Merge tag 'v3.3' into staging/for_v3.4

* tag 'v3.3': (1646 commits)
  Linux 3.3
  Don't limit non-nested epoll paths
  netfilter: ctnetlink: fix race between delete and timeout expiration
  ipv6: Don't dev_hold(dev) in ip6_mc_find_dev_rcu.
  nilfs2: fix NULL pointer dereference in nilfs_load_super_block()
  nilfs2: clamp ns_r_segments_percentage to [1, 99]
  afs: Remote abort can cause BUG in rxrpc code
  afs: Read of file returns EBADMSG
  C6X: remove dead code from entry.S
  wimax/i2400m: fix erroneous NETDEV_TX_BUSY use
  net/hyperv: fix erroneous NETDEV_TX_BUSY use
  net/usbnet: reserve headroom on rx skbs
  bnx2x: fix memory leak in bnx2x_init_firmware()
  bnx2x: fix a crash on corrupt firmware file
  sch_sfq: revert dont put new flow at the end of flows
  ipv6: fix icmp6_dst_alloc()
  MAINTAINERS: Add Serge as maintainer of capabilities
  drivers/video/backlight/s6e63m0.c: fix corruption storing gamma mode
  MAINTAINERS: add entry for exynos mipi display drivers
  MAINTAINERS: fix link to Gustavo Padovans tree
  ...

43 files changed, 638 insertions, 482 deletions

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index dac3633507c9..a68aed7fce02 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 		params = kmalloc(sizeof(*params), GFP_KERNEL);
 		if (!params)
 			return -ENOMEM;
-		if (copy_from_user(params, (void __user *)arg, sizeof(*params)))
-			return -EFAULT;
+		if (copy_from_user(params, (void __user *)arg, sizeof(*params))) {
+			retval = -EFAULT;
+			goto out;
+		}
 		retval = snd_compr_allocate_buffer(stream, params);
 		if (retval) {
-			kfree(params);
-			return -ENOMEM;
+			retval = -ENOMEM;
+			goto out;
 		}
 		retval = stream->ops->set_params(stream, params);
 		if (retval)
 			goto out;
 		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
-	} else
+	} else {
 		return -EPERM;
+	}
 out:
 	kfree(params);
 	return retval;
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index e09f144177f5..c99c6078be33 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -22,7 +22,6 @@
 #include "emu8000_local.h"
 #include <asm/uaccess.h>
 #include <linux/moduleparam.h>
-#include <linux/moduleparam.h>
 
 static int emu8000_reset_addr;
 module_param(emu8000_reset_addr, int, 0444);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6e..496f14c1a731 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
 		if (err < 0)
 			goto out_err;
+		opl3->private_data = chip;
 	}
 
-	opl3->private_data = chip;
-
 	sprintf(card->longname, "%s at 0x%lx, irq %i",
 		card->shortname, chip->ctrl_io, chip->irq);
 
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
index 5b68435d195b..501501ef36a9 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
 	/* Looks like the unsol event is incompatible with the standard
 	 * definition.  4bit tag is placed at 28 bit!
 	 */
-	switch (res >> 28) {
+	res >>= 28;
+	switch (res) {
 	case ALC_MIC_EVENT:
 		alc88x_simple_mic_automute(codec);
 		break;
 	default:
-		alc_sku_unsol_event(codec, res);
+		alc_exec_unsol_event(codec, res);
 		break;
 	}
 }
 
+static void alc880_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	alc_exec_unsol_event(codec, res >> 28);
+}
+
 static void alc880_uniwill_p53_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
 	/* Looks like the unsol event is incompatible with the standard
 	 * definition.  4bit tag is placed at 28 bit!
 	 */
-	if ((res >> 28) == ALC_DCVOL_EVENT)
+	res >>= 28;
+	if (res == ALC_DCVOL_EVENT)
 		alc880_uniwill_p53_dcvol_automute(codec);
 	else
-		alc_sku_unsol_event(codec, res);
+		alc_exec_unsol_event(codec, res);
 }
 
 /*
@@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_lg_ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc880_lg_capture_source,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc880_unsol_event,
 		.setup = alc880_lg_setup,
 		.init_hook = alc_hp_automute,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
index bdf0ed4ab3e2..bb364a53f546 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
 		alc889A_mb31_automute(codec);
 }
 
