alc880_pcm_analog_playback
+#define alc680_pcm_analog_capture	alc880_pcm_analog_capture
+#define alc680_pcm_analog_alt_capture	alc880_pcm_analog_alt_capture
+#define alc680_pcm_digital_playback	alc880_pcm_digital_playback
+
+static struct hda_input_mux alc680_capture_source = {
+	.num_items = 1,
+	.items = {
+		{ "Mic", 0x0 },
+	},
+};
+
+/*
+ * BIOS auto configuration
+ */
+static int alc680_parse_auto_config(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int err;
+	static hda_nid_t alc680_ignore[] = { 0 };
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+					   alc680_ignore);
+	if (err < 0)
+		return err;
+	if (!spec->autocfg.line_outs) {
+		if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+			spec->multiout.max_channels = 2;
+			spec->no_analog = 1;
+			goto dig_only;
+		}
+		return 0; /* can't find valid BIOS pin config */
+	}
+	err = alc680_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+
+	spec->multiout.max_channels = 2;
+
+ dig_only:
+	/* digital only support output */
+	if (spec->autocfg.dig_outs) {
+		spec->multiout.dig_out_nid = ALC680_DIGOUT_NID;
+		spec->dig_out_type = spec->autocfg.dig_out_type[0];
+	}
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
+
+	add_verb(spec, alc680_init_verbs);
+	spec->num_mux_defs = 1;
+	spec->input_mux = &alc680_capture_source;
+
+	err = alc_auto_add_mic_boost(codec);
+	if (err < 0)
+		return err;
+
+	return 1;
+}
+
+#define alc680_auto_init_analog_input	alc882_auto_init_analog_input
+
+/* init callback for auto-configuration model -- overriding the default init */
+static void alc680_auto_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	alc680_auto_init_multi_out(codec);
+	alc680_auto_init_hp_out(codec);
+	alc680_auto_init_analog_input(codec);
+	if (spec->unsol_event)
+		alc_inithook(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char *alc680_models[ALC680_MODEL_LAST] = {
+	[ALC680_BASE]		= "alc680_base",
+};
+
+static struct snd_pci_quirk alc680_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
+	{}
+};
+
+static struct alc_config_preset alc680_presets[] = {
+	[ALC680_BASE] = {
+		.mixers = { alc680_base_mixer },
+		.cap_mixer =  alc680_capture_mixer,
+		.init_verbs = { alc680_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc680_dac_nids),
+		.dac_nids = alc680_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc680_adc_nids),
+		.adc_nids = alc680_adc_nids,
+		.hp_nid = 0x04,
+		.dig_out_nid = ALC680_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc680_modes),
+		.channel_mode = alc680_modes,
+		.input_mux = &alc680_capture_source,
+	},
+};
+
+static int patch_alc680(struct hda_codec *codec)
+{
+	struct alc_spec *spec;
+	int board_config;
+	int err;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	board_config = snd_hda_check_board_config(codec, ALC680_MODEL_LAST,
+						  alc680_models,
+						  alc680_cfg_tbl);
+
+	if (board_config < 0 || board_config >= ALC680_MODEL_LAST) {
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
+		board_config = ALC680_AUTO;
+	}
+
+	if (board_config == ALC680_AUTO) {
+		/* automatic parse from the BIOS config */
+		err = alc680_parse_auto_config(codec);
+		if (err < 0) {
+			alc_free(codec);
+			return err;
+		} else if (!err) {
+			printk(KERN_INFO
+			       "hda_codec: Cannot set up configuration "
+			       "from BIOS.  Using base mode...\n");
+			board_config = ALC680_BASE;
+		}
+	}
+
+	if (board_config != ALC680_AUTO)
+		setup_preset(codec, &alc680_presets[board_config]);
+
+	spec->stream_analog_playback = &alc680_pcm_analog_playback;
+	spec->stream_analog_capture = &alc680_pcm_analog_capture;
+	spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture;
+	spec->stream_digital_playback = &alc680_pcm_digital_playback;
+
+	if (!spec->adc_nids) {
+		spec->adc_nids = alc680_adc_nids;
+		spec->num_adc_nids = ARRAY_SIZE(alc680_adc_nids);
+	}
+
+	if (!spec->cap_mixer)
+		set_capture_mixer(codec);
+
+	spec->vmaster_nid = 0x02;
+
+	codec->patch_ops = alc_patch_ops;
+	if (board_config == ALC680_AUTO)
+		spec->init_hook = alc680_auto_init;
+
+	return 0;
+}
+
 /*
  * patch entries
  */
@@ -18779,6 +19107,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
 	{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
 	{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
+	{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
 	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
 	{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
-- 
cgit v1.2.3


From d4a86d81944d3cccb3f4a309230e835823a61252 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 23 Jun 2010 17:51:26 +0200
Subject: ALSA: hda - Add missing ALC680_* definitions

Also update the documentation.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/HD-Audio-Models.txt |  5 +++++
 sound/pci/hda/patch_realtek.c                | 10 +++++++++-
 2 files changed, 14 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 84e81ad7ba2d..03771d7c5dd7 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -114,6 +114,11 @@ ALC662/663/272
   samsung-nc10	Samsung NC10 mini notebook
   auto		auto-config reading BIOS (default)
 
+ALC680
+======
+  base		Base model (ASUS NX90)
+  auto		auto-config reading BIOS (default)
+
 ALC882/883/885/888/889
 ======================
   3stack-dig	3-jack with SPDIF I/O
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 630e66743e8e..9b15a46e3ccc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -256,6 +256,13 @@ enum {
 	ALC882_MODEL_LAST,
 };
 
+/* ALC680 models */
+enum {
+	ALC680_BASE,
+	ALC680_AUTO,
+	ALC680_MODEL_LAST,
+};
+
 /* for GPIO Poll */
 #define GPIO_MASK	0x03
 
@@ -18997,7 +19004,8 @@ static void alc680_auto_init(struct hda_codec *codec)
  * configuration and preset
  */
 static const char *alc680_models[ALC680_MODEL_LAST] = {
-	[ALC680_BASE]		= "alc680_base",
+	[ALC680_BASE]		= "base",
+	[ALC680_AUTO]		= "auto",
 };
 
 static struct snd_pci_quirk alc680_cfg_tbl[] = {
-- 
cgit v1.2.3


From 3a3d5fd125f82200019ef406c4d51ba4d9f0a604 Mon Sep 17 00:00:00 2001
From: David Dillow
Date: Sun, 27 Jun 2010 00:04:32 +0200
Subject: sis7019: fix capture issues with multiple periods per buffer

When using a timing voice to clock out periods during capture, the
driver would slowly loose synchronization and never catch up, eventually
reaching a point where it no longer generated interrupts. To avoid
this situation, the virtual period clocking was changed to shorten the
next timing period when our timing voice falls too far behind the
capture voice. In addition, the first virtual period for the timing
voice was slightly too short, causing the timing voice to initially be
ahead of the capture voice.

While tracking down this problem, I noticed that the expected sample
offset was being incorrectly initialized, causing an overrun to be
incorrectly reported when the timing voice happened to be perfectly
synchronized.

Reported-by: Hans Schou <linux@schou.dk>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/sis7019.c | 10 ++++++----
 1 file changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 9cc1b5aa0148..614ff6e514fd 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -264,11 +264,13 @@ static void sis_update_voice(struct voice *voice)
 		 * if using small periods.
 		 *
 		 * If we're less than 9 samples behind, we're on target.
+		 * Otherwise, shorten the next vperiod by the amount we've
+		 * been delayed.
 		 */
 		if (sync > -9)
 			voice->vperiod = voice->sync_period_size + 1;
 		else
-			voice->vperiod = voice->sync_period_size - 4;
+			voice->vperiod = voice->sync_period_size + sync + 10;
 
 		if (voice->vperiod < voice->buffer_size) {
 			sis_update_sso(voice, voice->vperiod);
@@ -736,7 +738,7 @@ static void sis_prepare_timing_voice(struct voice *voice,
 	period_size = buffer_size;
 
 	/* Initially, we want to interrupt just a bit behind the end of
-	 * the period we're clocking out. 10 samples seems to give a good
+	 * the period we're clocking out. 12 samples seems to give a good
 	 * delay.
 	 *
 	 * We want to spread our interrupts throughout the virtual period,
@@ -747,7 +749,7 @@ static void sis_prepare_timing_voice(struct voice *voice,
 	 *
 	 * This is all moot if we don't need to use virtual periods.
 	 */
-	vperiod = runtime->period_size + 10;
+	vperiod = runtime->period_size + 12;
 	if (vperiod > period_size) {
 		u16 tail = vperiod % period_size;
 		u16 quarter_period = period_size / 4;
@@ -776,7 +778,7 @@ static void sis_prepare_timing_voice(struct voice *voice,
 	 */
 	timing->flags |= VOICE_SYNC_TIMING;
 	timing->sync_base = voice->ctrl_base;
-	timing->sync_cso = runtime->period_size - 1;
+	timing->sync_cso = runtime->period_size;
 	timing->sync_period_size = runtime->period_size;
 	timing->sync_buffer_size = runtime->buffer_size;
 	timing->period_size = period_size;
-- 
cgit v1.2.3


From 08b450988905505d12f7671bc24b8da73631d327 Mon Sep 17 00:00:00 2001
From: David Dillow
Date: Sun, 27 Jun 2010 00:07:57 +0200
Subject: sis7019: increase reset delays

A few boards using this controller are reported to need a little extra
time during their reset cycle.

Reported-by: Michael Goeke <michael.goeke@icachip.de>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/sis7019.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 614ff6e514fd..1b8f6742b5fa 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1049,7 +1049,7 @@ static int sis_chip_free(struct sis7019 *sis)
 	/* Reset the chip, and disable all interrputs.
 	 */
 	outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR);
-	udelay(10);
+	udelay(25);
 	outl(0, sis->ioport + SIS_GCR);
 	outl(0, sis->ioport + SIS_GIER);
 
@@ -1085,7 +1085,7 @@ static int sis_chip_init(struct sis7019 *sis)
 	/* Reset the audio controller
 	 */
 	outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR);
-	udelay(10);
+	udelay(25);
 	outl(0, io + SIS_GCR);
 
 	/* Get the AC-link semaphore, and reset the codecs
@@ -1098,7 +1098,7 @@ static int sis_chip_init(struct sis7019 *sis)
 		return -EIO;
 
 	outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD);
-	udelay(10);
+	udelay(250);
 
 	count = 0xffff;
 	while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
-- 
cgit v1.2.3


From 168f1b07ccc0e8edecb67fab2d0670861853e2fd Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Tue, 6 Jul 2010 08:37:06 +1200
Subject: ALSA: asihpi - HPI API updates

Remove some deprecated items.
Change compander api to one function per parameter.
Add a version string define.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c   | 10 +++----
 sound/pci/asihpi/hpi.h      | 68 ++++++++++++++++++++++++++++++++-------------
 sound/pci/asihpi/hpidebug.h |  4 +--
 3 files changed, 55 insertions(+), 27 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 1db586af4f9c..91218f77217f 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -1383,7 +1383,7 @@ static char *asihpi_src_names[] =
 
 compile_time_assert(
 	(ARRAY_SIZE(asihpi_src_names) ==
-		(HPI_SOURCENODE_LAST_INDEX-HPI_SOURCENODE_BASE+1)),
+		(HPI_SOURCENODE_LAST_INDEX-HPI_SOURCENODE_NONE+1)),
 	assert_src_names_size);
 
 #if ASI_STYLE_NAMES
@@ -1414,7 +1414,7 @@ static char *asihpi_dst_names[] =
 
 compile_time_assert(
 	(ARRAY_SIZE(asihpi_dst_names) ==
-		(HPI_DESTNODE_LAST_INDEX-HPI_DESTNODE_BASE+1)),
+		(HPI_DESTNODE_LAST_INDEX-HPI_DESTNODE_NONE+1)),
 	assert_dst_names_size);
 
 static inline int ctl_add(struct snd_card *card, struct snd_kcontrol_new *ctl,
@@ -2171,7 +2171,7 @@ static int snd_asihpi_mux_info(struct snd_kcontrol *kcontrol,
 					&src_node_type, &src_node_index);
 
 	sprintf(uinfo->value.enumerated.name, "%s %d",
-		asihpi_src_names[src_node_type - HPI_SOURCENODE_BASE],
+		asihpi_src_names[src_node_type - HPI_SOURCENODE_NONE],
 		src_node_index);
 	return 0;
 }
@@ -2603,8 +2603,8 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 
 		}
 
-		hpi_ctl.src_node_type -= HPI_SOURCENODE_BASE;
-		hpi_ctl.dst_node_type -= HPI_DESTNODE_BASE;
+		hpi_ctl.src_node_type -= HPI_SOURCENODE_NONE;
+		hpi_ctl.dst_node_type -= HPI_DESTNODE_NONE;
 
 		/* ASI50xx in SSX mode has multiple meters on the same node.
 		   Use subindex to create distinct ALSA controls
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 0173bbe62b67..cee4df460f68 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -50,7 +50,8 @@ i.e 3.05.02 is a development version
 #define HPI_VER_RELEASE(v) ((int)(v & 0xFF))
 
 /* Use single digits for versions less that 10 to avoid octal. */
-#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 25)
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 36)
+#define HPI_VER_STRING "4.03.36"
 
 /* Library version as documented in hpi-api-versions.txt */
 #define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(9, 0, 0)
@@ -203,8 +204,6 @@ enum HPI_SOURCENODES {
 	exists on a destination node can be searched for using a source
 	node value of either 0, or HPI_SOURCENODE_NONE */
 	HPI_SOURCENODE_NONE = 100,
-	/** \deprecated Use HPI_SOURCENODE_NONE instead. */
-	HPI_SOURCENODE_BASE = 100,
 	/** Out Stream (Play) node. */
 	HPI_SOURCENODE_OSTREAM = 101,
 	/** Line in node - could be analog, AES/EBU or network. */
@@ -235,8 +234,6 @@ enum HPI_DESTNODES {
 	exists on a source node can be searched for using a destination
 	node value of either 0, or HPI_DESTNODE_NONE */
 	HPI_DESTNODE_NONE = 200,
-	/** \deprecated Use HPI_DESTNODE_NONE instead. */
-	HPI_DESTNODE_BASE = 200,
 	/** In Stream (Record) node. */
 	HPI_DESTNODE_ISTREAM = 201,
 	HPI_DESTNODE_LINEOUT = 202,	    /**< line out node. */
@@ -432,7 +429,18 @@ Property 2 - adapter can do stream grouping (supports SSX2)
 Property 1 - adapter can do samplerate conversion (MRX)
 Property 2 - adapter can do timestretch (TSX)
 */
-	HPI_ADAPTER_PROPERTY_CAPS2 = 269
+	HPI_ADAPTER_PROPERTY_CAPS2 = 269,
+
+/** Readonly adapter sync header connection count.
+*/
+	HPI_ADAPTER_PROPERTY_SYNC_HEADER_CONNECTIONS = 270,
+/** Readonly supports SSX2 property.
+Indicates the adapter supports SSX2 in some mode setting. The
+return value is true (1) or false (0). If the current adapter
+mode is MONO SSX2 is disabled, even though this property will
+return true.
+*/
+	HPI_ADAPTER_PROPERTY_SUPPORTS_SSX2 = 271
 };
 
