From 8f398ae72fc7e03356fc1ee6b54beef79ba6be6a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 23 Jul 2011 18:57:11 +0200
Subject: ALSA: hda - Fix DAC filling for multi-connection pins in Realtek
 parser

Fix a regression in the DAC filling code in patch_realtek.c.  The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 52ce07534e5b..569d2aa4eeb5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2801,7 +2801,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	int i;
 
  again:
-	spec->multiout.num_dacs = 0;
+	/* set num_dacs once to full for alc_auto_look_for_dac() */
+	spec->multiout.num_dacs = cfg->line_outs;
 	spec->multiout.hp_nid = 0;
 	spec->multiout.extra_out_nid[0] = 0;
 	memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
@@ -2834,6 +2835,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 		}
 	}
 
+	/* re-count num_dacs and squash invalid entries */
+	spec->multiout.num_dacs = 0;
 	for (i = 0; i < cfg->line_outs; i++) {
 		if (spec->private_dac_nids[i])
 			spec->multiout.num_dacs++;
-- 
cgit v1.2.3


From acb03d440b8a723181e1d45e3517e43cb0792f8a Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Date: Sat, 23 Jul 2011 12:36:25 +1200
Subject: ALSA: Make snd_pcm_debug_name usable outside pcm_lib

Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.

[minor coding-style fixes by tiwai]

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 21 ---------------------
 1 file changed, 21 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index b941d2541dda..eae62ebbd295 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -41,31 +41,10 @@
 #include <sound/tlv.h>
 #include <sound/hwdep.h>
 
-
 MODULE_LICENSE("GPL");
 MODULE_AUTHOR("AudioScience inc. <support@audioscience.com>");
 MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx");
 
-#if defined CONFIG_SND_DEBUG
-/* copied from pcm_lib.c, hope later patch will make that version public
-and this copy can be removed */
-static inline void
-snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
-{
-	snprintf(buf, size, "pcmC%dD%d%c:%d",
-		 substream->pcm->card->number,
-		 substream->pcm->device,
-		 substream->stream ? 'c' : 'p',
-		 substream->number);
-}
-#else
-static inline void
-snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
-{
-	*buf = 0;
-}
-#endif
-
 #if defined CONFIG_SND_DEBUG_VERBOSE
 /**
  * snd_printddd - very verbose debug printk
-- 
cgit v1.2.3


From 0c27c1805269f9ff01cc1d77752a662065ebcfe5 Mon Sep 17 00:00:00 2001
From: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Date: Fri, 22 Jul 2011 17:50:37 -0500
Subject: ALSA: hda - Add support of the 4 internal speakers on certain HP
 laptops

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 51 +++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 50 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 56425a53cf1b..00ea2bd6bc14 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -95,6 +95,7 @@ enum {
 	STAC_92HD83XXX_PWR_REF,
 	STAC_DELL_S14,
 	STAC_92HD83XXX_HP,
+	STAC_92HD83XXX_HP_cNB11_INTQUAD,
 	STAC_HP_DV7_4000,
 	STAC_92HD83XXX_MODELS
 };
@@ -1636,10 +1637,17 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = {
 	0x40f000f0, 0x40f000f0,
 };
 
+static const unsigned int hp_cNB11_intquad_pin_configs[10] = {
+	0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110,
+	0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130,
+	0x40f000f0, 0x40f000f0,
+};
+
 static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
 	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
 	[STAC_DELL_S14] = dell_s14_pin_configs,
+	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs,
 	[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
 };
 
@@ -1649,6 +1657,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
 	[STAC_DELL_S14] = "dell-s14",
 	[STAC_92HD83XXX_HP] = "hp",
+	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
 	[STAC_HP_DV7_4000] = "hp-dv7-4000",
 };
 
@@ -1661,7 +1670,47 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba,
 		      "unknown Dell", STAC_DELL_S14),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600,
-		      "HP", STAC_92HD83XXX_HP),
+			  "HP", STAC_92HD83XXX_HP),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1657,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1658,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1659,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165A,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165B,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355B,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355C,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355D,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355E,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355F,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3560,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358B,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358C,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358D,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3591,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3592,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From d02667e6206fb3be0990c38af8447a4ed2b74c11 Mon Sep 17 00:00:00 2001
From: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Date: Fri, 22 Jul 2011 18:18:15 -0500
Subject: ALSA: hda - Fix invalid mute led state on resume of IDT codecs

Codec state is not restored immediately on resume but on the first
access when power-save is enabled.  That leads to an invalid mute led
state after resume until either sound is played or some control is
changed.  This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required.  And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      |  2 ++
 sound/pci/hda/hda_codec.h      |  3 +++
 sound/pci/hda/patch_sigmatel.c | 12 ++++++++++++
 3 files changed, 17 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 9c27a3a4c4d5..c0e83ed0b351 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -5103,6 +5103,8 @@ int snd_hda_resume(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
+		if (codec->patch_ops.pre_resume)
+			codec->patch_ops.pre_resume(codec);
 		if (snd_hda_codec_needs_resume(codec))
 			hda_call_codec_resume(codec);
 	}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index f465e07a4879..82161466d3b0 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -712,6 +712,9 @@ struct hda_codec_ops {
 	int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
 #endif
 	void (*reboot_notify)(struct hda_codec *codec);
+#ifdef SND_HDA_NEEDS_RESUME
+	int (*pre_resume)(struct hda_codec *codec);
+#endif
 };
 
 /* record for amp information cache */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 00ea2bd6bc14..c4a6ecb8e085 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4935,6 +4935,17 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer,
 #endif
 
 #ifdef SND_HDA_NEEDS_RESUME
+static int stac92xx_pre_resume(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	/* sync mute LED */
+	if (spec->gpio_led)
+		stac_gpio_set(codec, spec->gpio_mask,
+				spec->gpio_dir, spec->gpio_data);
+	return 0;
+}
+
 static int stac92xx_resume(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -5013,6 +5024,7 @@ static const struct hda_codec_ops stac92xx_patch_ops = {
 #ifdef SND_HDA_NEEDS_RESUME
 	.suspend = stac92xx_suspend,
 	.resume = stac92xx_resume,
+	.pre_resume = stac92xx_pre_resume,
 #endif
 	.reboot_notify = stac92xx_shutup,
 };
-- 
cgit v1.2.3


