Merge branch 'fix/hda' into topic/hda

Necessary for working on the jack-detection suppression feature.
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c2c65f6..65c0179 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2300,7 +2300,7 @@
 
 /* apply the function to all matching slave ctls in the mixer list */
 static int map_slaves(struct hda_codec *codec, const char * const *slaves,
-		      map_slave_func_t func, void *data) 
+		      const char *suffix, map_slave_func_t func, void *data) 
 {
 	struct hda_nid_item *items;
 	const char * const *s;
@@ -2313,7 +2313,14 @@
 		    sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
 			continue;
 		for (s = slaves; *s; s++) {
-			if (!strcmp(sctl->id.name, *s)) {
+			char tmpname[sizeof(sctl->id.name)];
+			const char *name = *s;
+			if (suffix) {
+				snprintf(tmpname, sizeof(tmpname), "%s %s",
+					 name, suffix);
+				name = tmpname;
+			}
+			if (!strcmp(sctl->id.name, name)) {
 				err = func(data, sctl);
 				if (err)
 					return err;
@@ -2335,6 +2342,7 @@
  * @name: vmaster control name
  * @tlv: TLV data (optional)
  * @slaves: slave control names (optional)
+ * @suffix: suffix string to each slave name (optional)
  *
  * Create a virtual master control with the given name.  The TLV data
  * must be either NULL or a valid data.
@@ -2346,12 +2354,13 @@
  * This function returns zero if successful or a negative error code.
  */
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
-			unsigned int *tlv, const char * const *slaves)
+			unsigned int *tlv, const char * const *slaves,
+			const char *suffix)
 {
 	struct snd_kcontrol *kctl;
 	int err;
 
-	err = map_slaves(codec, slaves, check_slave_present, NULL);
+	err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
 	if (err != 1) {
 		snd_printdd("No slave found for %s\n", name);
 		return 0;
@@ -2363,8 +2372,8 @@
 	if (err < 0)
 		return err;
 
-	err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave,
-			 kctl);
+	err = map_slaves(codec, slaves, suffix,
+			 (map_slave_func_t)snd_ctl_add_slave, kctl);
 	if (err < 0)
 		return err;
 	return 0;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 95dfb68..e354c16 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -94,7 +94,7 @@
 module_param(single_cmd, bool, 0444);
 MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
 		 "(for debugging only).");
-module_param(enable_msi, int, 0444);
+module_param(enable_msi, bint, 0444);
 MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
 module_param_array(patch, charp, NULL, 0444);
@@ -121,8 +121,8 @@
 MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
 #endif
 
-static bool align_buffer_size = 1;
-module_param(align_buffer_size, bool, 0644);
+static int align_buffer_size = -1;
+module_param(align_buffer_size, bint, 0644);
 MODULE_PARM_DESC(align_buffer_size,
 		"Force buffer and period sizes to be multiple of 128 bytes.");
 
@@ -148,6 +148,7 @@
 			 "{Intel, PCH},"
 			 "{Intel, CPT},"
 			 "{Intel, PPT},"
+			 "{Intel, LPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
 			 "{ATI, SB450},"
@@ -515,6 +516,7 @@
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
 #define AZX_DCAPS_BUFSIZE	(1 << 21)	/* no buffer size alignment */
+#define AZX_DCAPS_ALIGN_BUFSIZE	(1 << 22)	/* buffer size alignment */
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -527,7 +529,8 @@
 
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
-	(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI)
+	(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
+	 AZX_DCAPS_ALIGN_BUFSIZE)
 
 static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
@@ -2774,9 +2777,16 @@
 	}
 
 	/* disable buffer size rounding to 128-byte multiples if supported */
-	chip->align_buffer_size = align_buffer_size;
-	if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
-		chip->align_buffer_size = 0;
+	if (align_buffer_size >= 0)
+		chip->align_buffer_size = !!align_buffer_size;
+	else {
+		if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
+			chip->align_buffer_size = 0;
+		else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE)
+			chip->align_buffer_size = 1;
+		else
+			chip->align_buffer_size = 1;
+	}
 
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
@@ -2992,6 +3002,10 @@
 	{ PCI_DEVICE(0x8086, 0x1e20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
 	  AZX_DCAPS_BUFSIZE},
+	/* Lynx Point */
+	{ PCI_DEVICE(0x8086, 0x8c20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE},
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index aca8d31..6094dea8 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -140,7 +140,8 @@
 struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 					    const char *name);
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
-			unsigned int *tlv, const char * const *slaves);
+			unsigned int *tlv, const char * const *slaves,
+			const char *suffix);
 int snd_hda_codec_reset(struct hda_codec *codec);
 
 /* amp value bits */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 9cb14b4..9771b07 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -137,51 +137,17 @@
 	return 0;
 }
 