 /** Adapter mode commands
@@ -813,8 +821,6 @@ enum HPI_SAMPLECLOCK_SOURCES {
 /** The sampleclock output is derived from its local samplerate generator.
     The local samplerate may be set using HPI_SampleClock_SetLocalRate(). */
 	HPI_SAMPLECLOCK_SOURCE_LOCAL = 1,
-/** \deprecated Use HPI_SAMPLECLOCK_SOURCE_LOCAL instead */
-	HPI_SAMPLECLOCK_SOURCE_ADAPTER = 1,
 /** The adapter is clocked from a dedicated AES/EBU SampleClock input.*/
 	HPI_SAMPLECLOCK_SOURCE_AESEBU_SYNC = 2,
 /** From external wordclock connector */
@@ -825,10 +831,6 @@ enum HPI_SAMPLECLOCK_SOURCES {
 	HPI_SAMPLECLOCK_SOURCE_SMPTE = 5,
 /** One of the aesebu inputs */
 	HPI_SAMPLECLOCK_SOURCE_AESEBU_INPUT = 6,
-/** \deprecated The first aesebu input with a valid signal
-Superseded by separate Auto enable flag
-*/
-	HPI_SAMPLECLOCK_SOURCE_AESEBU_AUTO = 7,
 /** From a network interface e.g. Cobranet or Livewire at either 48 or 96kHz */
 	HPI_SAMPLECLOCK_SOURCE_NETWORK = 8,
 /** From previous adjacent module (ASI2416 only)*/
@@ -1015,8 +1017,6 @@ enum HPI_ERROR_CODES {
 	HPI_ERROR_CONTROL_DISABLED = 404,
 	/** I2C transaction failed due to a missing ACK. */
 	HPI_ERROR_CONTROL_I2C_MISSING_ACK = 405,
-	/** Control attribute is valid, but not supported by this hardware. */
-	HPI_ERROR_UNSUPPORTED_CONTROL_ATTRIBUTE = 406,
 	/** Control is busy, or coming out of
 	reset and cannot be accessed at this time. */
 	HPI_ERROR_CONTROL_NOT_READY = 407,
@@ -1827,13 +1827,41 @@ u16 hpi_parametricEQ__get_coeffs(const struct hpi_hsubsys *ph_subsys,
   Compressor Expander control
 *******************************/
 
-u16 hpi_compander_set(const struct hpi_hsubsys *ph_subsys, u32 h_control,
-	u16 attack, u16 decay, short ratio100, short threshold0_01dB,
-	short makeup_gain0_01dB);
+u16 hpi_compander_set_enable(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 on);
+
+u16 hpi_compander_get_enable(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 *pon);
+
+u16 hpi_compander_set_makeup_gain(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, short makeup_gain0_01dB);
+
+u16 hpi_compander_get_makeup_gain(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, short *pn_makeup_gain0_01dB);
+
+u16 hpi_compander_set_attack_time_constant(const struct hpi_hsubsys
+	*ph_subsys, u32 h_control, u32 index, u32 attack);
+
+u16 hpi_compander_get_attack_time_constant(const struct hpi_hsubsys
+	*ph_subsys, u32 h_control, u32 index, u32 *pw_attack);
+
+u16 hpi_compander_set_decay_time_constant(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, u32 decay);
+
+u16 hpi_compander_get_decay_time_constant(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, u32 *pw_decay);
+
+u16 hpi_compander_set_threshold(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, short threshold0_01dB);
+
+u16 hpi_compander_get_threshold(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, short *pn_threshold0_01dB);
+
+u16 hpi_compander_set_ratio(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, u32 ratio100);
 
-u16 hpi_compander_get(const struct hpi_hsubsys *ph_subsys, u32 h_control,
-	u16 *pw_attack, u16 *pw_decay, short *pw_ratio100,
-	short *pn_threshold0_01dB, short *pn_makeup_gain0_01dB);
+u16 hpi_compander_get_ratio(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, u32 *pw_ratio100);
 
 /*******************************
   Cobranet HMI control
diff --git a/sound/pci/asihpi/hpidebug.h b/sound/pci/asihpi/hpidebug.h
index 44dccadcc25b..a2f0952a99f0 100644
--- a/sound/pci/asihpi/hpidebug.h
+++ b/sound/pci/asihpi/hpidebug.h
@@ -356,7 +356,7 @@ compile_time_assert((HPI_CONTROL_LAST_INDEX + 1 == 27),
 	"HPI_SOURCENODE_ADAPTER" \
 }
 
-compile_time_assert((HPI_SOURCENODE_LAST_INDEX - HPI_SOURCENODE_BASE + 1) ==
+compile_time_assert((HPI_SOURCENODE_LAST_INDEX - HPI_SOURCENODE_NONE + 1) ==
 	(12), sourcenode_strings_match_defs);
 
 #define HPI_DESTNODE_STRINGS \
@@ -370,7 +370,7 @@ compile_time_assert((HPI_SOURCENODE_LAST_INDEX - HPI_SOURCENODE_BASE + 1) ==
 	"HPI_DESTNODE_COBRANET", \
 	"HPI_DESTNODE_ANALOG" \
 }
-compile_time_assert((HPI_DESTNODE_LAST_INDEX - HPI_DESTNODE_BASE + 1) == (8),
+compile_time_assert((HPI_DESTNODE_LAST_INDEX - HPI_DESTNODE_NONE + 1) == (8),
 	destnode_strings_match_defs);
 
 #define HPI_CONTROL_CHANNEL_MODE_STRINGS \
-- 
cgit v1.2.3


From 1dd6aaaafc930dd9bfaa6ea1d21bac2b4ec12527 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Tue, 6 Jul 2010 08:37:07 +1200
Subject: ALSA: asihpi - Use version string instead of printf formatting

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpioctl.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 7396ac54e99f..311499992a22 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -464,9 +464,7 @@ void __init asihpi_init(void)
 
 	memset(adapters, 0, sizeof(adapters));
 
-	printk(KERN_INFO "ASIHPI driver %d.%02d.%02d\n",
-		HPI_VER_MAJOR(HPI_VER), HPI_VER_MINOR(HPI_VER),
-		HPI_VER_RELEASE(HPI_VER));
+	printk(KERN_INFO "ASIHPI driver " HPI_VER_STRING "\n");
 
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
 		HPI_SUBSYS_DRIVER_LOAD);
-- 
cgit v1.2.3


From 38439146355de2c10c369f93136333be6107a16b Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Tue, 6 Jul 2010 08:37:08 +1200
Subject: ALSA: asihpi - Add ASI5200 family

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpi6000.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index 12dab5e4892c..f7e374ec4414 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -687,6 +687,7 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 	switch (pao->pci.subsys_device_id) {
 	case 0x5100:
 	case 0x5110:	/* ASI5100 revB or higher with C6711D */
+	case 0x5200:	/* ASI5200 PC_ie version of ASI5100 */
 	case 0x6100:
 	case 0x6200:
 		boot_load_family = HPI_ADAPTER_FAMILY_ASI(0x6200);
@@ -1133,6 +1134,12 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 						subsys_device_id) ==
 					HPI_ADAPTER_FAMILY_ASI(0x5100))
 					mask = 0x00000000L;
+				/* ASI5200 uses AX6 code, */
+				/* but has no PLD r/w register to test */
+				if (HPI_ADAPTER_FAMILY_ASI(pao->pci.
+						subsys_device_id) ==
+					HPI_ADAPTER_FAMILY_ASI(0x5200))
+					mask = 0x00000000L;
 				break;
 			case HPI_ADAPTER_FAMILY_ASI(0x8800):
 				/* ASI8800 has 16bit path to FPGA */
-- 
cgit v1.2.3


From 108ccb3f0fa617a003c6b076b73b74d4f85e4cde Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Tue, 6 Jul 2010 08:37:09 +1200
Subject: ALSA: asihpi - Change compander API and tidy

Compander API changed to one function per parameter.
Factor out some common code for stereo log value reading.
Make some more entity functions static.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpi_internal.h |  30 ++--
 sound/pci/asihpi/hpifunc.c      | 324 ++++++++++++++++++++++++----------------
 2 files changed, 211 insertions(+), 143 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index fdd0ce02aa68..7ae7a1d59853 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -142,12 +142,15 @@ enum HPI_BUSES {
 /******************************************* CONTROL ATTRIBUTES ****/
 /* (in order of control type ID */
 
-	/* This allows for 255 control types, 256 unique attributes each */
+/* This allows for 255 control types, 256 unique attributes each */
 #define HPI_CTL_ATTR(ctl, ai) (HPI_CONTROL_##ctl * 0x100 + ai)
 
 /* Get the sub-index of the attribute for a control type */
 #define HPI_CTL_ATTR_INDEX(i) (i&0xff)
 
+/* Extract the control from the control attribute */
+#define HPI_CTL_ATTR_CONTROL(i) (i>>8)
+
 /* Generic control attributes.  */
 
 /** Enable a control.
@@ -311,8 +314,7 @@ Used for HPI_ChannelModeSet/Get()
 /* Microphone control attributes */
 #define HPI_MICROPHONE_PHANTOM_POWER HPI_CTL_ATTR(MICROPHONE, 1)
 
-/** Equalizer control attributes
-*/
+/** Equalizer control attributes */
 /** Used to get number of filters in an EQ. (Can't set) */
 #define HPI_EQUALIZER_NUM_FILTERS HPI_CTL_ATTR(EQUALIZER, 1)
 /** Set/get the filter by type, freq, Q, gain */
@@ -320,13 +322,15 @@ Used for HPI_ChannelModeSet/Get()
 /** Get the biquad coefficients */
 #define HPI_EQUALIZER_COEFFICIENTS HPI_CTL_ATTR(EQUALIZER, 3)
 
-#define HPI_COMPANDER_PARAMS HPI_CTL_ATTR(COMPANDER, 1)
+/* Note compander also uses HPI_GENERIC_ENABLE */
+#define HPI_COMPANDER_PARAMS     HPI_CTL_ATTR(COMPANDER, 1)
+#define HPI_COMPANDER_MAKEUPGAIN HPI_CTL_ATTR(COMPANDER, 2)
+#define HPI_COMPANDER_THRESHOLD  HPI_CTL_ATTR(COMPANDER, 3)
+#define HPI_COMPANDER_RATIO      HPI_CTL_ATTR(COMPANDER, 4)
+#define HPI_COMPANDER_ATTACK     HPI_CTL_ATTR(COMPANDER, 5)
+#define HPI_COMPANDER_DECAY      HPI_CTL_ATTR(COMPANDER, 6)
 
-/* Cobranet control attributes.
-   MUST be distinct from all other control attributes.
-   This is so that host side processing can easily identify a Cobranet control
-   and apply additional host side operations (like copying data) as required.
-*/
+/* Cobranet control attributes. */
 #define HPI_COBRANET_SET         HPI_CTL_ATTR(COBRANET, 1)
 #define HPI_COBRANET_GET         HPI_CTL_ATTR(COBRANET, 2)
 #define HPI_COBRANET_SET_DATA    HPI_CTL_ATTR(COBRANET, 3)
@@ -1512,11 +1516,11 @@ struct hpi_control_cache_single {
 	struct hpi_control_cache_info i;
 	union {
 		struct {	/* volume */
-			u16 an_log[2];
+			short an_log[2];
 		} v;
 		struct {	/* peak meter */
-			u16 an_log_peak[2];
-			u16 an_logRMS[2];
+			short an_log_peak[2];
+			short an_logRMS[2];
 		} p;
 		struct {	/* channel mode */
 			u16 mode;
@@ -1526,7 +1530,7 @@ struct hpi_control_cache_single {
 			u16 source_node_index;
 		} x;
 		struct {	/* level/trim */
-			u16 an_log[2];
+			short an_log[2];
 		} l;
 		struct {	/* tuner - partial caching.
 				   some attributes go to the DSP. */
diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c
index 298eef3e20e9..9c6958ab9284 100644
--- a/sound/pci/asihpi/hpifunc.c
+++ b/sound/pci/asihpi/hpifunc.c
@@ -96,8 +96,7 @@ void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR)
 
 static struct hpi_hsubsys gh_subsys;
 
-struct hpi_hsubsys *hpi_subsys_create(void
-	)
+struct hpi_hsubsys *hpi_subsys_create(void)
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
@@ -302,6 +301,7 @@ u16 hpi_adapter_set_mode_ex(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
 		HPI_ADAPTER_SET_MODE);
 	hm.adapter_index = adapter_index;
@@ -510,7 +510,7 @@ u16 hpi_adapter_debug_read(const struct hpi_hsubsys *ph_subsys,
 	hm.adapter_index = adapter_index;
 	hm.u.ax.debug_read.dsp_address = dsp_address;
 
-	if (*count_bytes > sizeof(hr.u.bytes))
+	if (*count_bytes > (int)sizeof(hr.u.bytes))
 		*count_bytes = sizeof(hr.u.bytes);
 
 	hm.u.ax.debug_read.count_bytes = *count_bytes;
@@ -976,6 +976,7 @@ u16 hpi_outstream_ancillary_read(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM,
 		HPI_OSTREAM_ANC_READ);
 	u32TOINDEXES(h_outstream, &hm.adapter_index, &hm.obj_index);
@@ -1581,6 +1582,7 @@ u16 hpi_control_param_set(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_SET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -1591,6 +1593,22 @@ u16 hpi_control_param_set(const struct hpi_hsubsys *ph_subsys,
 	return hr.error;
 }
 
+static u16 hpi_control_log_set2(u32 h_control, u16 attrib, short sv0,
+	short sv1)
+{
+	struct hpi_message hm;
+	struct hpi_response hr;
+
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
+		HPI_CONTROL_SET_STATE);
+	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
+	hm.u.c.attribute = attrib;
+	hm.u.c.an_log_value[0] = sv0;
+	hm.u.c.an_log_value[1] = sv1;
+	hpi_send_recv(&hm, &hr);
+	return hr.error;
+}
+
 static
 u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys,
 	const u32 h_control, const u16 attrib, u32 param1, u32 param2,
@@ -1598,6 +1616,7 @@ u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -1605,8 +1624,8 @@ u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys,
 	hm.u.c.param1 = param1;
 	hm.u.c.param2 = param2;
 	hpi_send_recv(&hm, &hr);
-	if (pparam1)
-		*pparam1 = hr.u.c.param1;
+
+	*pparam1 = hr.u.c.param1;
 	if (pparam2)
 		*pparam2 = hr.u.c.param2;
 
@@ -1617,10 +1636,23 @@ u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys,
 		hpi_control_param_get(s, h, a, 0, 0, p1, NULL)
 #define hpi_control_param2_get(s, h, a, p1, p2) \
 		hpi_control_param_get(s, h, a, 0, 0, p1, p2)
-#define hpi_control_ex_param1_get(s, h, a, p1) \
-		hpi_control_ex_param_get(s, h, a, 0, 0, p1, NULL)
-#define hpi_control_ex_param2_get(s, h, a, p1, p2) \
-		hpi_control_ex_param_get(s, h, a, 0, 0, p1, p2)
+
+static u16 hpi_control_log_get2(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u16 attrib, short *sv0, short *sv1)
+{
+	struct hpi_message hm;
+	struct hpi_response hr;
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
+		HPI_CONTROL_GET_STATE);
+	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
+	hm.u.c.attribute = attrib;
+
+	hpi_send_recv(&hm, &hr);
+	*sv0 = hr.u.c.an_log_value[0];
+	if (sv1)
+		*sv1 = hr.u.c.an_log_value[1];
+	return hr.error;
+}
 
 static
 u16 hpi_control_query(const struct hpi_hsubsys *ph_subsys,
@@ -1629,6 +1661,7 @@ u16 hpi_control_query(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_INFO);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -1643,9 +1676,8 @@ u16 hpi_control_query(const struct hpi_hsubsys *ph_subsys,
 	return hr.error;
 }
 
-static u16 hpi_control_get_string(const struct hpi_hsubsys *ph_subsys,
-	const u32 h_control, const u16 attribute, char *psz_string,
-	const u32 string_length)
+static u16 hpi_control_get_string(const u32 h_control, const u16 attribute,
+	char *psz_string, const u32 string_length)
 {
 	unsigned int sub_string_index = 0, j = 0;
 	char c = 0;
@@ -1916,6 +1948,7 @@ u16 hpi_cobranet_hmi_write(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
 		HPI_CONTROL_SET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -1941,6 +1974,7 @@ u16 hpi_cobranet_hmi_read(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -1980,6 +2014,7 @@ u16 hpi_cobranet_hmi_get_status(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2006,6 +2041,7 @@ u16 hpi_cobranet_getI_paddress(const struct hpi_hsubsys *ph_subsys,
 	u32 byte_count;
 	u32 iP;
 	u16 error;
+
 	error = hpi_cobranet_hmi_read(ph_subsys, h_control,
 		HPI_COBRANET_HMI_cobra_ip_mon_currentIP, 4, &byte_count,
 		(u8 *)&iP);
@@ -2082,6 +2118,7 @@ u16 hpi_cobranet_getMA_caddress(const struct hpi_hsubsys *ph_subsys,
 	u32 byte_count;
 	u16 error;
 	u32 mAC;
+
 	error = hpi_cobranet_hmi_read(ph_subsys, h_control,
 		HPI_COBRANET_HMI_cobra_if_phy_address, 4, &byte_count,
 		(u8 *)&mAC);
@@ -2103,53 +2140,111 @@ u16 hpi_cobranet_getMA_caddress(const struct hpi_hsubsys *ph_subsys,
 	return error;
 }
 