From 7df1ce1a8197a4afec78584f56e74ab84dcab97c Mon Sep 17 00:00:00 2001
From: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Date: Mon, 25 Jul 2011 17:52:57 -0500
Subject: ALSA: hda - Make sure mute led reflects master mute state

This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 53 ++++++++++++++++++++++++++++--------------
 1 file changed, 35 insertions(+), 18 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c4a6ecb8e085..8f80796c366f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4961,29 +4961,19 @@ static int stac92xx_resume(struct hda_codec *codec)
 			stac_issue_unsol_event(codec,
 					       spec->autocfg.line_out_pins[0]);
 	}
-	/* sync mute LED */
-	if (spec->gpio_led)
-		hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 /*
- * using power check for controlling mute led of HP notebooks
- * check for mute state only on Speakers (nid = 0x10)
- *
- * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise
- * the LED is NOT working properly !
- *
- * Changed name to reflect that it now works for any designated
- * model, not just HP HDX.
+ * For this feature CONFIG_SND_HDA_POWER_SAVE is needed
+ * as mute LED state is updated in check_power_status hook
  */
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static int stac92xx_hp_check_power_status(struct hda_codec *codec,
-					      hda_nid_t nid)
+static int stac92xx_update_led_status(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int i, muted = 1;
+	int i, num_ext_dacs, muted = 1;
+	hda_nid_t nid;
 
 	for (i = 0; i < spec->multiout.num_dacs; i++) {
 		nid = spec->multiout.dac_nids[i];
@@ -4993,6 +4983,22 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 			break;
 		}
 	}
+	if (muted && spec->multiout.hp_nid)
+		if (!(snd_hda_codec_amp_read(codec,
+				spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) &
+					HDA_AMP_MUTE)) {
+			muted = 0; /* HP is not muted */
+		}
+	num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid);
+	for (i = 0; muted && i < num_ext_dacs; i++) {
+		nid = spec->multiout.extra_out_nid[i];
+		if (nid == 0)
+			break;
+		if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
+		      HDA_AMP_MUTE)) {
+			muted = 0; /* extra output is not muted */
+		}
+	}
 	if (muted)
 		spec->gpio_data &= ~spec->gpio_led; /* orange */
 	else
@@ -5006,6 +5012,17 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
 	return 0;
 }
+
+/*
+ * use power check for controlling mute led of HP notebooks
+ */
+static int stac92xx_check_power_status(struct hda_codec *codec,
+					      hda_nid_t nid)
+{
+	stac92xx_update_led_status(codec);
+
+	return 0;
+}
 #endif
 
 static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
@@ -5543,7 +5560,7 @@ again:
 		spec->gpio_data |= spec->gpio_led;
 		/* register check_power_status callback. */
 		codec->patch_ops.check_power_status =
-			stac92xx_hp_check_power_status;
+			stac92xx_check_power_status;
 	}
 #endif	
 
@@ -5871,7 +5888,7 @@ again:
 		spec->gpio_data |= spec->gpio_led;
 		/* register check_power_status callback. */
 		codec->patch_ops.check_power_status =
-			stac92xx_hp_check_power_status;
+			stac92xx_check_power_status;
 	}
 #endif	
 
-- 
cgit v1.2.3


From 2a43952a99072f43c92355882b7965c8762ae3f3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 26 Jul 2011 09:52:50 +0200
Subject: ALSA: hda - Make CONFIG_SND_HDA_POWER_SAVE depending on CONFIG_PM

It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig          |  1 +
 sound/pci/hda/hda_codec.c      | 16 ++++++++--------
 sound/pci/hda/hda_codec.h      | 12 +++---------
 sound/pci/hda/hda_local.h      |  2 +-
 sound/pci/hda/patch_analog.c   |  4 ++--
 sound/pci/hda/patch_realtek.c  | 10 +++++-----
 sound/pci/hda/patch_sigmatel.c |  6 +++---
 sound/pci/hda/patch_via.c      |  4 ++--
 8 files changed, 25 insertions(+), 30 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 7489b4608551..bb7e102d6726 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -243,6 +243,7 @@ config SND_HDA_GENERIC
 
 config SND_HDA_POWER_SAVE
 	bool "Aggressive power-saving on HD-audio"
+	depends on PM
 	help
 	  Say Y here to enable more aggressive power-saving mode on
 	  HD-audio driver.  The power-saving timeout can be configured
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c0e83ed0b351..27b0c78abb5b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1101,7 +1101,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
@@ -1499,7 +1499,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
 	}
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
@@ -1838,7 +1838,7 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo);
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /**
  * snd_hda_codec_resume_amp - Resume all AMP commands from the cache
  * @codec: HD-audio codec
@@ -1868,7 +1868,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp);
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir,
 			     unsigned int ofs)
@@ -3082,7 +3082,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 }
 EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls);
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /*
  * command cache
  */
@@ -3199,7 +3199,7 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec,
 					  seq->param);
 }
 EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache);
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 /*
  * set power state of the codec
@@ -3274,7 +3274,7 @@ static void hda_exec_init_verbs(struct hda_codec *codec)
 static inline void hda_exec_init_verbs(struct hda_codec *codec) {}
 #endif
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /*
  * call suspend and power-down; used both from PM and power-save
  */
@@ -3315,7 +3315,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 		snd_hda_codec_resume_cache(codec);
 	}
 }
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 
 /**
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 82161466d3b0..663aa4fc384a 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -26,10 +26,6 @@
 #include <sound/pcm.h>
 #include <sound/hwdep.h>
 