-static const char * const ad_slave_vols[] = {
-	"Front Playback Volume",
-	"Surround Playback Volume",
-	"Center Playback Volume",
-	"LFE Playback Volume",
-	"Side Playback Volume",
-	"Headphone Playback Volume",
-	"Mono Playback Volume",
-	"Speaker Playback Volume",
-	"IEC958 Playback Volume",
+static const char * const ad_slave_pfxs[] = {
+	"Front", "Surround", "Center", "LFE", "Side",
+	"Headphone", "Mono", "Speaker", "IEC958",
 	NULL
 };
 
-static const char * const ad_slave_sws[] = {
-	"Front Playback Switch",
-	"Surround Playback Switch",
-	"Center Playback Switch",
-	"LFE Playback Switch",
-	"Side Playback Switch",
-	"Headphone Playback Switch",
-	"Mono Playback Switch",
-	"Speaker Playback Switch",
-	"IEC958 Playback Switch",
+static const char * const ad1988_6stack_fp_slave_pfxs[] = {
+	"Front", "Surround", "Center", "LFE", "Side", "IEC958",
 	NULL
 };
 
-static const char * const ad1988_6stack_fp_slave_vols[] = {
-	"Front Playback Volume",
-	"Surround Playback Volume",
-	"Center Playback Volume",
-	"LFE Playback Volume",
-	"Side Playback Volume",
-	"IEC958 Playback Volume",
-	NULL
-};
-
-static const char * const ad1988_6stack_fp_slave_sws[] = {
-	"Front Playback Switch",
-	"Surround Playback Switch",
-	"Center Playback Switch",
-	"LFE Playback Switch",
-	"Side Playback Switch",
-	"IEC958 Playback Switch",
-	NULL
-};
 static void ad198x_free_kctls(struct hda_codec *codec);
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -260,7 +226,8 @@
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
 					  vmaster_tlv,
 					  (spec->slave_vols ?
-					   spec->slave_vols : ad_slave_vols));
+					   spec->slave_vols : ad_slave_pfxs),
+					  "Playback Volume");
 		if (err < 0)
 			return err;
 	}
@@ -268,7 +235,8 @@
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
 					  NULL,
 					  (spec->slave_sws ?
-					   spec->slave_sws : ad_slave_sws));
+					   spec->slave_sws : ad_slave_pfxs),
+					  "Playback Switch");
 		if (err < 0)
 			return err;
 	}
@@ -3385,8 +3353,8 @@
 
 	if (spec->autocfg.hp_pins[0]) {
 		spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
-		spec->slave_vols = ad1988_6stack_fp_slave_vols;
-		spec->slave_sws = ad1988_6stack_fp_slave_sws;
+		spec->slave_vols = ad1988_6stack_fp_slave_pfxs;
+		spec->slave_sws = ad1988_6stack_fp_slave_pfxs;
 		spec->alt_dac_nid = ad1988_alt_dac_nid;
 		spec->stream_analog_alt_playback =
 			&ad198x_pcm_analog_alt_playback;
@@ -3594,16 +3562,8 @@
 #endif
 
 static const char * const ad1884_slave_vols[] = {
-	"PCM Playback Volume",
-	"Mic Playback Volume",
-	"Mono Playback Volume",
-	"Front Mic Playback Volume",
-	"Mic Playback Volume",
-	"CD Playback Volume",
-	"Internal Mic Playback Volume",
-	"Docking Mic Playback Volume",
-	/* "Beep Playback Volume", */
-	"IEC958 Playback Volume",
+	"PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
+	"Internal Mic", "Docking Mic", /* "Beep", */ "IEC958",
 	NULL
 };
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a7a5733..266e5a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -465,21 +465,8 @@
 };
 #endif
 
-static const char * const slave_vols[] = {
-	"Headphone Playback Volume",
-	"Speaker Playback Volume",
-	"Front Playback Volume",
-	"Surround Playback Volume",
-	"CLFE Playback Volume",
-	NULL
-};
-
-static const char * const slave_sws[] = {
-	"Headphone Playback Switch",
-	"Speaker Playback Switch",
-	"Front Playback Switch",
-	"Surround Playback Switch",
-	"CLFE Playback Switch",
+static const char * const slave_pfxs[] = {
+	"Headphone", "Speaker", "Front", "Surround", "CLFE",
 	NULL
 };
 