-u16 hpi_compander_set(const struct hpi_hsubsys *ph_subsys, u32 h_control,
-	u16 attack, u16 decay, short ratio100, short threshold0_01dB,
-	short makeup_gain0_01dB)
+u16 hpi_compander_set_enable(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 enable)
+{
+	return hpi_control_param_set(ph_subsys, h_control, HPI_GENERIC_ENABLE,
+		enable, 0);
+}
+
+u16 hpi_compander_get_enable(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 *enable)
+{
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_GENERIC_ENABLE, enable);
+}
+
+u16 hpi_compander_set_makeup_gain(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, short makeup_gain0_01dB)
+{
+	return hpi_control_log_set2(h_control, HPI_COMPANDER_MAKEUPGAIN,
+		makeup_gain0_01dB, 0);
+}
+
+u16 hpi_compander_get_makeup_gain(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, short *makeup_gain0_01dB)
+{
+	return hpi_control_log_get2(ph_subsys, h_control,
+		HPI_COMPANDER_MAKEUPGAIN, makeup_gain0_01dB, NULL);
+}
+
+u16 hpi_compander_set_attack_time_constant(const struct hpi_hsubsys
+	*ph_subsys, u32 h_control, unsigned int index, u32 attack)
+{
+	return hpi_control_param_set(ph_subsys, h_control,
+		HPI_COMPANDER_ATTACK, attack, index);
+}
+
+u16 hpi_compander_get_attack_time_constant(const struct hpi_hsubsys
+	*ph_subsys, u32 h_control, unsigned int index, u32 *attack)
+{
+	return hpi_control_param_get(ph_subsys, h_control,
+		HPI_COMPANDER_ATTACK, 0, index, attack, &index);
+}
+
+u16 hpi_compander_set_decay_time_constant(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, unsigned int index, u32 decay)
+{
+	return hpi_control_param_set(ph_subsys, h_control,
+		HPI_COMPANDER_DECAY, decay, index);
+}
+
+u16 hpi_compander_get_decay_time_constant(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, unsigned int index, u32 *decay)
+{
+	return hpi_control_param_get(ph_subsys, h_control,
+		HPI_COMPANDER_DECAY, 0, index, decay, &index);
+
+}
+
+u16 hpi_compander_set_threshold(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, unsigned int index, short threshold0_01dB)
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_SET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
-
-	hm.u.c.param1 = attack + ((u32)ratio100 << 16);
-	hm.u.c.param2 = (decay & 0xFFFFL);
+	hm.u.c.attribute = HPI_COMPANDER_THRESHOLD;
+	hm.u.c.param2 = index;
 	hm.u.c.an_log_value[0] = threshold0_01dB;
-	hm.u.c.an_log_value[1] = makeup_gain0_01dB;
-	hm.u.c.attribute = HPI_COMPANDER_PARAMS;
 
 	hpi_send_recv(&hm, &hr);
 
 	return hr.error;
 }
 
-u16 hpi_compander_get(const struct hpi_hsubsys *ph_subsys, u32 h_control,
-	u16 *pw_attack, u16 *pw_decay, short *pw_ratio100,
-	short *pn_threshold0_01dB, short *pn_makeup_gain0_01dB)
+u16 hpi_compander_get_threshold(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, unsigned int index, short *threshold0_01dB)
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
-	hm.u.c.attribute = HPI_COMPANDER_PARAMS;
+	hm.u.c.attribute = HPI_COMPANDER_THRESHOLD;
+	hm.u.c.param2 = index;
 
 	hpi_send_recv(&hm, &hr);
+	*threshold0_01dB = hr.u.c.an_log_value[0];
 
-	if (pw_attack)
-		*pw_attack = (short)(hr.u.c.param1 & 0xFFFF);
-	if (pw_decay)
-		*pw_decay = (short)(hr.u.c.param2 & 0xFFFF);
-	if (pw_ratio100)
-		*pw_ratio100 = (short)(hr.u.c.param1 >> 16);
+	return hr.error;
+}
 
-	if (pn_threshold0_01dB)
-		*pn_threshold0_01dB = hr.u.c.an_log_value[0];
-	if (pn_makeup_gain0_01dB)
-		*pn_makeup_gain0_01dB = hr.u.c.an_log_value[1];
+u16 hpi_compander_set_ratio(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, u32 ratio100)
+{
+	return hpi_control_param_set(ph_subsys, h_control,
+		HPI_COMPANDER_RATIO, ratio100, index);
+}
 
-	return hr.error;
+u16 hpi_compander_get_ratio(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 index, u32 *ratio100)
+{
+	return hpi_control_param_get(ph_subsys, h_control,
+		HPI_COMPANDER_RATIO, 0, index, ratio100, &index);
 }
 
 u16 hpi_level_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control,
@@ -2157,6 +2252,7 @@ u16 hpi_level_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2181,37 +2277,16 @@ u16 hpi_level_set_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	short an_gain0_01dB[HPI_MAX_CHANNELS]
 	)
 {
-	struct hpi_message hm;
-	struct hpi_response hr;
-
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
-		HPI_CONTROL_SET_STATE);
-	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
-	memcpy(hm.u.c.an_log_value, an_gain0_01dB,
-		sizeof(short) * HPI_MAX_CHANNELS);
-	hm.u.c.attribute = HPI_LEVEL_GAIN;
-
-	hpi_send_recv(&hm, &hr);
-
-	return hr.error;
+	return hpi_control_log_set2(h_control, HPI_LEVEL_GAIN,
+		an_gain0_01dB[0], an_gain0_01dB[1]);
 }
 
 u16 hpi_level_get_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	short an_gain0_01dB[HPI_MAX_CHANNELS]
 	)
 {
-	struct hpi_message hm;
-	struct hpi_response hr;
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
-		HPI_CONTROL_GET_STATE);
-	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
-	hm.u.c.attribute = HPI_LEVEL_GAIN;
-
-	hpi_send_recv(&hm, &hr);
-
-	memcpy(an_gain0_01dB, hr.u.c.an_log_value,
-		sizeof(short) * HPI_MAX_CHANNELS);
-	return hr.error;
+	return hpi_control_log_get2(ph_subsys, h_control, HPI_LEVEL_GAIN,
+		&an_gain0_01dB[0], &an_gain0_01dB[1]);
 }
 
 u16 hpi_meter_query_channels(const struct hpi_hsubsys *ph_subsys,
@@ -2413,6 +2488,7 @@ u16 hpi_parametricEQ__get_band(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2439,6 +2515,7 @@ u16 hpi_parametricEQ__set_band(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_SET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2460,6 +2537,7 @@ u16 hpi_parametricEQ__get_coeffs(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2623,8 +2701,8 @@ u16 hpi_tone_detector_get_frequency(const struct hpi_hsubsys *ph_subsys,
 u16 hpi_tone_detector_get_state(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *state)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_TONEDETECTOR_STATE, 0, 0, (u32 *)state, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_TONEDETECTOR_STATE, state);
 }
 
 u16 hpi_tone_detector_set_enable(const struct hpi_hsubsys *ph_subsys,
@@ -2637,8 +2715,8 @@ u16 hpi_tone_detector_set_enable(const struct hpi_hsubsys *ph_subsys,
 u16 hpi_tone_detector_get_enable(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *enable)
 {
-	return hpi_control_param_get(ph_subsys, h_control, HPI_GENERIC_ENABLE,
-		0, 0, (u32 *)enable, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_GENERIC_ENABLE, enable);
 }
 
 u16 hpi_tone_detector_set_event_enable(const struct hpi_hsubsys *ph_subsys,
@@ -2651,8 +2729,8 @@ u16 hpi_tone_detector_set_event_enable(const struct hpi_hsubsys *ph_subsys,
 u16 hpi_tone_detector_get_event_enable(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *event_enable)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_GENERIC_EVENT_ENABLE, 0, 0, (u32 *)event_enable, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_GENERIC_EVENT_ENABLE, event_enable);
 }
 
 u16 hpi_tone_detector_set_threshold(const struct hpi_hsubsys *ph_subsys,
@@ -2665,15 +2743,15 @@ u16 hpi_tone_detector_set_threshold(const struct hpi_hsubsys *ph_subsys,
 u16 hpi_tone_detector_get_threshold(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, int *threshold)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_TONEDETECTOR_THRESHOLD, 0, 0, (u32 *)threshold, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_TONEDETECTOR_THRESHOLD, (u32 *)threshold);
 }
 
 u16 hpi_silence_detector_get_state(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *state)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_SILENCEDETECTOR_STATE, 0, 0, (u32 *)state, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_SILENCEDETECTOR_STATE, state);
 }
 
 u16 hpi_silence_detector_set_enable(const struct hpi_hsubsys *ph_subsys,
@@ -2686,50 +2764,50 @@ u16 hpi_silence_detector_set_enable(const struct hpi_hsubsys *ph_subsys,
 u16 hpi_silence_detector_get_enable(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *enable)
 {
-	return hpi_control_param_get(ph_subsys, h_control, HPI_GENERIC_ENABLE,
-		0, 0, (u32 *)enable, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_GENERIC_ENABLE, enable);
 }
 
 u16 hpi_silence_detector_set_event_enable(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 event_enable)
 {
 	return hpi_control_param_set(ph_subsys, h_control,
-		HPI_GENERIC_EVENT_ENABLE, (u32)event_enable, 0);
+		HPI_GENERIC_EVENT_ENABLE, event_enable, 0);
 }
 
 u16 hpi_silence_detector_get_event_enable(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *event_enable)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_GENERIC_EVENT_ENABLE, 0, 0, (u32 *)event_enable, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_GENERIC_EVENT_ENABLE, event_enable);
 }
 
 u16 hpi_silence_detector_set_delay(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 delay)
 {
 	return hpi_control_param_set(ph_subsys, h_control,
-		HPI_SILENCEDETECTOR_DELAY, (u32)delay, 0);
+		HPI_SILENCEDETECTOR_DELAY, delay, 0);
 }
 
 u16 hpi_silence_detector_get_delay(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *delay)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_SILENCEDETECTOR_DELAY, 0, 0, (u32 *)delay, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_SILENCEDETECTOR_DELAY, delay);
 }
 
 u16 hpi_silence_detector_set_threshold(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, int threshold)
 {
 	return hpi_control_param_set(ph_subsys, h_control,
-		HPI_SILENCEDETECTOR_THRESHOLD, (u32)threshold, 0);
+		HPI_SILENCEDETECTOR_THRESHOLD, threshold, 0);
 }
 
 u16 hpi_silence_detector_get_threshold(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, int *threshold)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_SILENCEDETECTOR_THRESHOLD, 0, 0, (u32 *)threshold, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_SILENCEDETECTOR_THRESHOLD, (u32 *)threshold);
 }
 
 u16 hpi_tuner_query_band(const struct hpi_hsubsys *ph_subsys,
@@ -2822,6 +2900,7 @@ u16 hpi_tuner_getRF_level(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2838,6 +2917,7 @@ u16 hpi_tuner_get_rawRF_level(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2894,14 +2974,14 @@ u16 hpi_tuner_get_program(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 u16 hpi_tuner_get_hd_radio_dsp_version(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, char *psz_dsp_version, const u32 string_size)
 {
-	return hpi_control_get_string(ph_subsys, h_control,
+	return hpi_control_get_string(h_control,
 		HPI_TUNER_HDRADIO_DSP_VERSION, psz_dsp_version, string_size);
 }
 
 u16 hpi_tuner_get_hd_radio_sdk_version(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, char *psz_sdk_version, const u32 string_size)
 {
-	return hpi_control_get_string(ph_subsys, h_control,
+	return hpi_control_get_string(h_control,
 		HPI_TUNER_HDRADIO_SDK_VERSION, psz_sdk_version, string_size);
 }
 
@@ -2942,15 +3022,15 @@ u16 hpi_tuner_get_mode(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 u16 hpi_tuner_get_hd_radio_signal_quality(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *pquality)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_TUNER_HDRADIO_SIGNAL_QUALITY, 0, 0, pquality, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_TUNER_HDRADIO_SIGNAL_QUALITY, pquality);
 }
 
 u16 hpi_tuner_get_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *pblend)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_TUNER_HDRADIO_BLEND, 0, 0, pblend, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_TUNER_HDRADIO_BLEND, pblend);
 }
 
 u16 hpi_tuner_set_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys,
@@ -2965,6 +3045,7 @@ u16 hpi_tuner_getRDS(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -2981,43 +3062,43 @@ u16 hpi_tuner_getRDS(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 u16 HPI_PAD__get_channel_name(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, char *psz_string, const u32 data_length)
 {
-	return hpi_control_get_string(ph_subsys, h_control,
-		HPI_PAD_CHANNEL_NAME, psz_string, data_length);
+	return hpi_control_get_string(h_control, HPI_PAD_CHANNEL_NAME,
+		psz_string, data_length);
 }
 
 u16 HPI_PAD__get_artist(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	char *psz_string, const u32 data_length)
 {
-	return hpi_control_get_string(ph_subsys, h_control, HPI_PAD_ARTIST,
-		psz_string, data_length);
+	return hpi_control_get_string(h_control, HPI_PAD_ARTIST, psz_string,
+		data_length);
 }
 
 u16 HPI_PAD__get_title(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	char *psz_string, const u32 data_length)
 {
-	return hpi_control_get_string(ph_subsys, h_control, HPI_PAD_TITLE,
-		psz_string, data_length);
+	return hpi_control_get_string(h_control, HPI_PAD_TITLE, psz_string,
+		data_length);
 }
 
 u16 HPI_PAD__get_comment(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	char *psz_string, const u32 data_length)
 {
-	return hpi_control_get_string(ph_subsys, h_control, HPI_PAD_COMMENT,
-		psz_string, data_length);
+	return hpi_control_get_string(h_control, HPI_PAD_COMMENT, psz_string,
+		data_length);
 }
 
 u16 HPI_PAD__get_program_type(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *ppTY)
 {
-	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_PAD_PROGRAM_TYPE, 0, 0, ppTY, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_PAD_PROGRAM_TYPE, ppTY);
 }
 
 u16 HPI_PAD__get_rdsPI(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	u32 *ppI)
 {
-	return hpi_control_param_get(ph_subsys, h_control, HPI_PAD_PROGRAM_ID,
-		0, 0, ppI, NULL);
+	return hpi_control_param1_get(ph_subsys, h_control,
+		HPI_PAD_PROGRAM_ID, ppI);
 }
 
 u16 hpi_volume_query_channels(const struct hpi_hsubsys *ph_subsys,
@@ -3031,36 +3112,16 @@ u16 hpi_volume_set_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	short an_log_gain[HPI_MAX_CHANNELS]
 	)
 {
-	struct hpi_message hm;
-	struct hpi_response hr;
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
-		HPI_CONTROL_SET_STATE);
-	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
-	memcpy(hm.u.c.an_log_value, an_log_gain,
-		sizeof(short) * HPI_MAX_CHANNELS);
-	hm.u.c.attribute = HPI_VOLUME_GAIN;
-
-	hpi_send_recv(&hm, &hr);
-
-	return hr.error;
+	return hpi_control_log_set2(h_control, HPI_VOLUME_GAIN,
+		an_log_gain[0], an_log_gain[1]);
 }
 
 u16 hpi_volume_get_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	short an_log_gain[HPI_MAX_CHANNELS]
 	)
 {
-	struct hpi_message hm;
-	struct hpi_response hr;
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
-		HPI_CONTROL_GET_STATE);
-	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
-	hm.u.c.attribute = HPI_VOLUME_GAIN;
-
-	hpi_send_recv(&hm, &hr);
-
-	memcpy(an_log_gain, hr.u.c.an_log_value,
-		sizeof(short) * HPI_MAX_CHANNELS);
-	return hr.error;
+	return hpi_control_log_get2(ph_subsys, h_control, HPI_VOLUME_GAIN,
+		&an_log_gain[0], &an_log_gain[1]);
 }
 
 u16 hpi_volume_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control,
@@ -3068,6 +3129,7 @@ u16 hpi_volume_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -3094,6 +3156,7 @@ u16 hpi_volume_auto_fade_profile(const struct hpi_hsubsys *ph_subsys,
 {
 	struct hpi_message hm;
 	struct hpi_response hr;
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_SET_STATE);
 	u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index);
@@ -3170,43 +3233,43 @@ static size_t entity_type_to_size[LAST_ENTITY_TYPE] = {
 	6 * sizeof(char),
 };
 