-#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE)
-#define SND_HDA_NEEDS_RESUME	/* resume control code is required */
-#endif
-
 /*
  * nodes
  */
@@ -704,17 +700,15 @@ struct hda_codec_ops {
 	int (*init)(struct hda_codec *codec);
 	void (*free)(struct hda_codec *codec);
 	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	int (*suspend)(struct hda_codec *codec, pm_message_t state);
+	int (*pre_resume)(struct hda_codec *codec);
 	int (*resume)(struct hda_codec *codec);
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
 #endif
 	void (*reboot_notify)(struct hda_codec *codec);
-#ifdef SND_HDA_NEEDS_RESUME
-	int (*pre_resume)(struct hda_codec *codec);
-#endif
 };
 
 /* record for amp information cache */
@@ -930,7 +924,7 @@ void snd_hda_sequence_write(struct hda_codec *codec,
 int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
 
 /* cached write */
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 			      int direct, unsigned int verb, unsigned int parm);
 void snd_hda_sequence_write_cache(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 88b277e97409..2e7ac31afa8d 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -131,7 +131,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 			     int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 			     int dir, int idx, int mask, int val);
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 void snd_hda_codec_resume_amp(struct hda_codec *codec);
 #endif
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1362c8ba4d1f..8648917acffb 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -563,7 +563,7 @@ static void ad198x_free(struct hda_codec *codec)
 	snd_hda_detach_beep_device(codec);
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	ad198x_shutup(codec);
@@ -579,7 +579,7 @@ static const struct hda_codec_ops ad198x_patch_ops = {
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.check_power_status = ad198x_check_power_status,
 #endif
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.suspend = ad198x_suspend,
 #endif
 	.reboot_notify = ad198x_shutup,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 569d2aa4eeb5..694327ae8b71 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2386,7 +2386,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state)
 }
 #endif
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int alc_resume(struct hda_codec *codec)
 {
 	msleep(150); /* to avoid pop noise */
@@ -2406,7 +2406,7 @@ static const struct hda_codec_ops alc_patch_ops = {
 	.init = alc_init,
 	.free = alc_free,
 	.unsol_event = alc_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.resume = alc_resume,
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -4413,7 +4413,7 @@ static void alc269_shutup(struct hda_codec *codec)
 	}
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int alc269_resume(struct hda_codec *codec)
 {
 	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
@@ -4436,7 +4436,7 @@ static int alc269_resume(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 static void alc269_fixup_hweq(struct hda_codec *codec,
 			       const struct alc_fixup *fix, int action)
@@ -4728,7 +4728,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	codec->patch_ops.resume = alc269_resume;
 #endif
 	if (board_config == ALC_MODEL_AUTO)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8f80796c366f..fcf4c7142103 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4934,7 +4934,7 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer,
 #define stac927x_proc_hook	NULL
 #endif
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int stac92xx_pre_resume(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -5030,7 +5030,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 	stac92xx_shutup(codec);
 	return 0;
 }
-#endif
+#endif /* CONFIG_PM */
 
 static const struct hda_codec_ops stac92xx_patch_ops = {
 	.build_controls = stac92xx_build_controls,
@@ -5038,7 +5038,7 @@ static const struct hda_codec_ops stac92xx_patch_ops = {
 	.init = stac92xx_init,
 	.free = stac92xx_free,
 	.unsol_event = stac92xx_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.suspend = stac92xx_suspend,
 	.resume = stac92xx_resume,
 	.pre_resume = stac92xx_pre_resume,
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index f38160b00e16..84d8798bf33a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1708,7 +1708,7 @@ static void via_unsol_event(struct hda_codec *codec,
 		via_gpio_control(codec);
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int via_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct via_spec *spec = codec->spec;
@@ -1736,7 +1736,7 @@ static const struct hda_codec_ops via_patch_ops = {
 	.init = via_init,
 	.free = via_free,
 	.unsol_event = via_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.suspend = via_suspend,
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-- 
cgit v1.2.3


From e581f3dba509f6d48e4939aa70e9b768aa5fd4f3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 26 Jul 2011 10:19:20 +0200
Subject: ALSA: hda - Add post_suspend patch ops

Add a new ops, post_suspend(), which is called after suspend() ops is
performed.  This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 18 ++++++++----------
 sound/pci/hda/hda_codec.h |  1 +
 2 files changed, 9 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 27b0c78abb5b..056cd9ade1fb 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -91,8 +91,10 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void hda_power_work(struct work_struct *work);
 static void hda_keep_power_on(struct hda_codec *codec);
+#define hda_codec_is_power_on(codec)	((codec)->power_on)
 #else
 static inline void hda_keep_power_on(struct hda_codec *codec) {}
+#define hda_codec_is_power_on(codec)	1
 #endif
 
 /**
@@ -4376,11 +4378,8 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus)
 	if (!bus)
 		return;
 	list_for_each_entry(codec, &bus->codec_list, list) {
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-		if (!codec->power_on)
-			continue;
-#endif
-		if (codec->patch_ops.reboot_notify)
+		if (hda_codec_is_power_on(codec) &&
+		    codec->patch_ops.reboot_notify)
 			codec->patch_ops.reboot_notify(codec);
 	}
 }
@@ -5079,11 +5078,10 @@ int snd_hda_suspend(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-		if (!codec->power_on)
-			continue;
-#endif
-		hda_call_codec_suspend(codec);
+		if (hda_codec_is_power_on(codec))
+			hda_call_codec_suspend(codec);
+		if (codec->patch_ops.post_suspend)
+			codec->patch_ops.post_suspend(codec);
 	}
 	return 0;
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 663aa4fc384a..c7ca753d94ee 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -702,6 +702,7 @@ struct hda_codec_ops {
 	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
 #ifdef CONFIG_PM
 	int (*suspend)(struct hda_codec *codec, pm_message_t state);
+	int (*post_suspend)(struct hda_codec *codec);
 	int (*pre_resume)(struct hda_codec *codec);
 	int (*resume)(struct hda_codec *codec);
 #endif
-- 
cgit v1.2.3


From 4d7fbdbcb1d563b1822c74da3c9e4aa4523d8d6d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 26 Jul 2011 10:33:10 +0200
Subject: ALSA: hda - Allow codec-specific set_power_state ops

The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.

This patch adds a new codec ops, set_power_state() to override the system
default function.  For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.