@@ -519,14 +506,16 @@
 		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
 					HDA_OUTPUT, vmaster_tlv);
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv, slave_vols);
+					  vmaster_tlv, slave_pfxs,
+					  "Playback Volume");
 		if (err < 0)
 			return err;
 	}
 	if (spec->vmaster_nid &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL, slave_sws);
+					  NULL, slave_pfxs,
+					  "Playback Switch");
 		if (err < 0)
 			return err;
 	}
@@ -3034,7 +3023,6 @@
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
 	SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
 	{}
 };
@@ -4414,6 +4402,18 @@
 	codec->patch_ops = cx_auto_patch_ops;
 	if (spec->beep_amp)
 		snd_hda_attach_beep_device(codec, spec->beep_amp);
+
+	/* Some laptops with Conexant chips show stalls in S3 resume,
+	 * which falls into the single-cmd mode.
+	 * Better to make reset, then.
+	 */
+	if (!codec->bus->sync_write) {
+		snd_printd("hda_codec: "
+			   "Enable sync_write for stable communication\n");
+		codec->bus->sync_write = 1;
+		codec->bus->allow_bus_reset = 1;
+	}
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9350f3c..4a2a49f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1845,36 +1845,10 @@
 /*
  * slave controls for virtual master
  */
-static const char * const alc_slave_vols[] = {
-	"Front Playback Volume",
-	"Surround Playback Volume",
-	"Center Playback Volume",
-	"LFE Playback Volume",
-	"Side Playback Volume",
-	"Headphone Playback Volume",
-	"Speaker Playback Volume",
-	"Mono Playback Volume",
-	"Line-Out Playback Volume",
-	"CLFE Playback Volume",
-	"Bass Speaker Playback Volume",
-	"PCM Playback Volume",
-	NULL,
-};
-
-static const char * const alc_slave_sws[] = {
-	"Front Playback Switch",
-	"Surround Playback Switch",
-	"Center Playback Switch",
-	"LFE Playback Switch",
-	"Side Playback Switch",
-	"Headphone Playback Switch",
-	"Speaker Playback Switch",
-	"Mono Playback Switch",
-	"IEC958 Playback Switch",
-	"Line-Out Playback Switch",
-	"CLFE Playback Switch",
-	"Bass Speaker Playback Switch",
-	"PCM Playback Switch",
+static const char * const alc_slave_pfxs[] = {
+	"Front", "Surround", "Center", "LFE", "Side",
+	"Headphone", "Speaker", "Mono", "Line-Out",
+	"CLFE", "Bass Speaker", "PCM",
 	NULL,
 };
 
@@ -1965,14 +1939,16 @@
 		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
 					HDA_OUTPUT, vmaster_tlv);
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv, alc_slave_vols);
+					  vmaster_tlv, alc_slave_pfxs,
+					  "Playback Volume");
 		if (err < 0)
 			return err;
 	}
 	if (!spec->no_analog &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL, alc_slave_sws);
+					  NULL, alc_slave_pfxs,
+					  "Playback Switch");
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df1..4c76940 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1060,26 +1060,9 @@
 	.put = stac92xx_smux_enum_put,
 };
 
-static const char * const slave_vols[] = {
-	"Front Playback Volume",
-	"Surround Playback Volume",
-	"Center Playback Volume",
-	"LFE Playback Volume",
-	"Side Playback Volume",
-	"Headphone Playback Volume",
-	"Speaker Playback Volume",
-	NULL
-};
-
-static const char * const slave_sws[] = {
-	"Front Playback Switch",
-	"Surround Playback Switch",
-	"Center Playback Switch",
-	"LFE Playback Switch",
-	"Side Playback Switch",
-	"Headphone Playback Switch",
-	"Speaker Playback Switch",
-	"IEC958 Playback Switch",
+static const char * const slave_pfxs[] = {
+	"Front", "Surround", "Center", "LFE", "Side",
+	"Headphone", "Speaker", "IEC958",
 	NULL
 };
 
@@ -1153,13 +1136,15 @@
 		/* minimum value is actually mute */
 		vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv, slave_vols);
+					  vmaster_tlv, slave_pfxs,
+					  "Playback Volume");
 		if (err < 0)
 			return err;
 	}
 	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL, slave_sws);
+					  NULL, slave_pfxs,
+					  "Playback Switch");
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 284e311..e5842fe 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1442,25 +1442,9 @@
 /*
  * slave controls for virtual master
  */
-static const char * const via_slave_vols[] = {
-	"Front Playback Volume",
-	"Surround Playback Volume",
-	"Center Playback Volume",
-	"LFE Playback Volume",
-	"Side Playback Volume",
-	"Headphone Playback Volume",
-	"Speaker Playback Volume",
-	NULL,
-};
-
-static const char * const via_slave_sws[] = {
-	"Front Playback Switch",
-	"Surround Playback Switch",
-	"Center Playback Switch",
-	"LFE Playback Switch",
-	"Side Playback Switch",
-	"Headphone Playback Switch",
-	"Speaker Playback Switch",
+static const char * const via_slave_pfxs[] = {
+	"Front", "Surround", "Center", "LFE", "Side",
+	"Headphone", "Speaker",
 	NULL,
 };
 
@@ -1505,13 +1489,15 @@
 		snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
 					HDA_OUTPUT, vmaster_tlv);
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv, via_slave_vols);
+					  vmaster_tlv, via_slave_pfxs,
+					  "Playback Volume");
 		if (err < 0)
 			return err;
 	}
 	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL, via_slave_sws);
+					  NULL, via_slave_pfxs,
+					  "Playback Switch");
 		if (err < 0)
 			return err;
 	}