-inline size_t hpi_entity_size(struct hpi_entity *entity_ptr)
+static inline size_t hpi_entity_size(struct hpi_entity *entity_ptr)
 {
 	return entity_ptr->header.size;
 }
 
-inline size_t hpi_entity_header_size(struct hpi_entity *entity_ptr)
+static inline size_t hpi_entity_header_size(struct hpi_entity *entity_ptr)
 {
 	return sizeof(entity_ptr->header);
 }
 
-inline size_t hpi_entity_value_size(struct hpi_entity *entity_ptr)
+static inline size_t hpi_entity_value_size(struct hpi_entity *entity_ptr)
 {
 	return hpi_entity_size(entity_ptr) -
 		hpi_entity_header_size(entity_ptr);
 }
 
-inline size_t hpi_entity_item_count(struct hpi_entity *entity_ptr)
+static inline size_t hpi_entity_item_count(struct hpi_entity *entity_ptr)
 {
 	return hpi_entity_value_size(entity_ptr) /
 		entity_type_to_size[entity_ptr->header.type];
 }
 
-inline struct hpi_entity *hpi_entity_ptr_to_next(struct hpi_entity
+static inline struct hpi_entity *hpi_entity_ptr_to_next(struct hpi_entity
 	*entity_ptr)
 {
 	return (void *)(((uint8_t *) entity_ptr) +
 		hpi_entity_size(entity_ptr));
 }
 
-inline u16 hpi_entity_check_type(const enum e_entity_type t)
+static inline u16 hpi_entity_check_type(const enum e_entity_type t)
 {
 	if (t >= 0 && t < STR_TYPE_FIELD_MAX)
 		return 0;
 	return HPI_ERROR_ENTITY_TYPE_INVALID;
 }
 
-inline u16 hpi_entity_check_role(const enum e_entity_role r)
+static inline u16 hpi_entity_check_role(const enum e_entity_role r)
 {
 	if (r >= 0 && r < STR_ROLE_FIELD_MAX)
 		return 0;
@@ -3624,6 +3687,7 @@ u16 hpi_async_event_wait(const struct hpi_hsubsys *ph_subsys, u32 h_async,
 	u16 maximum_events, struct hpi_async_event *p_events,
 	u16 *pw_number_returned)
 {
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From 36ed8bdd867314660b8dca2d1b6d9e92352b319b Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Tue, 6 Jul 2010 08:37:10 +1200
Subject: ALSA: asihpi - Minor HPI error handling fixes

Handle errors in tuner level caching,
Ccorrect error code for aesebu rx status.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpicmn.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index fcd64539d9ef..dda4f1c6f658 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -353,7 +353,12 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 			phr->u.c.param1 = pC->u.t.band;
 		else if ((phm->u.c.attribute == HPI_TUNER_LEVEL)
 			&& (phm->u.c.param1 == HPI_TUNER_LEVEL_AVERAGE))
-			phr->u.c.param1 = pC->u.t.level;
+			if (pC->u.t.level == HPI_ERROR_ILLEGAL_CACHE_VALUE) {
+				phr->u.c.param1 = 0;
+				phr->error =
+					HPI_ERROR_INVALID_CONTROL_ATTRIBUTE;
+			} else
+				phr->u.c.param1 = pC->u.t.level;
 		else
 			found = 0;
 		break;
@@ -397,7 +402,8 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 			if (pC->u.clk.source_index ==
 				HPI_ERROR_ILLEGAL_CACHE_VALUE) {
 				phr->u.c.param1 = 0;
-				phr->error = HPI_ERROR_INVALID_OPERATION;
+				phr->error =
+					HPI_ERROR_INVALID_CONTROL_ATTRIBUTE;
 			} else
 				phr->u.c.param1 = pC->u.clk.source_index;
 		} else if (phm->u.c.attribute == HPI_SAMPLECLOCK_SAMPLERATE)
-- 
cgit v1.2.3


From f978d36da4024ee22957f74276e944624a8c7f6d Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Tue, 6 Jul 2010 08:37:11 +1200
Subject: ALSA: asihpi - Remove unneeded ;

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpidebug.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpidebug.c b/sound/pci/asihpi/hpidebug.c
index 4cd85a401b34..949836ec913a 100644
--- a/sound/pci/asihpi/hpidebug.c
+++ b/sound/pci/asihpi/hpidebug.c
@@ -111,7 +111,7 @@ make_treenode_from_array(hpi_control_type_strings, HPI_CONTROL_TYPE_STRINGS)
   &hpi_profile_strings,\
   &hpi_control_strings, \
   &hpi_asyncevent_strings \
-};
+}
 	make_treenode_from_array(hpi_function_strings, HPI_FUNCTION_STRINGS)
 
 	compile_time_assert(HPI_OBJ_MAXINDEX == 14, obj_list_doesnt_match);
-- 
cgit v1.2.3


From 3507e2a8f171f4322bf78f9d618a4e435de843ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Thu, 8 Jul 2010 18:39:00 +0200
Subject: ALSA: hda - Add beep mixer support to Conexant codecs

Added the beep mixer controls to Conexant codecs.
They simply control the digital beep generator widget.

For cx5047, I couldn't find any beep generator, so it's not implemented
there.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 55 +++++++++++++++++++++++++++++++++++++++---
 1 file changed, 51 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 54f74191ebca..3b789ee548b4 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -131,6 +131,8 @@ struct conexant_spec {
 	unsigned int dc_enable;
 	unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */
 	unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
+
+	unsigned int beep_amp;
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -515,6 +517,15 @@ static struct snd_kcontrol_new cxt_capture_mixers[] = {
 	{}
 };
 
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+/* additional beep mixers; the actual parameters are overwritten at build */
+static struct snd_kcontrol_new cxt_beep_mixer[] = {
+	HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT),
+	{ } /* end */
+};
+#endif
+
 static const char *slave_vols[] = {
 	"Headphone Playback Volume",
 	"Speaker Playback Volume",
@@ -580,6 +591,23 @@ static int conexant_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+	/* create beep controls if needed */
+	if (spec->beep_amp) {
+		struct snd_kcontrol_new *knew;
+		for (knew = cxt_beep_mixer; knew->name; knew++) {
+			struct snd_kcontrol *kctl;
+			kctl = snd_ctl_new1(knew, codec);
+			if (!kctl)
+				return -ENOMEM;
+			kctl->private_value = spec->beep_amp;
+			err = snd_hda_ctl_add(codec, 0, kctl);
+			if (err < 0)
+				return err;
+		}
+	}
+#endif
+
 	return 0;
 }
 
@@ -590,6 +618,13 @@ static struct hda_codec_ops conexant_patch_ops = {
 	.free = conexant_free,
 };
 
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+#define set_beep_amp(spec, nid, idx, dir) \
+	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir))
+#else
+#define set_beep_amp(spec, nid, idx, dir) /* NOP */
+#endif
+
 /*
  * EAPD control
  * the private value = nid | (invert << 8)
@@ -1130,9 +1165,10 @@ static int patch_cxt5045(struct hda_codec *codec)
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5045_init_verbs;
 	spec->spdif_route = 0;
-	spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes),
-	spec->channel_mode = cxt5045_modes,
+	spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes);
+	spec->channel_mode = cxt5045_modes;
 
+	set_beep_amp(spec, 0x16, 0, 1);
 
 	codec->patch_ops = conexant_patch_ops;
 
@@ -1211,6 +1247,9 @@ static int patch_cxt5045(struct hda_codec *codec)
 		break;
 	}
 
+	if (spec->beep_amp)
+		snd_hda_attach_beep_device(codec, spec->beep_amp);
+
 	return 0;
 }
 
@@ -1987,6 +2026,8 @@ static int patch_cxt5051(struct hda_codec *codec)
 	spec->cur_adc = 0;
 	spec->cur_adc_idx = 0;
 
+	set_beep_amp(spec, 0x13, 0, HDA_OUTPUT);
+
 	codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
 
 	board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
@@ -2021,6 +2062,9 @@ static int patch_cxt5051(struct hda_codec *codec)
 		break;
 	}
 
+	if (spec->beep_amp)
+		snd_hda_attach_beep_device(codec, spec->beep_amp);
+
 	return 0;
 }
 
@@ -2636,7 +2680,6 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
 		.put = cxt5066_mic_boost_mux_enum_put,
 		.private_value = 0x23 | 0x100,
 	},
-	HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
 	{}
 };
 
@@ -3034,6 +3077,8 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->cur_adc = 0;
 	spec->cur_adc_idx = 0;
 
+	set_beep_amp(spec, 0x13, 0, HDA_OUTPUT);
+
 	board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
 						  cxt5066_models, cxt5066_cfg_tbl);
 	switch (board_config) {
@@ -3082,7 +3127,6 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->port_d_mode = 0;
 		spec->dell_vostro = 1;
 		spec->mic_boost = 3; /* default 30dB gain */
-		snd_hda_attach_beep_device(codec, 0x13);
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
@@ -3124,6 +3168,9 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	}
 
+	if (spec->beep_amp)
+		snd_hda_attach_beep_device(codec, spec->beep_amp);
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From afbd9b8448f4b7d15673c6858012f384f18d28b8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Thu, 8 Jul 2010 18:40:37 +0200
Subject: ALSA: hda - Limit the amp value to write

Check the amp max value at put callbacks and set the upper limit
so that the driver won't write any invalid value over the defined
range.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 32 +++++++++++++++++++++-----------
 1 file changed, 21 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a3d638c8c1fd..88a1c6acbcbd 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1539,6 +1539,17 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
 EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp);
 #endif /* SND_HDA_NEEDS_RESUME */
 
+static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir,
+			     unsigned int ofs)
+{
+	u32 caps = query_amp_caps(codec, nid, dir);
+	/* get num steps */
+	caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+	if (ofs < caps)
+		caps -= ofs;
+	return caps;
+}
+
 /**
  * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer
  *
@@ -1553,23 +1564,17 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 	u8 chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
 	unsigned int ofs = get_amp_offset(kcontrol);
-	u32 caps;
 
-	caps = query_amp_caps(codec, nid, dir);
-	/* num steps */
-	caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
-	if (!caps) {
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = chs == 3 ? 2 : 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = get_amp_max_value(codec, nid, dir, ofs);
+	if (!uinfo->value.integer.max) {
 		printk(KERN_WARNING "hda_codec: "
 		       "num_steps = 0 for NID=0x%x (ctl = %s)\n", nid,
 		       kcontrol->id.name);
 		return -EINVAL;
 	}
-	if (ofs < caps)
-		caps -= ofs;
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = chs == 3 ? 2 : 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = caps;
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info);
@@ -1594,8 +1599,13 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid,
 		 int ch, int dir, int idx, unsigned int ofs,
 		 unsigned int val)
 {
+	unsigned int maxval;
+
 	if (val > 0)
 		val += ofs;
+	maxval = get_amp_max_value(codec, nid, dir, ofs);
+	if (val > maxval)
+		val = maxval;
 	return snd_hda_codec_amp_update(codec, nid, ch, dir, idx,
 					HDA_AMP_VOLMASK, val);
 }
-- 
cgit v1.2.3


From 395c61d19621e80b763810cc988416dc1b6bfd3e Mon Sep 17 00:00:00 2001
From: Clemens Ladisch
Date: Mon, 12 Jul 2010 16:27:24 +0200
Subject: ALSA: via82xx: allow changing the initial DXS volume

As per-stream volume controls, the DXS controls are not intended to
adjust the overall sound level and so are initialized every time
a stream is opened.  However, there are special situations where one
wants to reduce the overall volume in the digital domain, i.e., before
the AC'97 codec's PCM volume control.  To allow this, add a module
parameter that sets the initial DXS volume.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/via82xx.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 7e494b6a1d0e..8c5f8b5a59f0 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -85,6 +85,7 @@ static int joystick;
 static int ac97_clock = 48000;
 static char *ac97_quirk;
 static int dxs_support;
+static int dxs_init_volume = 31;
 static int nodelay;
 
 module_param(index, int, 0444);
@@ -103,6 +104,8 @@ module_param(ac97_quirk, charp, 0444);
 MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
 module_param(dxs_support, int, 0444);
 MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)");
+module_param(dxs_init_volume, int, 0644);
+MODULE_PARM_DESC(dxs_init_volume, "initial DXS volume (0-31)");
 module_param(nodelay, int, 0444);
 MODULE_PARM_DESC(nodelay, "Disable 500ms init delay");
 
@@ -1245,8 +1248,10 @@ static int snd_via8233_playback_open(struct snd_pcm_substream *substream)
 		return err;
 	stream = viadev->reg_offset / 0x10;
 	if (chip->dxs_controls[stream]) {
-		chip->playback_volume[stream][0] = 0;
-		chip->playback_volume[stream][1] = 0;
+		chip->playback_volume[stream][0] =
+				VIA_DXS_MAX_VOLUME - (dxs_init_volume & 31);
+		chip->playback_volume[stream][1] =
+				VIA_DXS_MAX_VOLUME - (dxs_init_volume & 31);
 		chip->dxs_controls[stream]->vd[0].access &=
 			~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
 		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
-- 
cgit v1.2.3


From 32e0191d7909022e5016beb75dda6710a28b3c61 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch
Date: Mon, 12 Jul 2010 16:28:50 +0200
Subject: ALSA: HDA: VT1708S: fix Smart5.1 mode

Correctly configure bidirectional pins when resuming; do not power down
widgets when they are needed for Smart5.1 output; and on 3-jack boards,
create the streams and controls needed for six channels.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Viliam Kubis <viliam.kubis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 32 +++++++++++++++++++++++++-------
 1 file changed, 25 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 73453814e098..ae3acb2b42d1 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -552,24 +552,30 @@ static void via_auto_init_hp_out(struct hda_codec *codec)
 	}
 }
 
+static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin);
+
 static void via_auto_init_analog_input(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
+	unsigned int ctl;
 	int i;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
+		if (!nid)
+			continue;
 
+		if (spec->smart51_enabled && is_smart51_pins(spec, nid))
+			ctl = PIN_OUT;
+		else if (i <= AUTO_PIN_FRONT_MIC)
+			ctl = PIN_VREF50;
+		else
+			ctl = PIN_IN;
 		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    (i <= AUTO_PIN_FRONT_MIC ?
-				     PIN_VREF50 : PIN_IN));
-
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
 	}
 }
 
-static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin);
-
 static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int *affected_parm)
 {
@@ -658,6 +664,8 @@ static void set_jack_power_state(struct hda_codec *codec)
 		/* PW0 (19h), SW1 (18h), AOW1 (11h) */
 		parm = AC_PWRST_D3;
 		set_pin_power_state(codec, 0x19, &parm);
+		if (spec->smart51_enabled)
+			parm = AC_PWRST_D0;
 		snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE,
 				    parm);
 		snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
@@ -667,6 +675,8 @@ static void set_jack_power_state(struct hda_codec *codec)
 		if (is_8ch) {
 			parm = AC_PWRST_D3;
 			set_pin_power_state(codec, 0x22, &parm);
+			if (spec->smart51_enabled)
+				parm = AC_PWRST_D0;
 			snd_hda_codec_write(codec, 0x26, 0,
 					    AC_VERB_SET_POWER_STATE, parm);
 			snd_hda_codec_write(codec, 0x24, 0,
@@ -3915,6 +3925,13 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec,
 		}
 	}
 
+	/* for Smart 5.1, line/mic inputs double as output pins */
+	if (cfg->line_outs == 1) {
+		spec->multiout.num_dacs = 3;
+		spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11;
+		spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24;
+	}
+
 	return 0;
 }
 
@@ -3932,7 +3949,8 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec,
 	for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
 		nid = cfg->line_out_pins[i];
 
-		if (!nid)
+		/* for Smart 5.1, there are always at least six channels */
+		if (!nid && i > AUTO_SEQ_CENLFE)
 			continue;
 
 		nid_vol = nid_vols[i];
-- 
cgit v1.2.3


From 840b64c08032a86ab39b85ddd342918da0d559c8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 13 Jul 2010 22:49:01 +0200
Subject: ALSA: hda - Add support of dual-ADCs for Realtek ALC275