As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 78 ++++++++++++++++++++----------------------
 sound/pci/hda/hda_codec.h      |  5 +++
 sound/pci/hda/patch_conexant.c | 14 ++++++++
 3 files changed, 56 insertions(+), 41 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 056cd9ade1fb..3e7850c238c3 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3203,51 +3203,30 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec,
 EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache);
 #endif /* CONFIG_PM */
 
-/*
- * set power state of the codec
- */
-static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
-				unsigned int power_state)
+void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
+				    unsigned int power_state,
+				    bool eapd_workaround)
 {
-	hda_nid_t nid;
+	hda_nid_t nid = codec->start_nid;
 	int i;
 
-	/* this delay seems necessary to avoid click noise at power-down */
-	if (power_state == AC_PWRST_D3)
-		msleep(100);
-	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
-			    power_state);
-	/* partial workaround for "azx_get_response timeout" */
-	if (power_state == AC_PWRST_D0 &&
-	    (codec->vendor_id & 0xffff0000) == 0x14f10000)
-		msleep(10);
-
-	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int wcaps = get_wcaps(codec, nid);
-		if (wcaps & AC_WCAP_POWER) {
-			unsigned int wid_type = get_wcaps_type(wcaps);
-			if (power_state == AC_PWRST_D3 &&
-			    wid_type == AC_WID_PIN) {
-				unsigned int pincap;
-				/*
-				 * don't power down the widget if it controls
-				 * eapd and EAPD_BTLENABLE is set.
-				 */
-				pincap = snd_hda_query_pin_caps(codec, nid);
-				if (pincap & AC_PINCAP_EAPD) {
-					int eapd = snd_hda_codec_read(codec,
-						nid, 0,
+		if (!(wcaps & AC_WCAP_POWER))
+			continue;
+		/* don't power down the widget if it controls eapd and
+		 * EAPD_BTLENABLE is set.
+		 */
+		if (eapd_workaround && power_state == AC_PWRST_D3 &&
+		    get_wcaps_type(wcaps) == AC_WID_PIN &&
+		    (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) {
+			int eapd = snd_hda_codec_read(codec, nid, 0,
 						AC_VERB_GET_EAPD_BTLENABLE, 0);
-					eapd &= 0x02;
-					if (eapd)
-						continue;
-				}
-			}
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    power_state);
+			if (eapd & 0x02)
+				continue;
 		}
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE,
+				    power_state);
 	}
 
 	if (power_state == AC_PWRST_D0) {
@@ -3264,6 +3243,26 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 		} while (time_after_eq(end_time, jiffies));
 	}
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all);
+
+/*
+ * set power state of the codec
+ */
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state)
+{
+	if (codec->patch_ops.set_power_state) {
+		codec->patch_ops.set_power_state(codec, fg, power_state);
+		return;
+	}
+
+	/* this delay seems necessary to avoid click noise at power-down */
+	if (power_state == AC_PWRST_D3)
+		msleep(100);
+	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+			    power_state);
+	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+}
 
 #ifdef CONFIG_SND_HDA_HWDEP
 /* execute additional init verbs */
@@ -4073,9 +4072,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec,
 EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls);
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
-				unsigned int power_state);
-
 static void hda_power_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c7ca753d94ee..755f2b0f9d8e 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -700,6 +700,8 @@ struct hda_codec_ops {
 	int (*init)(struct hda_codec *codec);
 	void (*free)(struct hda_codec *codec);
 	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+	void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state);
 #ifdef CONFIG_PM
 	int (*suspend)(struct hda_codec *codec, pm_message_t state);
 	int (*post_suspend)(struct hda_codec *codec);
@@ -1006,6 +1008,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
  */
 void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
 void snd_hda_bus_reboot_notify(struct hda_bus *bus);
+void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
+				    unsigned int power_state,
+				    bool eapd_workaround);
 
 /*
  * power management
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 884f67b8f4e0..502fc9499453 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -446,6 +446,19 @@ static int conexant_init_jacks(struct hda_codec *codec)
 	return 0;
 }
 
+static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg,
+			       unsigned int power_state)
+{
+	if (power_state == AC_PWRST_D3)
+		msleep(100);
+	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+			    power_state);
+	/* partial workaround for "azx_get_response timeout" */
+	if (power_state == AC_PWRST_D0)
+		msleep(10);
+	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+}
+
 static int conexant_init(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -588,6 +601,7 @@ static const struct hda_codec_ops conexant_patch_ops = {
 	.build_pcms = conexant_build_pcms,
 	.init = conexant_init,
 	.free = conexant_free,
+	.set_power_state = conexant_set_power,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.suspend = conexant_suspend,
 #endif
-- 
cgit v1.2.3


From 56487c279fe9fc23c5f15e2b935eb896ab7ba280 Mon Sep 17 00:00:00 2001
From: Tim Howe <tim.howe@cirrus.com>
Date: Fri, 22 Jul 2011 16:41:00 -0500
Subject: ALSA: hda - Cirrus Logic CS421x support

This update includes the changes necessary for supporting the
CS421x family of codecs.  Previously this file only supported
the CS420x family of codecs.

This file also contains init verbs to correct several issues in
the CS421x hardware.

Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.

Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.