Some VAIO models with ALC275 have dual ADCs for both internal and external
mics, and the driver needs to switch one of them appropriately.
This patch adds a basic support for this functionality, dynamic switching
between two ADCs per jack plug state.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 178 ++++++++++++++++++++++++++++++++++++------
 1 file changed, 154 insertions(+), 24 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a7592f5e97d4..ca1a87a4812c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -333,6 +333,12 @@ struct alc_spec {
 	hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 
+	/* capture setup for dynamic dual-adc switch */
+	unsigned int cur_adc_idx;
+	hda_nid_t cur_adc;
+	unsigned int cur_adc_stream_tag;
+	unsigned int cur_adc_format;
+
 	/* capture source */
 	unsigned int num_mux_defs;
 	const struct hda_input_mux *input_mux;
@@ -374,6 +380,7 @@ struct alc_spec {
 
 	/* other flags */
 	unsigned int no_analog :1; /* digital I/O only */
+	unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */
 	int init_amp;
 
 	/* for virtual master */
@@ -1010,6 +1017,29 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
 	return -1;
 }
 
+/* switch the current ADC according to the jack state */
+static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int present;
+	hda_nid_t new_adc;
+
+	present = snd_hda_jack_detect(codec, spec->ext_mic.pin);
+	if (present)
+		spec->cur_adc_idx = 1;
+	else
+		spec->cur_adc_idx = 0;
+	new_adc = spec->adc_nids[spec->cur_adc_idx];
+	if (spec->cur_adc && spec->cur_adc != new_adc) {
+		/* stream is running, let's swap the current ADC */
+		snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+		spec->cur_adc = new_adc;
+		snd_hda_codec_setup_stream(codec, new_adc,
+					   spec->cur_adc_stream_tag, 0,
+					   spec->cur_adc_format);
+	}
+}
+
 static void alc_mic_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1024,6 +1054,11 @@ static void alc_mic_automute(struct hda_codec *codec)
 	if (snd_BUG_ON(!spec->adc_nids))
 		return;
 
+	if (spec->dual_adc_switch) {
+		alc_dual_mic_adc_auto_switch(codec);
+		return;
+	}
+
 	cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
 
 	present = snd_hda_jack_detect(codec, spec->ext_mic.pin);
@@ -3614,6 +3649,41 @@ static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
+/* analog capture with dynamic dual-adc changes */
+static int dualmic_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       unsigned int stream_tag,
+				       unsigned int format,
+				       struct snd_pcm_substream *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->cur_adc = spec->adc_nids[spec->cur_adc_idx];
+	spec->cur_adc_stream_tag = stream_tag;
+	spec->cur_adc_format = format;
+	snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+	return 0;
+}
+
+static int dualmic_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       struct snd_pcm_substream *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+	spec->cur_adc = 0;
+	return 0;
+}
+
+static struct hda_pcm_stream dualmic_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0, /* fill later */
+	.ops = {
+		.prepare = dualmic_capture_pcm_prepare,
+		.cleanup = dualmic_capture_pcm_cleanup
+	},
+};
 
 /*
  */
@@ -5052,24 +5122,12 @@ static void fixup_automic_adc(struct hda_codec *codec)
 	spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
-/* choose the ADC/MUX containing the input pin and initialize the setup */
-static void fixup_single_adc(struct hda_codec *codec)
+/* set the default connection to that pin */
+static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin = 0;
 	int i;
 
-	/* search for the input pin; there must be only one */
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (spec->autocfg.input_pins[i]) {
-			pin = spec->autocfg.input_pins[i];
-			break;
-		}
-	}
-	if (!pin)
-		return;
-
-	/* set the default connection to that pin */
 	for (i = 0; i < spec->num_adc_nids; i++) {
 		hda_nid_t cap = spec->capsrc_nids ?
 			spec->capsrc_nids[i] : spec->adc_nids[i];
@@ -5078,11 +5136,6 @@ static void fixup_single_adc(struct hda_codec *codec)
 		idx = get_connection_index(codec, cap, pin);
 		if (idx < 0)
 			continue;
-		/* use only this ADC */
-		if (spec->capsrc_nids)
-			spec->capsrc_nids += i;
-		spec->adc_nids += i;
-		spec->num_adc_nids = 1;
 		/* select or unmute this route */
 		if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
 			snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
@@ -5091,10 +5144,45 @@ static void fixup_single_adc(struct hda_codec *codec)
 			snd_hda_codec_write_cache(codec, cap, 0,
 					  AC_VERB_SET_CONNECT_SEL, idx);
 		}
+		return i; /* return the found index */
+	}
+	return -1; /* not found */
+}
+
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin = 0;
+	int i;
+
+	/* search for the input pin; there must be only one */
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (spec->autocfg.input_pins[i]) {
+			pin = spec->autocfg.input_pins[i];
+			break;
+		}
+	}
+	if (!pin)
 		return;
+	i = init_capsrc_for_pin(codec, pin);
+	if (i >= 0) {
+		/* use only this ADC */
+		if (spec->capsrc_nids)
+			spec->capsrc_nids += i;
+		spec->adc_nids += i;
+		spec->num_adc_nids = 1;
 	}
 }
 
+/* initialize dual adcs */
+static void fixup_dual_adc_switch(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	init_capsrc_for_pin(codec, spec->ext_mic.pin);
+	init_capsrc_for_pin(codec, spec->int_mic.pin);
+}
+
 static void set_capture_mixer(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -5108,7 +5196,10 @@ static void set_capture_mixer(struct hda_codec *codec)
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
 		int mux = 0;
-		if (spec->auto_mic)
+		int num_adcs = spec->num_adc_nids;
+		if (spec->dual_adc_switch)
+			fixup_dual_adc_switch(codec);
+		else if (spec->auto_mic)
 			fixup_automic_adc(codec);
 		else if (spec->input_mux) {
 			if (spec->input_mux->num_items > 1)
@@ -5116,7 +5207,9 @@ static void set_capture_mixer(struct hda_codec *codec)
 			else if (spec->input_mux->num_items == 1)
 				fixup_single_adc(codec);
 		}
-		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
+		if (spec->dual_adc_switch)
+			num_adcs = 1;
+		spec->cap_mixer = caps[mux][num_adcs - 1];
 	}
 }
 
@@ -14141,6 +14234,36 @@ static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid)
 }
 #endif /* CONFIG_SND_HDA_POWER_SAVE */
 
+static int alc275_setup_dual_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (codec->vendor_id != 0x10ec0275 || !spec->auto_mic)
+		return 0;
+	if ((spec->ext_mic.pin >= 0x18 && spec->int_mic.pin <= 0x13) ||
+	    (spec->ext_mic.pin <= 0x12 && spec->int_mic.pin >= 0x18)) {
+		if (spec->ext_mic.pin <= 0x12) {
+			spec->private_adc_nids[0] = 0x08;
+			spec->private_adc_nids[1] = 0x11;
+			spec->private_capsrc_nids[0] = 0x23;
+			spec->private_capsrc_nids[1] = 0x22;
+		} else {
+			spec->private_adc_nids[0] = 0x11;
+			spec->private_adc_nids[1] = 0x08;
+			spec->private_capsrc_nids[0] = 0x22;
+			spec->private_capsrc_nids[1] = 0x23;
+		}
+		spec->adc_nids = spec->private_adc_nids;
+		spec->capsrc_nids = spec->private_capsrc_nids;
+		spec->num_adc_nids = 2;
+		spec->dual_adc_switch = 1;
+		snd_printdd("realtek: enabling dual ADC switchg (%02x:%02x)\n",
+			    spec->adc_nids[0], spec->adc_nids[1]);
+		return 1;
+	}
+	return 0;
+}
+
 /*
  * BIOS auto configuration
  */
@@ -14180,11 +14303,14 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
-	fillup_priv_adc_nids(codec, alc269_adc_candidates,
-			     sizeof(alc269_adc_candidates));
+
+	if (!alc275_setup_dual_adc(codec))
+		fillup_priv_adc_nids(codec, alc269_adc_candidates,
+				     sizeof(alc269_adc_candidates));
 
 	/* set default input source */
-	snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
+	if (!spec->dual_adc_switch)
+		snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
 				  0, AC_VERB_SET_CONNECT_SEL,
 				  spec->input_mux->items[0].index);
 
@@ -14480,6 +14606,10 @@ static int patch_alc269(struct hda_codec *codec)
 		 */
 		spec->stream_analog_playback = &alc269_44k_pcm_analog_playback;
 		spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
+	} else if (spec->dual_adc_switch) {
+		spec->stream_analog_playback = &alc269_pcm_analog_playback;
+		/* switch ADC dynamically */
+		spec->stream_analog_capture = &dualmic_pcm_analog_capture;
 	} else {
 		spec->stream_analog_playback = &alc269_pcm_analog_playback;
 		spec->stream_analog_capture = &alc269_pcm_analog_capture;
-- 
cgit v1.2.3


From 315e8f7501ad929acacfa94c251283e837f281ed Mon Sep 17 00:00:00 2001
From: Kulikov Vasiliy
Date: Thu, 15 Jul 2010 22:48:19 +0400
Subject: ALSA: asihpi: fix sign bug

bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we
would not see it.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 91218f77217f..c80b0b863c54 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -460,6 +460,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_card_asihpi *card = snd_pcm_substream_chip(substream);
 	int err;
 	u16 format;
+	int width;
 	unsigned int bytes_per_sec;
 
 	print_hwparams(params);
@@ -512,9 +513,10 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 				dpcm->hpi_buffer_attached);
 	}
 	bytes_per_sec = params_rate(params) * params_channels(params);
-	bytes_per_sec *= snd_pcm_format_width(params_format(params));
+	width = snd_pcm_format_width(params_format(params));
+	bytes_per_sec *= width;
 	bytes_per_sec /= 8;
-	if (bytes_per_sec <= 0)
+	if (width < 0 || bytes_per_sec == 0)
 		return -EINVAL;
 
 	dpcm->bytes_per_sec = bytes_per_sec;
-- 
cgit v1.2.3


From 8d4bbee77e63981b91e4af7c569dc6a585ee0eb0 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Fri, 16 Jul 2010 17:50:59 +1200
Subject: ALSA: asihpi - HPI version 4.04.01

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpi.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index cee4df460f68..23399d02f666 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -50,8 +50,8 @@ i.e 3.05.02 is a development version
 #define HPI_VER_RELEASE(v) ((int)(v & 0xFF))
 
 /* Use single digits for versions less that 10 to avoid octal. */
-#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 36)
-#define HPI_VER_STRING "4.03.36"
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 4, 1)
+#define HPI_VER_STRING "4.04.01"
 
 /* Library version as documented in hpi-api-versions.txt */
 #define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(9, 0, 0)
-- 
cgit v1.2.3


From 604a440a9dd08d45570c555d78a17a4602c843d5 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Fri, 16 Jul 2010 17:51:00 +1200
Subject: ALSA: asihpi - Avoid using c99 uintX types.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpi_internal.h | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index 7ae7a1d59853..16f502d459de 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -104,9 +104,9 @@ typedef void hpi_handler_func(struct hpi_message *, struct hpi_response *);
 #define STR_ROLE_FIELD_MAX 255U
 
 struct hpi_entity_str {
-	uint16_t size;
-	uint8_t type;
-	uint8_t role;
+	u16 size;
+	u8 type;
+	u8 role;
 };
 
 #if defined(_MSC_VER)
@@ -119,11 +119,11 @@ struct hpi_entity {
 #if ! defined(HPI_OS_DSP_C6000) || (defined(HPI_OS_DSP_C6000) && (__TI_COMPILER_VERSION__ > 6000008))
 	/* DSP C6000 compiler v6.0.8 and lower
 	   do not support  flexible array member */
-	uint8_t value[];
+	u8 value[];
 #else
 	/* NOTE! Using sizeof(struct hpi_entity) will give erroneous results */
 #define HPI_INTERNAL_WARN_ABOUT_ENTITY_VALUE
-	uint8_t value[1];
+	u8 value[1];
 #endif
 };
 
-- 
cgit v1.2.3


From e2768c0c223d86a20ec392528bafd25996ce7585 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett
Date: Fri, 16 Jul 2010 17:51:01 +1200
Subject: ALSA: asihpi - Avoid useless assignment of returned index values.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpifunc.c | 9 ++++-----
 1 file changed, 4 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c
index 9c6958ab9284..1e92eb6dd509 100644
--- a/sound/pci/asihpi/hpifunc.c
+++ b/sound/pci/asihpi/hpifunc.c
@@ -2179,7 +2179,7 @@ u16 hpi_compander_get_attack_time_constant(const struct hpi_hsubsys
 	*ph_subsys, u32 h_control, unsigned int index, u32 *attack)
 {
 	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_COMPANDER_ATTACK, 0, index, attack, &index);
+		HPI_COMPANDER_ATTACK, 0, index, attack, NULL);
 }
 
 u16 hpi_compander_set_decay_time_constant(const struct hpi_hsubsys *ph_subsys,
@@ -2193,7 +2193,7 @@ u16 hpi_compander_get_decay_time_constant(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, unsigned int index, u32 *decay)
 {
 	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_COMPANDER_DECAY, 0, index, decay, &index);
+		HPI_COMPANDER_DECAY, 0, index, decay, NULL);
 
 }
 
@@ -2244,7 +2244,7 @@ u16 hpi_compander_get_ratio(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 index, u32 *ratio100)
 {
 	return hpi_control_param_get(ph_subsys, h_control,
-		HPI_COMPANDER_RATIO, 0, index, ratio100, &index);
+		HPI_COMPANDER_RATIO, 0, index, ratio100, NULL);
 }
 
 u16 hpi_level_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control,
@@ -3258,8 +3258,7 @@ static inline size_t hpi_entity_item_count(struct hpi_entity *entity_ptr)
 static inline struct hpi_entity *hpi_entity_ptr_to_next(struct hpi_entity
 	*entity_ptr)
 {
-	return (void *)(((uint8_t *) entity_ptr) +
-		hpi_entity_size(entity_ptr));
+	return (void *)(((u8 *)entity_ptr) + hpi_entity_size(entity_ptr));
 }
 
 static inline u16 hpi_entity_check_type(const enum e_entity_type t)
-- 
cgit v1.2.3


From 79c944ad136c4d14388d803b51113dcaaa1d179d Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela
Date: Mon, 19 Jul 2010 15:52:39 +0200
Subject: ALSA: hda-intel - do not mix audio and modem function IDs

The function IDs are different for audio and modem. Do not mix them.
Also, show the unsolicited bit in the function_id register.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/hda_codec.c | 8 +++++---
 sound/pci/hda/hda_codec.h | 5 ++++-
 sound/pci/hda/hda_proc.c  | 7 ++++++-
 3 files changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a3d638c8c1fd..6e0de65f1f3a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -730,15 +730,17 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)
 	total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);
 	for (i = 0; i < total_nodes; i++, nid++) {
 		function_id = snd_hda_param_read(codec, nid,
-						AC_PAR_FUNCTION_TYPE) & 0xff;
+						AC_PAR_FUNCTION_TYPE);
 		switch (function_id) {
 		case AC_GRP_AUDIO_FUNCTION:
 			codec->afg = nid;
-			codec->function_id = function_id;
+			codec->afg_function_id = function_id & 0xff;
+			codec->afg_unsol = (function_id >> 8) & 1;
 			break;
 		case AC_GRP_MODEM_FUNCTION:
 			codec->mfg = nid;
-			codec->function_id = function_id;
+			codec->mfg_function_id = function_id & 0xff;
+			codec->mfg_unsol = (function_id >> 8) & 1;
 			break;
 		default:
 			break;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 49e939e7e5cd..f96e909f549c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -760,7 +760,10 @@ struct hda_codec {
 	hda_nid_t mfg;	/* MFG node id */
 
 	/* ids */
-	u32 function_id;
+	u8 afg_function_id;
+	u8 mfg_function_id;
+	u8 afg_unsol;
+	u8 mfg_unsol;
 	u32 vendor_id;
 	u32 subsystem_id;
 	u32 revision_id;
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index f97d35de66c4..f025200f2a62 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -557,7 +557,12 @@ static void print_codec_info(struct snd_info_entry *entry,
 	else
 		snd_iprintf(buffer, "Not Set\n");
 	snd_iprintf(buffer, "Address: %d\n", codec->addr);
-	snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id);
+	if (codec->afg)
+		snd_iprintf(buffer, "AFG Function Id: 0x%x (unsol %u)\n",
+			codec->afg_function_id, codec->afg_unsol);
+	if (codec->mfg)
+		snd_iprintf(buffer, "MFG Function Id: 0x%x (unsol %u)\n",
+			codec->mfg_function_id, codec->mfg_unsol);
 	snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id);
 	snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id);
 	snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id);
-- 
cgit v1.2.3


From 0b6d092c8eeeb43893503afd2f6c1c67ceafc863 Mon Sep 17 00:00:00 2001
From: Kulikov Vasiliy
Date: Fri, 16 Jul 2010 20:15:43 +0400
Subject: ALSA: echoaudio: check kmalloc() result

If kmalloc() fails exit with -ENOMEM.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Ack-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 668a5ec04499..20763dd03fa0 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2250,6 +2250,8 @@ static int snd_echo_resume(struct pci_dev *pci)
 	DE_INIT(("resume start\n"));
 	pci_restore_state(pci);
 	commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL);
+	if (commpage_bak == NULL)
+		return -ENOMEM;
 	commpage = chip->comm_page;
 	memcpy(commpage_bak, commpage, sizeof(struct comm_page));
 
-- 
cgit v1.2.3


From 68bf57001f4995a25ca65f3711ff05b6ea25e8b6 Mon Sep 17 00:00:00 2001
From: Kulikov Vasiliy
Date: Fri, 16 Jul 2010 20:15:59 +0400
Subject: ALSA: riptide: check kzalloc() result

If kzalloc() fails exit with -ENOMEM.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/riptide/riptide.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 59d79962f236..f64fb7d988cb 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1615,7 +1615,10 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream)
 
 	chip->playback_substream[sub_num] = substream;
 	runtime->hw = snd_riptide_playback;
+
 	data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL);
+	if (data == NULL)
+		return -ENOMEM;
 	data->paths = lbus_play_paths[sub_num];
 	data->id = play_ids[sub_num];
 	data->source = play_sources[sub_num];
@@ -1635,7 +1638,10 @@ static int snd_riptide_capture_open(struct snd_pcm_substream *substream)
 
 	chip->capture_substream = substream;
 	runtime->hw = snd_riptide_capture;
+
 	data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL);
+	if (data == NULL)
+		return -ENOMEM;
 	data->paths = lbus_rec_path;
 	data->id = PADC;
 	data->source = ACLNK2PADC;
-- 
cgit v1.2.3


From ab85457f0a46b9dab17aaa01ceefc755b124d48d Mon Sep 17 00:00:00 2001
From: Jerone Young
Date: Mon, 19 Jul 2010 08:30:58 -0500
Subject: ALSA: hda - Add conexant quirk for AMD based Lenovo G series machines

This is a follow on patch adds support for AMD based Lenovo G series
machines, such as the Lenovo G555.

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3b789ee548b4..c99425ab2e70 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3040,8 +3040,10 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
+ 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
-	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
 	{}
 };
 
-- 
cgit v1.2.3


From cd7643bfb772dc7103ed6fc8dda6b233a8e14178 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela
Date: Tue, 20 Jul 2010 12:11:25 +0200
Subject: ALSA: hda-intel - fix function_id rework (add missing bitmask)

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 6e0de65f1f3a..3252945f3743 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -731,7 +731,7 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)
 	for (i = 0; i < total_nodes; i++, nid++) {
 		function_id = snd_hda_param_read(codec, nid,
 						AC_PAR_FUNCTION_TYPE);
-		switch (function_id) {
+		switch (function_id & 0xff) {
 		case AC_GRP_AUDIO_FUNCTION:
 			codec->afg = nid;
 			codec->afg_function_id = function_id & 0xff;
-- 
cgit v1.2.3


From 2385b789f1525542396d8f6b0cc37c1eb2493b4c Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Wed, 2 Jun 2010 16:56:41 +0200
Subject: ALSA: hda - Ensure codec patch files are checked for the correct
 codec ID

Signed-off-by: David Henningsson <diwic@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index a1fc83753cc6..bf3ced51e0f8 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -649,7 +649,9 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus,
 	*codecp = NULL;
 	if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) {
 		list_for_each_entry(codec, &bus->codec_list, list) {
-			if (codec->addr == caddr) {
+			if (codec->vendor_id == vendorid &&
+			    codec->subsystem_id == subid &&
+			    codec->addr == caddr) {
 				*codecp = codec;
 				break;
 			}
-- 
cgit v1.2.3


From 7ccc3eface57b6e1773fce009dac8a3da081b8b1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Mon, 26 Jul 2010 17:00:15 +0200
Subject: ALSA: hda - Fix max amp cap calculation for IDT/STAC codecs

The commit afbd9b8448f4b7d15673c6858012f384f18d28b8
    ALSA: hda - Limit the amp value to write
introduced a regression for codec setups with amp offsets like IDT/STAC
codecs.  The limit value should be a raw value without offset calculation.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 501cbc411a83..e5c3484388f0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1629,7 +1629,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid,
 
 	if (val > 0)
 		val += ofs;
-	maxval = get_amp_max_value(codec, nid, dir, ofs);
+	/* ofs = 0: raw max value */
+	maxval = get_amp_max_value(codec, nid, dir, 0);
 	if (val > maxval)
 		val = maxval;
 	return snd_hda_codec_amp_update(codec, nid, ch, dir, idx,
-- 
cgit v1.2.3


From 63818c448ac6f4dd75aa42997acaa746f86acb6b Mon Sep 17 00:00:00 2001
From: Axel Lin
Date: Wed, 28 Jul 2010 16:58:42 +0800
Subject: ALSA: hpimsgx: fix wrong sizeof

The correct size should be sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS),
sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS) is incorrect.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpimsgx.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index 2ee90dc3d897..f01ab964f602 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -741,7 +741,7 @@ static void HPIMSGX__reset(u16 adapter_index)
 		hpi_init_response(&hr, HPI_OBJ_SUBSYSTEM,
 			HPI_SUBSYS_FIND_ADAPTERS, 0);
 		memcpy(&gRESP_HPI_SUBSYS_FIND_ADAPTERS, &hr,
-			sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS));
+			sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS));
 
 		for (adapter = 0; adapter < HPI_MAX_ADAPTERS; adapter++) {
 
-- 
cgit v1.2.3


From ec9d04b2a8f00b14a3df4714820cb2cda46dc4d6 Mon Sep 17 00:00:00 2001
From: Kulikov Vasiliy
Date: Wed, 28 Jul 2010 20:41:56 +0400
Subject: ALSA: asihpi: check return value of get_user()

get_user() may fail, if so return -EFAULT.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpioctl.c | 17 +++++++++++++----
 1 file changed, 13 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 311499992a22..62895a719fcb 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -121,11 +121,17 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 	phpi_ioctl_data = (struct hpi_ioctl_linux __user *)arg;
 
 	/* Read the message and response pointers from user space.  */
-	get_user(puhm, &phpi_ioctl_data->phm);
-	get_user(puhr, &phpi_ioctl_data->phr);
+	if (get_user(puhm, &phpi_ioctl_data->phm) ||
+	    get_user(puhr, &phpi_ioctl_data->phr)) {
+		err = -EFAULT;
+		goto out;
+	}
 
 	/* Now read the message size and data from user space.  */
-	get_user(hm->h.size, (u16 __user *)puhm);
+	if (get_user(hm->h.size, (u16 __user *)puhm)) {
+		err = -EFAULT;
+		goto out;
+	}
 	if (hm->h.size > sizeof(*hm))
 		hm->h.size = sizeof(*hm);
 
@@ -138,7 +144,10 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 		goto out;
 	}
 
-	get_user(res_max_size, (u16 __user *)puhr);
+	if (get_user(res_max_size, (u16 __user *)puhr)) {
+		err = -EFAULT;
+		goto out;
+	}
 	/* printk(KERN_INFO "user response size %d\n", res_max_size); */
 	if (res_max_size < sizeof(struct hpi_response_header)) {
 		HPI_DEBUG_LOG(WARNING, "small res size %d\n", res_max_size);
-- 
cgit v1.2.3


From 150b432f448281d5518f5229d240923f9a9c5459 Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Thu, 29 Jul 2010 14:46:42 +0200
Subject: ALSA: hda - Rename iMic to Int Mic on Lenovo NB0763

The non-standard name "iMic" makes PulseAudio ignore the microphone.
BugLink: https://launchpad.net/bugs/605101

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 439d6e77040e..14ef38352088 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7103,7 +7103,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
-		{ "iMic", 0x1 },
+		{ "Int Mic", 0x1 },
 		{ "Line", 0x2 },
 		{ "CD", 0x4 },
 	},
@@ -8673,8 +8673,8 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
 	{ } /* end */
 };
 
-- 
cgit v1.2.3


From 5aacc2186cc075880a9eca42e6b7f9bb3096d0ea Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 10:36:29 +0200
Subject: ALSA: hda - Make error messages more verbose

Add a prefix and more information for error messages regarding the
connection-list in hda_codec.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 9 ++++++---
 1 file changed, 6 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e5c3484388f0..d9d1c91dfd1b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -396,15 +396,18 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 			}
 			for (n = prev_nid + 1; n <= val; n++) {
 				if (conns >= max_conns) {
-					snd_printk(KERN_ERR
-						   "Too many connections\n");
+					snd_printk(KERN_ERR "hda_codec: "
+						   "Too many connections %d for NID 0x%x\n",
+						   conns, nid);
 					return -EINVAL;
 				}
 				conn_list[conns++] = n;
 			}
 		} else {
 			if (conns >= max_conns) {
-				snd_printk(KERN_ERR "Too many connections\n");
+				snd_printk(KERN_ERR "hda_codec: "
+					   "Too many connections %d for NID 0x%x\n",
+					   conns, nid);
 				return -EINVAL;
 			}
 			conn_list[conns++] = val;
-- 
cgit v1.2.3


From ce503f38bdb59c9175a9076215a3ba579fad4e64 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 10:37:29 +0200
Subject: ALSA: hda - Increase the connection list size for ALC662

Some ALC662-compatible codecs like ALC892 may have more than 4
connections for the input source.  Use HDA_MAX_CONNECTIONS instead of
the fixed magic number 4.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 14ef38352088..cf9f20805173 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -18657,7 +18657,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t dac)
 {
 	int i, num;
-	hda_nid_t srcs[4];
+	hda_nid_t srcs[HDA_MAX_CONNECTIONS];
 
 	alc_set_pin_output(codec, nid, pin_type);
 	/* need the manual connection? */
-- 
cgit v1.2.3


From 757899aceebc33d9f86bbc481be7b7bf861e89ac Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 10:48:14 +0200
Subject: ALSA: hda - Share digital I/O parser in patch_realtek.c

Make a helper function to parse the digital I/Os of all Realtek codecs
to simplify the code and to ensure the setups.
Also, initialize digital I/O pins properly in init callbacks.  Some BIOS
seem to leave pins uninitialized.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 140 +++++++++++++++++++++++-------------------
 1 file changed, 78 insertions(+), 62 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cf9f20805173..442adef38c5b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1541,6 +1541,63 @@ static int alc_read_coef_idx(struct hda_codec *codec,
 	return val;
 }
 
+/* set right pin controls for digital I/O */
+static void alc_auto_init_digital(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+	hda_nid_t pin;
+
+	for (i = 0; i < spec->autocfg.dig_outs; i++) {
+		pin = spec->autocfg.dig_out_pins[i];
+		if (pin) {
+			snd_hda_codec_write(codec, pin, 0,
+					    AC_VERB_SET_PIN_WIDGET_CONTROL,
+					    PIN_OUT);
+		}
+	}
+	pin = spec->autocfg.dig_in_pin;
+	if (pin)
+		snd_hda_codec_write(codec, pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+				    PIN_IN);
+}
+
+/* parse digital I/Os and set up NIDs in BIOS auto-parse mode */
+static void alc_auto_parse_digital(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int i, err;
+	hda_nid_t dig_nid;
+
+	/* support multiple SPDIFs; the secondary is set up as a slave */
+	for (i = 0; i < spec->autocfg.dig_outs; i++) {
+		err = snd_hda_get_connections(codec,
+					      spec->autocfg.dig_out_pins[i],
+					      &dig_nid, 1);
+		if (err < 0)
+			continue;
+		if (!i) {
+			spec->multiout.dig_out_nid = dig_nid;
+			spec->dig_out_type = spec->autocfg.dig_out_type[0];
+		} else {
+			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
+				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
+		}
+	}
+
+	if (spec->autocfg.dig_in_pin) {
+		hda_nid_t dig_nid;
+		err = snd_hda_get_connections(codec,
+					      spec->autocfg.dig_in_pin,
+					      &dig_nid, 1);
+		if (err > 0)
+			spec->dig_in_nid = dig_nid;
+	}
+}
+
 /*
  * ALC888
  */
@@ -5013,7 +5070,7 @@ static void alc880_auto_init_input_src(struct hda_codec *codec)
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int i, err;
+	int err;
 	static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
@@ -5044,25 +5101,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	/* check multiple SPDIF-out (for recent codecs) */
-	for (i = 0; i < spec->autocfg.dig_outs; i++) {
-		hda_nid_t dig_nid;
-		err = snd_hda_get_connections(codec,
-					      spec->autocfg.dig_out_pins[i],
-					      &dig_nid, 1);
-		if (err < 0)
-			continue;
-		if (!i)
-			spec->multiout.dig_out_nid = dig_nid;
-		else {
-			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
-				break;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-		}
-	}
-	if (spec->autocfg.dig_in_pin)
-		spec->dig_in_nid = ALC880_DIGIN_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -5085,6 +5124,7 @@ static void alc880_auto_init(struct hda_codec *codec)
 	alc880_auto_init_extra_out(codec);
 	alc880_auto_init_analog_input(codec);
 	alc880_auto_init_input_src(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -6724,6 +6764,7 @@ static void alc260_auto_init(struct hda_codec *codec)
 	alc260_auto_init_multi_out(codec);
 	alc260_auto_init_analog_input(codec);
 	alc260_auto_init_input_src(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -10546,7 +10587,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	static hda_nid_t alc882_ignore[] = { 0x1d, 0 };
-	int i, err;
+	int err;
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc882_ignore);
@@ -10576,25 +10617,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	/* check multiple SPDIF-out (for recent codecs) */
-	for (i = 0; i < spec->autocfg.dig_outs; i++) {
-		hda_nid_t dig_nid;
-		err = snd_hda_get_connections(codec,
-					      spec->autocfg.dig_out_pins[i],
-					      &dig_nid, 1);
-		if (err < 0)
-			continue;
-		if (!i)
-			spec->multiout.dig_out_nid = dig_nid;
-		else {
-			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
-				break;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-		}
-	}
-	if (spec->autocfg.dig_in_pin)
-		spec->dig_in_nid = ALC880_DIGIN_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -10624,6 +10647,7 @@ static void alc882_auto_init(struct hda_codec *codec)
 	alc882_auto_init_hp_out(codec);
 	alc882_auto_init_analog_input(codec);
 	alc882_auto_init_input_src(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -12154,12 +12178,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
  dig_only:
-	if (spec->autocfg.dig_outs) {
-		spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
-		spec->dig_out_type = spec->autocfg.dig_out_type[0];
-	}
-	if (spec->autocfg.dig_in_pin)
-		spec->dig_in_nid = ALC262_DIGIN_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -12191,6 +12210,7 @@ static void alc262_auto_init(struct hda_codec *codec)
 	alc262_auto_init_hp_out(codec);
 	alc262_auto_init_analog_input(codec);
 	alc262_auto_init_input_src(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -13327,10 +13347,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 
  dig_only:
 	/* digital only support output */
-	if (spec->autocfg.dig_outs) {
-		spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
-		spec->dig_out_type = spec->autocfg.dig_out_type[0];
-	}
+	alc_auto_parse_digital(codec);
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
@@ -13360,6 +13377,7 @@ static void alc268_auto_init(struct hda_codec *codec)
 	alc268_auto_init_hp_out(codec);
 	alc268_auto_init_mono_speaker_out(codec);
 	alc268_auto_init_analog_input(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -14305,8 +14323,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_outs)
-		spec->multiout.dig_out_nid = ALC269_DIGOUT_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -14354,6 +14371,7 @@ static void alc269_auto_init(struct hda_codec *codec)
 	alc269_auto_init_multi_out(codec);
 	alc269_auto_init_hp_out(codec);
 	alc269_auto_init_analog_input(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -15515,8 +15533,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_outs)
-		spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -15542,6 +15559,7 @@ static void alc861_auto_init(struct hda_codec *codec)
 	alc861_auto_init_multi_out(codec);
 	alc861_auto_init_hp_out(codec);
 	alc861_auto_init_analog_input(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -16646,8 +16664,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_outs)
-		spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -16674,6 +16691,7 @@ static void alc861vd_auto_init(struct hda_codec *codec)
 	alc861vd_auto_init_hp_out(codec);
 	alc861vd_auto_init_analog_input(codec);
 	alc861vd_auto_init_input_src(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -18761,8 +18779,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_outs)
-		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+	alc_auto_parse_digital(codec);
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -18799,6 +18816,7 @@ static void alc662_auto_init(struct hda_codec *codec)
 	alc662_auto_init_hp_out(codec);
 	alc662_auto_init_analog_input(codec);
 	alc662_auto_init_input_src(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -19124,10 +19142,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
 
  dig_only:
 	/* digital only support output */
-	if (spec->autocfg.dig_outs) {
-		spec->multiout.dig_out_nid = ALC680_DIGOUT_NID;
-		spec->dig_out_type = spec->autocfg.dig_out_type[0];
-	}
+	alc_auto_parse_digital(codec);
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
@@ -19151,6 +19166,7 @@ static void alc680_auto_init(struct hda_codec *codec)
 	alc680_auto_init_multi_out(codec);
 	alc680_auto_init_hp_out(codec);
 	alc680_auto_init_analog_input(codec);
+	alc_auto_init_digital(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
-- 
cgit v1.2.3