[Fix const usages and adaption for new APIs by tiwai]

Signed-off-by: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 743 +++++++++++++++++++++++++++++++++++++++++--
 1 file changed, 709 insertions(+), 34 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 7f93739b1e33..47d6ffc9b5b5 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -25,6 +25,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include <sound/tlv.h>
 
 /*
  */
@@ -61,9 +62,15 @@ struct cs_spec {
 
 	unsigned int hp_detect:1;
 	unsigned int mic_detect:1;
+	/* CS421x */
+	unsigned int spdif_detect:1;
+	unsigned int sense_b:1;
+	hda_nid_t vendor_nid;
+	struct hda_input_mux input_mux;
+	unsigned int last_input;
 };
 
-/* available models */
+/* available models with CS420x */
 enum {
 	CS420X_MBP53,
 	CS420X_MBP55,
@@ -72,6 +79,12 @@ enum {
 	CS420X_MODELS
 };
 
+/* CS421x boards */
+enum {
+	CS421X_CDB4210,
+	CS421X_MODELS
+};
+
 /* Vendor-specific processing widget */
 #define CS420X_VENDOR_NID	0x11
 #define CS_DIG_OUT1_PIN_NID	0x10
@@ -111,21 +124,42 @@ enum {
 /* 0x0009 - 0x0014 -> 12 test regs */
 /* 0x0015 - visibility reg */
 
+/*
+ * Cirrus Logic CS4210
+ *
+ * 1 DAC => HP(sense) / Speakers,
+ * 1 ADC <= LineIn(sense) / MicIn / DMicIn,
+ * 1 SPDIF OUT => SPDIF Trasmitter(sense)
+*/
+#define CS4210_DAC_NID		0x02
+#define CS4210_ADC_NID		0x03
+#define CS421X_VENDOR_NID	0x0B
+#define CS421X_DMIC_PIN_NID	0x09 /* Port E */
+#define CS421X_SPDIF_PIN_NID	0x0A /* Port H */
+
+#define CS421X_IDX_DEV_CFG	0x01
+#define CS421X_IDX_ADC_CFG	0x02
+#define CS421X_IDX_DAC_CFG	0x03
+#define CS421X_IDX_SPK_CTL	0x04
+
+#define SPDIF_EVENT		0x04
 
 static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
 {
-	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+	struct cs_spec *spec = codec->spec;
+	snd_hda_codec_write(codec, spec->vendor_nid, 0,
 			    AC_VERB_SET_COEF_INDEX, idx);
-	return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0,
+	return snd_hda_codec_read(codec, spec->vendor_nid, 0,
 				  AC_VERB_GET_PROC_COEF, 0);
 }
 
 static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
 				      unsigned int coef)
 {
-	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+	struct cs_spec *spec = codec->spec;
+	snd_hda_codec_write(codec, spec->vendor_nid, 0,
 			    AC_VERB_SET_COEF_INDEX, idx);
-	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+	snd_hda_codec_write(codec, spec->vendor_nid, 0,
 			    AC_VERB_SET_PROC_COEF, coef);
 }
 
@@ -347,15 +381,12 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int type;
-		int idx;
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		idx = snd_hda_get_conn_index(codec, nid, pin, 0);
-		if (idx >= 0) {
-			*idxp = idx;
+		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (*idxp >= 0)
 			return nid;
-		}
 	}
 	return 0;
 }
@@ -835,6 +866,8 @@ static int build_digital_input(struct hda_codec *codec)
 
 /*
  * auto-mute and auto-mic switching
+ * CS421x auto-output redirecting
+ * HP/SPK/SPDIF
  */
 
 static void cs_automute(struct hda_codec *codec)
@@ -842,9 +875,25 @@ static void cs_automute(struct hda_codec *codec)
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	unsigned int hp_present;
+	unsigned int spdif_present;
 	hda_nid_t nid;
 	int i;
 
+	spdif_present = 0;
+	if (cfg->dig_outs) {
+		nid = cfg->dig_out_pins[0];
+		if (is_jack_detectable(codec, nid)) {
+			/*
+			TODO: SPDIF output redirect when SENSE_B is enabled.
+			Shared (SENSE_A) jack (e.g HP/mini-TOSLINK)
+			assumed.
+			*/
+			if (snd_hda_jack_detect(codec, nid)
+				/* && spec->sense_b */)
+				spdif_present = 1;
+		}
+	}
+
 	hp_present = 0;
 	for (i = 0; i < cfg->hp_outs; i++) {
 		nid = cfg->hp_pins[i];
@@ -854,11 +903,19 @@ static void cs_automute(struct hda_codec *codec)
 		if (hp_present)
 			break;
 	}
+
+	/* mute speakers if spdif or hp jack is plugged in */
 	for (i = 0; i < cfg->speaker_outs; i++) {
 		nid = cfg->speaker_pins[i];
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL,
 				    hp_present ? 0 : PIN_OUT);
+		/* detect on spdif is specific to CS421x */
+		if (spec->vendor_nid == CS421X_VENDOR_NID) {
+			snd_hda_codec_write(codec, nid, 0,
+					AC_VERB_SET_PIN_WIDGET_CONTROL,
+					spdif_present ? 0 : PIN_OUT);
+		}
 	}
 	if (spec->board_config == CS420X_MBP53 ||
 	    spec->board_config == CS420X_MBP55 ||
@@ -867,21 +924,62 @@ static void cs_automute(struct hda_codec *codec)
 		snd_hda_codec_write(codec, 0x01, 0,
 				    AC_VERB_SET_GPIO_DATA, gpio);
 	}
+
+	/* specific to CS421x */
+	if (spec->vendor_nid == CS421X_VENDOR_NID) {
+		/* mute HPs if spdif jack (SENSE_B) is present */
+		for (i = 0; i < cfg->hp_outs; i++) {
+			nid = cfg->hp_pins[i];
+			snd_hda_codec_write(codec, nid, 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL,
+				(spdif_present && spec->sense_b) ? 0 : PIN_HP);
+		}
+
+		/* SPDIF TX on/off */
+		if (cfg->dig_outs) {
+			nid = cfg->dig_out_pins[0];
+			snd_hda_codec_write(codec, nid, 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL,
+				spdif_present ? PIN_OUT : 0);
+
+		}
+		/* Update board GPIOs if neccessary ... */
+	}
 }
 
+/*
+ * Auto-input redirect for CS421x
+ * Switch max 3 inputs of a single ADC (nid 3)
+*/
+
 static void cs_automic(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	hda_nid_t nid;
 	unsigned int present;
-	
+
 	nid = cfg->inputs[spec->automic_idx].pin;
 	present = snd_hda_jack_detect(codec, nid);
-	if (present)
-		change_cur_input(codec, spec->automic_idx, 0);
-	else
-		change_cur_input(codec, !spec->automic_idx, 0);
+
+	/* specific to CS421x, single ADC */
+	if (spec->vendor_nid == CS421X_VENDOR_NID) {
+		if (present) {
+			spec->last_input = spec->cur_input;
+			spec->cur_input = spec->automic_idx;
+		} else  {
+			spec->cur_input = spec->last_input;
+		}
+
+		snd_hda_codec_write_cache(codec, spec->cur_adc, 0,
+					AC_VERB_SET_CONNECT_SEL,
+					spec->adc_idx[spec->cur_input]);
+	} else {
+		if (present)
+			change_cur_input(codec, spec->automic_idx, 0);
+		else
+			change_cur_input(codec, !spec->automic_idx, 0);
+	}
 }
 
 /*
@@ -911,23 +1009,28 @@ static void init_output(struct hda_codec *codec)
 	for (i = 0; i < cfg->line_outs; i++)
 		snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	/* HP */
 	for (i = 0; i < cfg->hp_outs; i++) {
 		hda_nid_t nid = cfg->hp_pins[i];
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
 		if (!cfg->speaker_outs)
 			continue;
-		if (is_jack_detectable(codec, nid)) {
+		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_UNSOLICITED_ENABLE,
 					    AC_USRSP_EN | HP_EVENT);
 			spec->hp_detect = 1;
 		}
 	}
+
+	/* Speaker */
 	for (i = 0; i < cfg->speaker_outs; i++)
 		snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-	if (spec->hp_detect)
+
+	/* SPDIF is enabled on presence detect for CS421x */
+	if (spec->hp_detect || spec->spdif_detect)
 		cs_automute(codec);
 }
 
@@ -961,19 +1064,31 @@ static void init_input(struct hda_codec *codec)
 					    AC_VERB_SET_UNSOLICITED_ENABLE,
 					    AC_USRSP_EN | MIC_EVENT);
 	}
-	change_cur_input(codec, spec->cur_input, 1);
-	if (spec->mic_detect)
-		cs_automic(codec);
-
-	coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
-	if (is_active_pin(codec, CS_DMIC2_PIN_NID))
-		coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
-	if (is_active_pin(codec, CS_DMIC1_PIN_NID))
-		coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 
-				 * No effect if SPDIF_OUT2 is selected in 
-				 * IDX_SPDIF_CTL.
-				  */
-	cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+	/* specific to CS421x */
+	if (spec->vendor_nid == CS421X_VENDOR_NID) {
+		if (spec->mic_detect)
+			cs_automic(codec);
+		else  {
+			spec->cur_adc = spec->adc_nid[spec->cur_input];
+			snd_hda_codec_write(codec, spec->cur_adc, 0,
+					AC_VERB_SET_CONNECT_SEL,
+					spec->adc_idx[spec->cur_input]);
+		}
+	} else {
+		change_cur_input(codec, spec->cur_input, 1);
+		if (spec->mic_detect)
+			cs_automic(codec);
+
+		coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
+		if (is_active_pin(codec, CS_DMIC2_PIN_NID))
+			coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */
+		if (is_active_pin(codec, CS_DMIC1_PIN_NID))
+			coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off
+					 * No effect if SPDIF_OUT2 is
+					 * selected in IDX_SPDIF_CTL.
+					*/
+		cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+	}
 }
 
 static const struct hda_verb cs_coef_init_verbs[] = {
@@ -1221,16 +1336,16 @@ static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
 	[CS420X_IMAC27] = imac27_pincfgs,
 };
 
-static void fix_pincfg(struct hda_codec *codec, int model)
+static void fix_pincfg(struct hda_codec *codec, int model,
+		       const struct cs_pincfg **pin_configs)
 {
-	const struct cs_pincfg *cfg = cs_pincfgs[model];
+	const struct cs_pincfg *cfg = pin_configs[model];
 	if (!cfg)
 		return;
 	for (; cfg->nid; cfg++)
 		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 }
 
-
 static int patch_cs420x(struct hda_codec *codec)
 {
 	struct cs_spec *spec;
@@ -1241,11 +1356,13 @@ static int patch_cs420x(struct hda_codec *codec)
 		return -ENOMEM;
 	codec->spec = spec;
 
+	spec->vendor_nid = CS420X_VENDOR_NID;
+
 	spec->board_config =
 		snd_hda_check_board_config(codec, CS420X_MODELS,
 					   cs420x_models, cs420x_cfg_tbl);
 	if (spec->board_config >= 0)
-		fix_pincfg(codec, spec->board_config);
+		fix_pincfg(codec, spec->board_config, cs_pincfgs);
 
 	switch (spec->board_config) {
 	case CS420X_IMAC27:
@@ -1272,6 +1389,562 @@ static int patch_cs420x(struct hda_codec *codec)
 	return err;
 }
 