From 5d4abf93ea3192cc666430225a29a4978c97c57d Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 10:51:10 +0200
Subject: ALSA: hda - Handle missing NID 0x1b on ALC259 codec

Since ALC259/269 use the same parser of ALC268, the pin 0x1b was ignored
as an invalid widget.  Just add this NID to handle properly.
This will add the missing mixer controls for some devices.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 442adef38c5b..bdea95aee448 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13144,6 +13144,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x1b:
 	case 0x21: /* ALC269vb has this pin, too */
 		dac = 0x03;
 		break;
-- 
cgit v1.2.3


From 954a29c881bd0c61352af0946f2c39d738d43c1b Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 10:55:44 +0200
Subject: ALSA: hda - Prefer VREF50 if BIOS sets for Realtek codecs

If BIOS sets up the input pin as VREF 50, use the value as is instead of
overriding forcibly to VREF 80.  This fixes the quality of inputs on
some devices like Packard-Bell M5210.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++--
 1 file changed, 16 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bdea95aee448..4d3a6f05c703 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -847,9 +847,13 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
 
 	if (auto_pin_type <= AUTO_PIN_FRONT_MIC) {
 		unsigned int pincap;
+		unsigned int oldval;
+		oldval = snd_hda_codec_read(codec, nid, 0,
+					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		pincap = snd_hda_query_pin_caps(codec, nid);
 		pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
-		if (pincap & AC_PINCAP_VREF_80)
+		/* if the default pin setup is vref50, we give it priority */
+		if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
 			val = PIN_VREF80;
 		else if (pincap & AC_PINCAP_VREF_50)
 			val = PIN_VREF50;
@@ -10406,7 +10410,8 @@ static struct alc_config_preset alc882_presets[] = {
  * Pin config fixes
  */
 enum {
-	PINFIX_ABIT_AW9D_MAX
+	PINFIX_ABIT_AW9D_MAX,
+	PINFIX_PB_M5210,
 };
 
 static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
@@ -10416,13 +10421,22 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
 	{ }
 };
 
+static const struct hda_verb pb_m5210_verbs[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
+	{}
+};
+
 static const struct alc_fixup alc882_fixups[] = {
 	[PINFIX_ABIT_AW9D_MAX] = {
 		.pins = alc882_abit_aw9d_pinfix
 	},
+	[PINFIX_PB_M5210] = {
+		.verbs = pb_m5210_verbs
+	},
 };
 
 static struct snd_pci_quirk alc882_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	{}
 };
-- 
cgit v1.2.3


From 697c373e34613609cb5450f98b91fefb6e910588 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 11:28:02 +0200
Subject: ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs

Call snd_hda_shutup_pins() for power-saving and reboot-notifier in
patch_conexant.c as well as other codecs.  This will reduce the pop
noise in power-save mode.

Reference: bnc#624896
	https://bugzilla.novell.com/show_bug.cgi?id=624896

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c99425ab2e70..d6341f3fef01 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -611,11 +611,23 @@ static int conexant_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int conexant_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	snd_hda_shutup_pins(codec);
+	return 0;
+}
+#endif
+
 static struct hda_codec_ops conexant_patch_ops = {
 	.build_controls = conexant_build_controls,
 	.build_pcms = conexant_build_pcms,
 	.init = conexant_init,
 	.free = conexant_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	.suspend = conexant_suspend,
+#endif
+	.reboot_notify = snd_hda_shutup_pins,
 };
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-- 
cgit v1.2.3


From b08b1637ce1c0196970348bcabf40f04b6b3d58e Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 14:08:25 +0200
Subject: ALSA: hda - Handle pin NID 0x1a on ALC259/269

The pin NID 0x1a should be handled as well as NID 0x1b.
Also added comments.

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 4d3a6f05c703..ce6c3a9c08c0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13158,7 +13158,8 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
-	case 0x1b:
+	case 0x1a: /* ALC259/269 only */
+	case 0x1b: /* ALC259/269 only */
 	case 0x21: /* ALC269vb has this pin, too */
 		dac = 0x03;
 		break;
-- 
cgit v1.2.3


From c7a9434dd6ea74464b0419a274463c914197bc98 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 30 Jul 2010 14:10:43 +0200
Subject: ALSA: hda - Add a warning for ignored pins with ALC259/268/269

The current ALC259/268/269 parser ignores some pins as unhandled,
but user won't notice what goes wrong.  So, added a warning message
for the ignored pins as a hint.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ce6c3a9c08c0..49c04fc8b516 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13164,6 +13164,8 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x03;
 		break;
 	default:
+		snd_printd(KERN_WARNING "hda_codec: "
+			   "ignoring pin 0x%x as unknown\n", nid);
 		return 0;
 	}
 	if (spec->multiout.dac_nids[0] != dac &&
-- 
cgit v1.2.3


From 7bfb9c031ec2d220d48bf679553d6177c2e66625 Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Mon, 2 Aug 2010 13:13:25 +0200
Subject: ALSA: hda - Do not try to create speaker NIDs for ALC268 if there
 aren't any

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 49c04fc8b516..cf14b00155d0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13216,7 +13216,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
 		if (err < 0)
 			return err;
-	} else {
+	} else if (nid) {
 		err = alc268_new_analog_output(spec, nid, "Speaker", 0);
 		if (err < 0)
 			return err;
-- 
cgit v1.2.3


From 607bc3e4888443cdd21a795f7312f64c2de26b5c Mon Sep 17 00:00:00 2001
From: Jerone Young
Date: Tue, 3 Aug 2010 01:46:42 -0500
Subject: ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does
 not have the ability to use SPDIF

The Lenovo X301 does not have the ability to connect to a docking
station to use the SPDIF port. It also does not have the ability to do
SPDIF though the headphone jack or Display Port jacks.

This patch fixes it so this is not exposed for the X301 and users do
think it has the ability to do SPDIF.

I tested both headphone & display port jacks and it is not there. I have
tested this patch and it works great.

Also to add the other Thinkpads have different subsystem codec IDs.
Here are examples:

X301:
http://launchpadlibrarian.net/31561902/Card0.Codecs.codec.0.txt

X200:
http://launchpadlibrarian.net/49055036/Card0.Codecs.codec.0.txt

W500:
http://launchpadlibrarian.net/36276057/Card0.Codecs.codec.0.txt

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d6341f3fef01..df8b19b17308 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2057,6 +2057,10 @@ static int patch_cxt5051(struct hda_codec *codec)
 		break;
 	case CXT5051_LENOVO_X200:
 		spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
+		/* Thinkpad X301 does not have S/PDIF wired and no ability
+		   to use a docking station. */
+		if (codec->subsystem_id == 0x17aa211f)
+			spec->multiout.dig_out_nid = 0;
 		break;
 	case CXT5051_F700:
 		spec->init_verbs[0] = cxt5051_f700_init_verbs;
-- 
cgit v1.2.3


From 68c18697910fdcacea36bd58d2d3d8febfa199a2 Mon Sep 17 00:00:00 2001
From: Jerone Young
Date: Tue, 3 Aug 2010 01:46:44 -0500
Subject: ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed

Just as with the X301. The X300 does not have a way to do SPDIF either.
It does not have a dock connector, nor does it have the SPDIF through
the headphone jack.

This patch fixes it so X300 does not show SPDIF, since it cannot do it.

To add all Lenovo Thinkpads had different codec subsytem IDs:

X300:
http://launchpadlibrarian.net/34862838/Card0.Codecs.codec.0.txt

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index afbe314a5bf3..b697fd2a6f8b 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3662,7 +3662,12 @@ static int patch_ad1984(struct hda_codec *codec)
 		codec->patch_ops.build_pcms = ad1984_build_pcms;
 		break;
 	case AD1984_THINKPAD:
-		spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
+		if (codec->subsystem_id == 0x17aa20fb) {
+			/* Thinpad X300 does not have the ability to do SPDIF,
+			   or attach to docking station to use SPDIF */
+			spec->multiout.dig_out_nid = 0;
+		} else
+			spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
 		spec->input_mux = &ad1984_thinkpad_capture_source;
 		spec->mixers[0] = ad1984_thinkpad_mixers;
 		spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
-- 
cgit v1.2.3


From 32c168c892e2c6936c714d1653ba5e19e07d5c26 Mon Sep 17 00:00:00 2001
From: Anssi Hannula
Date: Tue, 3 Aug 2010 13:28:57 +0300
Subject: ALSA: hda - Set Stream Type in Stream Format according to AES0

Set bit 15 (Stream Type) of HDA Stream Format to 1 (Non-PCM) when IEC958
channel status bit 1 (AES0 & 0x02) is set to 1 (non-audio).

This is a prequisite for HDMI HBR passthrough.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 6 +++++-
 sound/pci/hda/hda_codec.h | 3 ++-
 sound/pci/hda/hda_intel.c | 3 ++-
 3 files changed, 9 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index d9d1c91dfd1b..bd8d7a63d7fe 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3051,7 +3051,8 @@ static struct hda_rate_tbl rate_bits[] = {
 unsigned int snd_hda_calc_stream_format(unsigned int rate,
 					unsigned int channels,
 					unsigned int format,
-					unsigned int maxbps)
+					unsigned int maxbps,
+					unsigned short spdif_ctls)
 {
 	int i;
 	unsigned int val = 0;
@@ -3095,6 +3096,9 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 		return 0;
 	}
 
+	if (spdif_ctls & AC_DIG1_NONAUDIO)
+		val |= 0x8000;
+
 	return val;
 }
 EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 5991d14e1ec0..4797416aa3d9 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -928,7 +928,8 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid);
 unsigned int snd_hda_calc_stream_format(unsigned int rate,
 					unsigned int channels,
 					unsigned int format,
-					unsigned int maxbps);
+					unsigned int maxbps,
+					unsigned short spdif_ctls);
 int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int format);
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1df25cf5ce38..f8a2f5aa4026 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1653,7 +1653,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 	format_val = snd_hda_calc_stream_format(runtime->rate,
 						runtime->channels,
 						runtime->format,
-						hinfo->maxbps);
+						hinfo->maxbps,
+						apcm->codec->spdif_ctls);
 	if (!format_val) {
 		snd_printk(KERN_ERR SFX
 			   "invalid format_val, rate=%d, ch=%d, format=%d\n",
-- 
cgit v1.2.3


From ea87d1c493aba9cf3f645eae0d6d9c0fd44d3189 Mon Sep 17 00:00:00 2001
From: Anssi Hannula
Date: Tue, 3 Aug 2010 13:28:58 +0300
Subject: ALSA: hda - Add support for HDMI HBR passthrough

Passing IEC 61937 encapsulated compressed audio at bitrates over 6.144
Mbps (i.e. more than a single 2-channel 16-bit 192kHz IEC 60958 link)
over HDMI requires the use of HBR Audio Stream Packets instead of Audio
Sample Packets.

Enable HBR mode when the stream has 8 channels and the Non-PCM bit is
set.

If the audio converter is not connected to any HBR-capable pins, return
-EINVAL in prepare().

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.h       |  3 +++
 sound/pci/hda/patch_hdmi.c      | 40 +++++++++++++++++++++++++++++++++++++++-
 sound/pci/hda/patch_intelhdmi.c |  3 +--
 sound/pci/hda/patch_nvhdmi.c    |  3 +--
 4 files changed, 44 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 4797416aa3d9..48b33671e727 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -364,6 +364,9 @@ enum {
 #define AC_DIG2_CC			(0x7f<<0)
 
 /* Pin widget control - 8bit */
+#define AC_PINCTL_EPT			(0x3<<0)
+#define AC_PINCTL_EPT_NATIVE		0
+#define AC_PINCTL_EPT_HBR		3
 #define AC_PINCTL_VREFEN		(0x7<<0)
 #define AC_PINCTL_VREF_HIZ		0	/* Hi-Z */
 #define AC_PINCTL_VREF_50		1	/* 50% */
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2fc53961054e..8534792591fc 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -698,11 +698,48 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
  * Callbacks
  */
 
-static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 			      u32 stream_tag, int format)
 {
+	struct hdmi_spec *spec = codec->spec;
 	int tag;
 	int fmt;
+	int pinctl;
+	int new_pinctl = 0;
+	int i;
+
+	for (i = 0; i < spec->num_pins; i++) {
+		if (spec->pin_cvt[i] != nid)
+			continue;
+		if (!(snd_hda_query_pin_caps(codec, spec->pin[i]) & AC_PINCAP_HBR))
+			continue;
+
+		pinctl = snd_hda_codec_read(codec, spec->pin[i], 0,
+					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+
+		new_pinctl = pinctl & ~AC_PINCTL_EPT;
+		/* Non-PCM, 8 channels */
+		if ((format & 0x8000) && (format & 0x0f) == 7)
+			new_pinctl |= AC_PINCTL_EPT_HBR;
+		else
+			new_pinctl |= AC_PINCTL_EPT_NATIVE;
+
+		snd_printdd("hdmi_setup_stream: "
+			    "NID=0x%x, %spinctl=0x%x\n",
+			    spec->pin[i],
+			    pinctl == new_pinctl ? "" : "new-",
+			    new_pinctl);
+
+		if (pinctl != new_pinctl)
+			snd_hda_codec_write(codec, spec->pin[i], 0,
+					    AC_VERB_SET_PIN_WIDGET_CONTROL,
+					    new_pinctl);
+	}
+
+	if ((format & 0x8000) && (format & 0x0f) == 7 && !new_pinctl) {
+		snd_printdd("hdmi_setup_stream: HBR is not supported\n");
+		return -EINVAL;
+	}
 
 	tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4;
 	fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0);
@@ -722,6 +759,7 @@ static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 	if (fmt != format)
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_STREAM_FORMAT, format);
+	return 0;
 }
 
 /*
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index b81d23e42ace..5972d5e7d01f 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -66,8 +66,7 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 
 	hdmi_setup_audio_infoframe(codec, hinfo->nid, substream);
 
-	hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
-	return 0;
+	return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
 }
 
 static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index b0652acee9b2..a281836fd472 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -202,8 +202,7 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch_89(struct hda_pcm_stream *hinfo,
 
 	hdmi_setup_audio_infoframe(codec, hinfo->nid, substream);
 
-	hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
-	return 0;
+	return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
 }
 
 static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo,
-- 
cgit v1.2.3


From 1b0e372d7b52c9fc96348779015a6db7df7f286e Mon Sep 17 00:00:00 2001
From: Daniel J Blueman
Date: Tue, 3 Aug 2010 11:09:13 +0100
Subject: ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs

Fix HDA beep frequency on IDT 92HD73xx and 92HD71Bxx codecs.
These codecs use the standard beep frequency calculation although the
datasheet says it's linear frequency.

Other IDT/STAC codecs might have the same problem.  They should be
fixed individually later.