+/*
+ * Cirrus Logic CS4210
+ *
+ * 1 DAC => HP(sense) / Speakers,
+ * 1 ADC <= LineIn(sense) / MicIn / DMicIn,
+ * 1 SPDIF OUT => SPDIF Trasmitter(sense)
+*/
+
+/* CS4210 board names */
+static const char *cs421x_models[CS421X_MODELS] = {
+	[CS421X_CDB4210] = "cdb4210",
+};
+
+static const struct snd_pci_quirk cs421x_cfg_tbl[] = {
+	/* Test Intel board + CDB2410  */
+	SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210),
+	{} /* terminator */
+};
+
+/* CS4210 board pinconfigs */
+/* Default CS4210 (CDB4210)*/
+static const struct cs_pincfg cdb4210_pincfgs[] = {
+	{ 0x05, 0x0321401f },
+	{ 0x06, 0x90170010 },
+	{ 0x07, 0x03813031 },
+	{ 0x08, 0xb7a70037 },
+	{ 0x09, 0xb7a6003e },
+	{ 0x0a, 0x034510f0 },
+	{} /* terminator */
+};
+
+static const struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = {
+	[CS421X_CDB4210] = cdb4210_pincfgs,
+};
+
+static const struct hda_verb cs421x_coef_init_verbs[] = {
+	{0x0B, AC_VERB_SET_PROC_STATE, 1},
+	{0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG},
+	/*
+	    Disable Coefficient Index Auto-Increment(DAI)=1,
+	    PDREF=0
+	*/
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x0001 },
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG},
+	/* ADC SZCMode = Digital Soft Ramp */
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x0002 },
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG},
+	{0x0B, AC_VERB_SET_PROC_COEF,
+	 (0x0002 /* DAC SZCMode = Digital Soft Ramp */
+	  | 0x0004 /* Mute DAC on FIFO error */
+	  | 0x0008 /* Enable DAC High Pass Filter */
+	  )},
+	{} /* terminator */
+};
+
+/* Errata: CS4210 rev A1 Silicon
+ *
+ * http://www.cirrus.com/en/pubs/errata/
+ *
+ * Description:
+ * 1. Performance degredation is present in the ADC.
+ * 2. Speaker output is not completely muted upon HP detect.
+ * 3. Noise is present when clipping occurs on the amplified
+ *    speaker outputs.
+ *
+ * Workaround:
+ * The following verb sequence written to the registers during
+ * initialization will correct the issues listed above.
+ */
+
+static const struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = {
+	{0x0B, AC_VERB_SET_PROC_STATE, 0x01},  /* VPW: processing on */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x0006},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x000A},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x0011},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x001A},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x001B},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */
+
+	{} /* terminator */
+};
+
+/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */
+static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0);
+
+static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 3;
+	return 0;
+}
+
+static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003;
+	return 0;
+}
+
+static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	unsigned int vol = ucontrol->value.integer.value[0];
+	unsigned int coef =
+		cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL);
+	unsigned int original_coef = coef;
+
+	coef &= ~0x0003;
+	coef |= (vol & 0x0003);
+	if (original_coef == coef)
+		return 0;
+	else {
+		cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef);
+		return 1;
+	}
+}
+
+static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = {
+
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.name = "Speaker Boost Playback Volume",
+	.info = cs421x_boost_vol_info,
+	.get = cs421x_boost_vol_get,
+	.put = cs421x_boost_vol_put,
+	.tlv = { .p = cs421x_speaker_boost_db_scale },
+};
+
+static void cs421x_pinmux_init(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	unsigned int def_conf, coef;
+
+	/* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */
+	coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG);
+
+	if (spec->gpio_mask)
+		coef |= 0x0008; /* B1,B2 are GPIOs */
+	else
+		coef &= ~0x0008;
+
+	if (spec->sense_b)
+		coef |= 0x0010; /* B2 is SENSE_B, not inverted  */
+	else
+		coef &= ~0x0010;
+
+	cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef);
+
+	if ((spec->gpio_mask || spec->sense_b) &&
+	    is_active_pin(codec, CS421X_DMIC_PIN_NID)) {
+
+		/*
+		    GPIO or SENSE_B forced - disconnect the DMIC pin.
+		*/
+		def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID);
+		def_conf &= ~AC_DEFCFG_PORT_CONN;
+		def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT);
+		snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf);
+	}
+}
+
+static void init_cs421x_digital(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+
+	for (i = 0; i < cfg->dig_outs; i++) {
+		hda_nid_t nid = cfg->dig_out_pins[i];
+		if (!cfg->speaker_outs)
+			continue;
+		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
+
+			snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_UNSOLICITED_ENABLE,
+				    AC_USRSP_EN | SPDIF_EVENT);
+			spec->spdif_detect = 1;
+		}
+	}
+}
+
+static int cs421x_init(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+
+	snd_hda_sequence_write(codec, cs421x_coef_init_verbs);
+	snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes);
+
+	cs421x_pinmux_init(codec);
+
+	if (spec->gpio_mask) {
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK,
+				    spec->gpio_mask);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION,
+				    spec->gpio_dir);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+				    spec->gpio_data);
+	}
+
+	init_output(codec);
+	init_input(codec);
+	init_cs421x_digital(codec);
+
+	return 0;
+}
+
+/*
+ * CS4210 Input MUX (1 ADC)
+ */
+static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+
+	return snd_hda_input_mux_info(&spec->input_mux, uinfo);
+}
+
+static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_input;
+	return 0;
+}
+
+static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+
+	return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol,
+				spec->adc_nid[0], &spec->cur_input);
+
+}
+
+static struct snd_kcontrol_new cs421x_capture_source = {
+
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Capture Source",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = cs421x_mux_enum_info,
+	.get = cs421x_mux_enum_get,
+	.put = cs421x_mux_enum_put,
+};
+
+static int cs421x_add_input_volume_control(struct hda_codec *codec, int item)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	const struct hda_input_mux *imux = &spec->input_mux;
+	hda_nid_t pin = cfg->inputs[item].pin;
+	struct snd_kcontrol *kctl;
+	u32 caps;
+
+	if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP))
+		return 0;
+
+	caps = query_amp_caps(codec, pin, HDA_INPUT);
+	caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+	if (caps <= 1)
+		return 0;
+
+	return add_volume(codec,  imux->items[item].label, 0,
+			  HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl);
+}
+
+/* add a (input-boost) volume control to the given input pin */
+static int build_cs421x_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct hda_input_mux *imux = &spec->input_mux;
+	int i, err, type_idx;
+	const char *label;
+
+	if (!spec->num_inputs)
+		return 0;
+
+	/* make bind-capture */
+	spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw);
+	spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
+	for (i = 0; i < 2; i++) {
+		struct snd_kcontrol *kctl;
+		int n;
+		if (!spec->capture_bind[i])
+			return -ENOMEM;
+		kctl = snd_ctl_new1(&cs_capture_ctls[i], codec);
+		if (!kctl)
+			return -ENOMEM;
+		kctl->private_value = (long)spec->capture_bind[i];
+		err = snd_hda_ctl_add(codec, 0, kctl);
+		if (err < 0)
+			return err;
+		for (n = 0; n < AUTO_PIN_LAST; n++) {
+			if (!spec->adc_nid[n])
+				continue;
+			err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	/* Add Input MUX Items + Capture Volume/Switch */
+	for (i = 0; i < spec->num_inputs; i++) {
+		label = hda_get_autocfg_input_label(codec, cfg, i);
+		snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx);
+
+		err = cs421x_add_input_volume_control(codec, i);
+		if (err < 0)
+			return err;
+	}
+
+	/*
+	    Add 'Capture Source' Switch if
+		* 2 inputs and no mic detec
+		* 3 inputs
+	*/
+	if ((spec->num_inputs == 2 && !spec->mic_detect) ||
+	    (spec->num_inputs == 3)) {
+
+		err = snd_hda_ctl_add(codec, spec->adc_nid[0],
+			      snd_ctl_new1(&cs421x_capture_source, codec));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Single DAC (Mute/Gain) */
+static int build_cs421x_output(struct hda_codec *codec)
+{
+	hda_nid_t dac = CS4210_DAC_NID;
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct snd_kcontrol *kctl;
+	int err;
+	char *name = "HP/Speakers";
+
+	fix_volume_caps(codec, dac);
+	if (!spec->vmaster_sw) {
+		err = add_vmaster(codec, dac);
+		if (err < 0)
+			return err;
+	}
+
+	err = add_mute(codec, name, 0,
+			HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_slave(spec->vmaster_sw, kctl);
+	if (err < 0)
+		return err;
+
+	err = add_volume(codec, name, 0,
+			HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_slave(spec->vmaster_vol, kctl);
+	if (err < 0)
+		return err;
+
+	if (cfg->speaker_outs) {
+		err = snd_hda_ctl_add(codec, 0,
+			snd_ctl_new1(&cs421x_speaker_bost_ctl, codec));
+		if (err < 0)
+			return err;
+	}
+	return err;
+}
+
+static int cs421x_build_controls(struct hda_codec *codec)
+{
+	int err;
+
+	err = build_cs421x_output(codec);
+	if (err < 0)
+		return err;
+	err = build_cs421x_input(codec);
+	if (err < 0)
+		return err;
+	err = build_digital_output(codec);
+	if (err < 0)
+		return err;
+	return cs421x_init(codec);
+}
+
+static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	switch ((res >> 26) & 0x3f) {
+	case HP_EVENT:
+	case SPDIF_EVENT:
+		cs_automute(codec);
+		break;
+
+	case MIC_EVENT:
+		cs_automic(codec);
+		break;
+	}
+}
+
+static int parse_cs421x_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		hda_nid_t pin = cfg->inputs[i].pin;
+		spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
+		spec->cur_input = spec->last_input = i;
+		spec->num_inputs++;
+
+		/* check whether the automatic mic switch is available */
+		if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) {
+			spec->mic_detect = 1;
+			spec->automic_idx = i;
+		}
+	}
+	return 0;
+}
+
+static int cs421x_parse_auto_config(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+	if (err < 0)
+		return err;
+	err = parse_output(codec);
+	if (err < 0)
+		return err;
+	err = parse_cs421x_input(codec);
+	if (err < 0)
+		return err;
+	err = parse_digital_output(codec);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+	Manage PDREF, when transitioning to D3hot
+	(DAC,ADC) -> D3, PDREF=1, AFG->D3
+*/
+static int cs421x_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	unsigned int coef;
+
+	snd_hda_shutup_pins(codec);
+
+	snd_hda_codec_write(codec, CS4210_DAC_NID, 0,
+			    AC_VERB_SET_POWER_STATE,  AC_PWRST_D3);
+	snd_hda_codec_write(codec, CS4210_ADC_NID, 0,
+			    AC_VERB_SET_POWER_STATE,  AC_PWRST_D3);
+
+	coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG);
+	coef |= 0x0004; /* PDREF */
+	cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef);
+
+	return 0;
+}
+#endif
+
+static struct hda_codec_ops cs4210_patch_ops = {
+	.build_controls = cs421x_build_controls,
+	.build_pcms = cs_build_pcms,
+	.init = cs421x_init,
+	.free = cs_free,
+	.unsol_event = cs421x_unsol_event,
+#ifdef CONFIG_PM
+	.suspend = cs421x_suspend,
+#endif
+};
+
+static int patch_cs421x(struct hda_codec *codec)
+{
+	struct cs_spec *spec;
+	int err;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+
+	spec->vendor_nid = CS421X_VENDOR_NID;
+
+	spec->board_config =
+		snd_hda_check_board_config(codec, CS421X_MODELS,
+					   cs421x_models, cs421x_cfg_tbl);
+	if (spec->board_config >= 0)
+		fix_pincfg(codec, spec->board_config, cs421x_pincfgs);
+	/*
+	    Setup GPIO/SENSE for each board (if used)
+	*/
+	switch (spec->board_config) {
+	case CS421X_CDB4210:
+		snd_printd("CS4210 board: %s\n",
+			cs421x_models[spec->board_config]);
+/*		spec->gpio_mask = 3;
+		spec->gpio_dir = 3;
+		spec->gpio_data = 3;
+*/
+		spec->sense_b = 1;
+
+		break;
+	}
+
+	/*
+	    Update the GPIO/DMIC/SENSE_B pinmux before the configuration
+	    is auto-parsed. If GPIO or SENSE_B is forced, DMIC input
+	    is disabled.
+	*/
+	cs421x_pinmux_init(codec);
+
+	err = cs421x_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
+
+	codec->patch_ops = cs4210_patch_ops;
+
+	return 0;
+
+ error:
+	kfree(codec->spec);
+	codec->spec = NULL;
+	return err;
+}
+
 
 /*
  * patch entries
@@ -1279,11 +1952,13 @@ static int patch_cs420x(struct hda_codec *codec)
 static const struct hda_codec_preset snd_hda_preset_cirrus[] = {
 	{ .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
 	{ .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
+	{ .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x },
 	{} /* terminator */
 };
 
 MODULE_ALIAS("snd-hda-codec-id:10134206");
 MODULE_ALIAS("snd-hda-codec-id:10134207");
+MODULE_ALIAS("snd-hda-codec-id:10134210");
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
-- 
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