Signed-off-by: Daniel J Blueman <daniel.blueman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 12 +++++++++++-
 1 file changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f1e7babd6920..b8d730c47df1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -202,6 +202,7 @@ struct sigmatel_spec {
 	unsigned int spdif_mute: 1;
 	unsigned int check_volume_offset:1;
 	unsigned int auto_mic:1;
+	unsigned int linear_tone_beep:1;
 
 	/* gpio lines */
 	unsigned int eapd_mask;
@@ -3802,7 +3803,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 			return err;
 		if (codec->beep) {
 			/* IDT/STAC codecs have linear beep tone parameter */
-			codec->beep->linear_tone = 1;
+			codec->beep->linear_tone = spec->linear_tone_beep;
 			/* if no beep switch is available, make its own one */
 			caps = query_amp_caps(codec, nid, HDA_OUTPUT);
 			if (!(caps & AC_AMPCAP_MUTE)) {
@@ -5005,6 +5006,7 @@ static int patch_stac9200(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	spec->num_pins = ARRAY_SIZE(stac9200_pin_nids);
 	spec->pin_nids = stac9200_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS,
@@ -5068,6 +5070,7 @@ static int patch_stac925x(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	spec->num_pins = ARRAY_SIZE(stac925x_pin_nids);
 	spec->pin_nids = stac925x_pin_nids;
 
@@ -5153,6 +5156,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 0;
 	codec->slave_dig_outs = stac92hd73xx_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids);
 	spec->pin_nids = stac92hd73xx_pin_nids;
@@ -5300,6 +5304,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
 	spec->digbeep_nid = 0x21;
 	spec->mux_nids = stac92hd83xxx_mux_nids;
@@ -5522,6 +5527,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 0;
 	codec->patch_ops = stac92xx_patch_ops;
 	spec->num_pins = STAC92HD71BXX_NUM_PINS;
 	switch (codec->vendor_id) {
@@ -5779,6 +5785,7 @@ static int patch_stac922x(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	spec->num_pins = ARRAY_SIZE(stac922x_pin_nids);
 	spec->pin_nids = stac922x_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS,
@@ -5883,6 +5890,7 @@ static int patch_stac927x(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	codec->slave_dig_outs = stac927x_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
 	spec->pin_nids = stac927x_pin_nids;
@@ -6018,6 +6026,7 @@ static int patch_stac9205(struct hda_codec *codec)
 
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	spec->num_pins = ARRAY_SIZE(stac9205_pin_nids);
 	spec->pin_nids = stac9205_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS,
@@ -6174,6 +6183,7 @@ static int patch_stac9872(struct hda_codec *codec)
 		return -ENOMEM;
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
+	spec->linear_tone_beep = 1;
 	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
 	spec->pin_nids = stac9872_pin_nids;
 
-- 
cgit v1.2.3


From 92f10b3f5d53f9e35da5285eb8ea4bc88082b71e Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 3 Aug 2010 14:21:00 +0200
Subject: ALSA: hda - Define AC_FMT_* constants

Define constants for the HD-audio stream format bits, and replace the
magic numbers in codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c  | 41 +++++++++++++++++++++++------------------
 sound/pci/hda/hda_codec.h  | 21 +++++++++++++++++++++
 sound/pci/hda/patch_hdmi.c |  9 ++++++---
 3 files changed, 50 insertions(+), 21 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index bd8d7a63d7fe..05e8995f9aec 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3013,26 +3013,31 @@ struct hda_rate_tbl {
 	unsigned int hda_fmt;
 };
 
+/* rate = base * mult / div */
+#define HDA_RATE(base, mult, div) \
+	(AC_FMT_BASE_##base##K | (((mult) - 1) << AC_FMT_MULT_SHIFT) | \
+	 (((div) - 1) << AC_FMT_DIV_SHIFT))
+
 static struct hda_rate_tbl rate_bits[] = {
 	/* rate in Hz, ALSA rate bitmask, HDA format value */
 
 	/* autodetected value used in snd_hda_query_supported_pcm */
-	{ 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */
-	{ 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */
-	{ 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */
-	{ 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */
-	{ 32000, SNDRV_PCM_RATE_32000, 0x0a00 }, /* 2/3 x 48 */
-	{ 44100, SNDRV_PCM_RATE_44100, 0x4000 }, /* 44 */
-	{ 48000, SNDRV_PCM_RATE_48000, 0x0000 }, /* 48 */
-	{ 88200, SNDRV_PCM_RATE_88200, 0x4800 }, /* 2 x 44 */
-	{ 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */
-	{ 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */
-	{ 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */
+	{ 8000, SNDRV_PCM_RATE_8000, HDA_RATE(48, 1, 6) },
+	{ 11025, SNDRV_PCM_RATE_11025, HDA_RATE(44, 1, 4) },
+	{ 16000, SNDRV_PCM_RATE_16000, HDA_RATE(48, 1, 3) },
+	{ 22050, SNDRV_PCM_RATE_22050, HDA_RATE(44, 1, 2) },
+	{ 32000, SNDRV_PCM_RATE_32000, HDA_RATE(48, 2, 3) },
+	{ 44100, SNDRV_PCM_RATE_44100, HDA_RATE(44, 1, 1) },
+	{ 48000, SNDRV_PCM_RATE_48000, HDA_RATE(48, 1, 1) },
+	{ 88200, SNDRV_PCM_RATE_88200, HDA_RATE(44, 2, 1) },
+	{ 96000, SNDRV_PCM_RATE_96000, HDA_RATE(48, 2, 1) },
+	{ 176400, SNDRV_PCM_RATE_176400, HDA_RATE(44, 4, 1) },
+	{ 192000, SNDRV_PCM_RATE_192000, HDA_RATE(48, 4, 1) },
 #define AC_PAR_PCM_RATE_BITS	11
 	/* up to bits 10, 384kHZ isn't supported properly */
 
 	/* not autodetected value */
-	{ 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */
+	{ 9600, SNDRV_PCM_RATE_KNOT, HDA_RATE(48, 1, 5) },
 
 	{ 0 } /* terminator */
 };
@@ -3075,20 +3080,20 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 
 	switch (snd_pcm_format_width(format)) {
 	case 8:
-		val |= 0x00;
+		val |= AC_FMT_BITS_8;
 		break;
 	case 16:
-		val |= 0x10;
+		val |= AC_FMT_BITS_16;
 		break;
 	case 20:
 	case 24:
 	case 32:
 		if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE)
-			val |= 0x40;
+			val |= AC_FMT_BITS_32;
 		else if (maxbps >= 24)
-			val |= 0x30;
+			val |= AC_FMT_BITS_24;
 		else
-			val |= 0x20;
+			val |= AC_FMT_BITS_20;
 		break;
 	default:
 		snd_printdd("invalid format width %d\n",
@@ -3097,7 +3102,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 	}
 
 	if (spdif_ctls & AC_DIG1_NONAUDIO)
-		val |= 0x8000;
+		val |= AC_FMT_TYPE_NON_PCM;
 
 	return val;
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 48b33671e727..46f75bccf0d3 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -224,6 +224,27 @@ enum {
 /* Input converter SDI select */
 #define AC_SDI_SELECT			(0xf<<0)
 
+/* stream format id */
+#define AC_FMT_CHAN_SHIFT		0
+#define AC_FMT_CHAN_MASK		(0x0f << 0)
+#define AC_FMT_BITS_SHIFT		4
+#define AC_FMT_BITS_MASK		(7 << 4)
+#define AC_FMT_BITS_8			(0 << 4)
+#define AC_FMT_BITS_16			(1 << 4)
+#define AC_FMT_BITS_20			(2 << 4)
+#define AC_FMT_BITS_24			(3 << 4)
+#define AC_FMT_BITS_32			(4 << 4)
+#define AC_FMT_DIV_SHIFT		8
+#define AC_FMT_DIV_MASK			(7 << 8)
+#define AC_FMT_MULT_SHIFT		11
+#define AC_FMT_MULT_MASK		(7 << 11)
+#define AC_FMT_BASE_SHIFT		14
+#define AC_FMT_BASE_48K			(0 << 14)
+#define AC_FMT_BASE_44K			(1 << 14)
+#define AC_FMT_TYPE_SHIFT		15
+#define AC_FMT_TYPE_PCM			(0 << 15)
+#define AC_FMT_TYPE_NON_PCM		(1 << 15)
+
 /* Unsolicited response control */
 #define AC_UNSOL_TAG			(0x3f<<0)
 #define AC_UNSOL_ENABLED		(1<<7)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 8534792591fc..522e0748ee99 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -698,6 +698,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
  * Callbacks
  */
 
+/* HBR should be Non-PCM, 8 channels */
+#define is_hbr_format(format) \
+	((format & AC_FMT_TYPE_NON_PCM) && (format & AC_FMT_CHAN_MASK) == 7)
+
 static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 			      u32 stream_tag, int format)
 {
@@ -718,8 +722,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 
 		new_pinctl = pinctl & ~AC_PINCTL_EPT;
-		/* Non-PCM, 8 channels */
-		if ((format & 0x8000) && (format & 0x0f) == 7)
+		if (is_hbr_format(format))
 			new_pinctl |= AC_PINCTL_EPT_HBR;
 		else
 			new_pinctl |= AC_PINCTL_EPT_NATIVE;
@@ -736,7 +739,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 					    new_pinctl);
 	}
 
-	if ((format & 0x8000) && (format & 0x0f) == 7 && !new_pinctl) {
+	if (is_hbr_format(format) && !new_pinctl) {
 		snd_printdd("hdmi_setup_stream: HBR is not supported\n");
 		return -EINVAL;
 	}
-- 
cgit v1.2.3


From 08af495f22f43eff3b5a347dc10384ebcf356e41 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 3 Aug 2010 14:39:04 +0200
Subject: ALSA: hda - More relax for pending period handling

Since the pending periods are often bogus and take long time until
actually processed, it often results in a high CPU usage of the hd-audio
workq.  Overall it's better to have low CPU consumption by avoiding a
too tight loop rather than the wake-up timing accuracy.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f8a2f5aa4026..66d420212d9a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1961,7 +1961,7 @@ static void azx_irq_pending_work(struct work_struct *work)
 		spin_unlock_irq(&chip->reg_lock);
 		if (!pending)
 			return;
-		cond_resched();
+		msleep(1);
 	}
 }
 
-- 
cgit v1.2.3


From e096c8e6d5ed965f346d94befbbec2275dde3621 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 3 Aug 2010 17:20:35 +0200
Subject: ALSA: hda - Add PC-beep whitelist for an Intel board

An Intel board needs a white-list entry to enable PC-beep.
Otherwise the driver misdetects (due to bogus BIOS info) and ignores
the PC-beep on 2.6.35.

Reported-and-tested-by: Leandro Lucarella <luca@llucax.com.ar>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cf14b00155d0..6c588ef26685 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5327,6 +5327,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
 
 static struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
+	SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
 	{}
 };
 
-- 
cgit v1.2.3


From 748cce431eb413e794c8f1d1974b78b47a6174ef Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 4 Aug 2010 07:37:39 +0200
Subject: ALSA: hda - Fix initial capsrc selection in patch_alc269()

In patch_alc269(), we initialize the primary capsrc so that the device
works from the beginning.  It issues CONNECT_SEL verb no matter which
widget is although some widget (e.g. 0x23) has no connection selection
but a mixer, which requires unmuting instead.

This patch fixes the initialization of capsrc by re-using the code as
a helper function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++-----------
 1 file changed, 16 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6c588ef26685..c8070620a4d8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5166,6 +5166,19 @@ static void fixup_automic_adc(struct hda_codec *codec)
 	spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
+/* select or unmute the given capsrc route */
+static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap,
+				    int idx)
+{
+	if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+		snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+					 HDA_AMP_MUTE, 0);
+	} else {
+		snd_hda_codec_write_cache(codec, cap, 0,
+					  AC_VERB_SET_CONNECT_SEL, idx);
+	}
+}
+
 /* set the default connection to that pin */
 static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
 {
@@ -5180,14 +5193,7 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
 		idx = get_connection_index(codec, cap, pin);
 		if (idx < 0)
 			continue;
-		/* select or unmute this route */
-		if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
-			snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
-						 HDA_AMP_MUTE, 0);
-		} else {
-			snd_hda_codec_write_cache(codec, cap, 0,
-					  AC_VERB_SET_CONNECT_SEL, idx);
-		}
+		select_or_unmute_capsrc(codec, cap, idx);
 		return i; /* return the found index */
 	}
 	return -1; /* not found */
@@ -14364,9 +14370,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	/* set default input source */
 	if (!spec->dual_adc_switch)
-		snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
-				  0, AC_VERB_SET_CONNECT_SEL,
-				  spec->input_mux->items[0].index);
+		select_or_unmute_capsrc(codec, spec->capsrc_nids[0],
+					spec->input_mux->items[0].index);
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
-- 
cgit v1.2.3


From b9619230e1f55a763bc41848c1cd971a394c878c Mon Sep 17 00:00:00 2001
From: Ondrej Zary
Date: Tue, 3 Aug 2010 23:57:05 +0200
Subject: ALSA: als4000: enable burst mode

Enable burst mode to prevent dropouts during high PCI bus usage.
The card is useless in X without this because of dropouts when anything moves
on the screen (at least with PCI VGA card). Enabling this is also recommended
by the datasheet (page 48).

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/als4000.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 6cf1de8042e8..036a9ba8e1a5 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -763,9 +763,9 @@ static void snd_als4000_configure(struct snd_sb *chip)
 	/* SPECS_PAGE: 39 */
 	for (i = ALS4K_GCR91_DMA0_ADDR; i <= ALS4K_GCR96_DMA3_MODE_COUNT; ++i)
 		snd_als4k_gcr_write(chip, i, 0);
-	
+	/* enable burst mode to prevent dropouts during high PCI bus usage */
 	snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL,
-		snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL));
+		snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL) | 0x04);
 	spin_unlock_irq(&chip->reg_lock);
 }
 
-- 
cgit v1.2.3


From c4685849b4d725ab80cd29f5e09f5f128b4724b5 Mon Sep 17 00:00:00 2001
From: Ondrej Zary
Date: Wed, 4 Aug 2010 21:56:44 +0200
Subject: ALSA: als4000: Fix potentially invalid DMA mode setup

My previous patch assumed that the DMA mode (represented by 3 lowest bits of
ALS4K_GCR99_DMA_EMULATION_CTRL register) is set to the default value 0. If
that's not the case, it might result in invalid mode to be set.
This patch fixes this potential problem.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/als4000.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 036a9ba8e1a5..0e247cb90ecc 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -765,7 +765,7 @@ static void snd_als4000_configure(struct snd_sb *chip)
 		snd_als4k_gcr_write(chip, i, 0);
 	/* enable burst mode to prevent dropouts during high PCI bus usage */
 	snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL,
-		snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL) | 0x04);
+		(snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL) & ~0x07) | 0x04);
 	spin_unlock_irq(&chip->reg_lock);
 }
 
-- 
cgit v1.2.3


From fc091769a5aa65c045bfbda149c424ba33d0abbb Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 4 Aug 2010 23:53:36 +0200
Subject: ALSA: hda - Add pin-fix for HP dc5750

The NID 0x11 on HP dc5750 with ALC260 should be a speaker although BIOS
gives it as a line-out.  This patch adds a quirk to fix the pin config
so that the real line-out is used properly.

Reference: bnc#624118
	https://bugzilla.novell.com/show_bug.cgi?id=624118

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++++++++++++++
 1 file changed, 29 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c8070620a4d8..6ac53f7de549 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6791,6 +6791,29 @@ static struct hda_amp_list alc260_loopbacks[] = {
 };
 #endif
 
+/*
+ * Pin config fixes
+ */
+enum {
+	PINFIX_HP_DC5750,
+};
+
+static struct alc_pincfg alc260_hp_dc5750_pinfix[] = {
+	{ 0x11, 0x90130110 }, /* speaker */
+	{ }
+};
+
+static const struct alc_fixup alc260_fixups[] = {
+	[PINFIX_HP_DC5750] = {
+		.pins = alc260_hp_dc5750_pinfix
+	},
+};
+
+static struct snd_pci_quirk alc260_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750),
+	{}
+};
+
 /*
  * ALC260 configurations
  */
@@ -6990,6 +7013,9 @@ static int patch_alc260(struct hda_codec *codec)
 		board_config = ALC260_AUTO;
 	}
 
+	if (board_config == ALC260_AUTO)
+		alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 1);
+
 	if (board_config == ALC260_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc260_parse_auto_config(codec);
@@ -7035,6 +7061,9 @@ static int patch_alc260(struct hda_codec *codec)
 	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
 
+	if (board_config == ALC260_AUTO)
+		alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 0);
+
 	spec->vmaster_nid = 0x08;
 
 	codec->patch_ops = alc_patch_ops;
-- 
cgit v1.2.3