+static void alc882_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	alc_exec_unsol_event(codec, res >> 26);
+}
+
 /*
  * configuration and preset
  */
@@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = {
 			.channel_mode = alc885_mba21_ch_modes,
 			.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
 			.input_mux = &alc882_capture_source,
-			.unsol_event = alc_sku_unsol_event,
+			.unsol_event = alc882_unsol_event,
 			.setup = alc885_mba21_setup,
 			.init_hook = alc_hp_automute,
        },
@@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc882_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_mbp3_setup,
 		.init_hook = alc_hp_automute,
 	},
@@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_mb5_setup,
 		.init_hook = alc_hp_automute,
 	},
@@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &macmini3_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_macmini3_setup,
 		.init_hook = alc_hp_automute,
 	},
@@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc889A_imac91_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_imac91_setup,
 		.init_hook = alc_hp_automute,
 	},
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4df72c0e8c37..684307372d73 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 		for (i = 0; i < c->cvt_setups.used; i++) {
 			p = snd_array_elem(&c->cvt_setups, i);
 			if (!p->active && p->stream_tag == stream_tag &&
-			    get_wcaps_type(get_wcaps(codec, p->nid)) == type)
+			    get_wcaps_type(get_wcaps(c, p->nid)) == type)
 				p->dirty = 1;
 		}
 	}
@@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
 	parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
 	parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
 	parm |= index << AC_AMP_SET_INDEX_SHIFT;
-	parm |= val;
+	if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+	    (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+		; /* set the zero value as a fake mute */
+	else
+		parm |= val;
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
 	info->vol[ch] = val;
 }
@@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
 	val1 += ofs;
 	val1 = ((int)val1) * ((int)val2);
-	if (min_mute)
+	if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
 		val2 |= TLV_DB_SCALE_MUTE;
 	if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
 		return -EFAULT;
@@ -5114,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
 	const char *pfx = "", *sfx = "";
 
 	/* handle as a speaker if it's a fixed line-out */
-	if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT)
+	if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
 		name = "Speaker";
 	/* check the location */
 	switch (attr) {
@@ -5173,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 
 	switch (get_defcfg_device(def_conf)) {
 	case AC_JACK_LINE_OUT:
-		return fill_audio_out_name(codec, nid, cfg, "Line-Out",
+		return fill_audio_out_name(codec, nid, cfg, "Line Out",
 					   label, maxlen, indexp);
 	case AC_JACK_SPEAKER:
 		return fill_audio_out_name(codec, nid, cfg, "Speaker",
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index e9f71dc0d464..f0f1943a4b2c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -298,6 +298,9 @@ enum {
 #define AC_AMPCAP_MUTE			(1<<31)    /* mute capable */
 #define AC_AMPCAP_MUTE_SHIFT		31
 
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE		(1 << 30) /* min-volume = mute */
+
 /* Connection list */
 #define AC_CLIST_LENGTH			(0x7f<<0)
 #define AC_CLIST_LONG			(1<<7)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index fb35474c1203..95dfb6874941 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -469,6 +469,7 @@ struct azx {
 	unsigned int irq_pending_warned :1;
 	unsigned int probing :1; /* codec probing phase */
 	unsigned int snoop:1;
+	unsigned int align_buffer_size:1;
 
 	/* for debugging */
 	unsigned int last_cmd[AZX_MAX_CODECS];
@@ -1690,7 +1691,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	runtime->hw.rates = hinfo->rates;
 	snd_pcm_limit_hw_rates(runtime);
 	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
-	if (align_buffer_size)
+	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
 		   access but isn't required by the HDA spec and
@@ -2773,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	/* disable buffer size rounding to 128-byte multiples if supported */
+	chip->align_buffer_size = align_buffer_size;
 	if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
-		align_buffer_size = 0;
+		chip->align_buffer_size = 0;
 
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d8a35da0803f..9d819c4b4923 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
 
 static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
-			 const struct auto_pin_cfg *cfg)
+			 const struct auto_pin_cfg *cfg,
+			 char *lastname, int *lastidx)
 {
 	unsigned int def_conf, conn;
 	char name[44];
@@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
 		return 0;
 
 	snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
+	if (!strcmp(name, lastname) && idx == *lastidx)
+		idx++;
+	strncpy(lastname, name, 44);
+	*lastidx = idx;
 	err = snd_hda_jack_add_kctl(codec, nid, name, idx);
 	if (err < 0)
 		return err;
@@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
 			   const struct auto_pin_cfg *cfg)
 {
 	const hda_nid_t *p;
-	int i, err;
+	int i, err, lastidx = 0;
+	char lastname[44] = "";
 
 	for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
-		err = add_jack_kctl(codec, *p, cfg);
+		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
 		if (*p == *cfg->line_out_pins) /* might be duplicated */
 			break;
-		err = add_jack_kctl(codec, *p, cfg);
+		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
 		if (*p == *cfg->line_out_pins) /* might be duplicated */
 			break;
-		err = add_jack_kctl(codec, *p, cfg);
+		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0; i < cfg->num_inputs; i++) {
-		err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
+		err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
-		err = add_jack_kctl(codec, *p, cfg);
+		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
 		if (err < 0)
 			return err;
 	}
-	err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
+	err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx);
 	if (err < 0)
 		return err;
-	err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
+	err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx);
 	if (err < 0)
 		return err;
 	return 0;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 35abe3c62908..21d91d580da8 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol,
 
 	err = chipio_read(codec, REG_CODEC_MUTE, &data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	/* *valp 0 is mute, 1 is unmute */
 	data = (data & 0x7f) | (*valp ? 0 : 0x80);
-	chipio_write(codec, REG_CODEC_MUTE, data);
+	err = chipio_write(codec, REG_CODEC_MUTE, data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	spec->curr_hp_switch = *valp;
 
+ exit:
 	snd_hda_power_down(codec);
-	return 1;
+	return err < 0 ? err : 1;
 }
 
 static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol,
@@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol,
 
 	err = chipio_read(codec, REG_CODEC_MUTE, &data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	/* *valp 0 is mute, 1 is unmute */
 	data = (data & 0xef) | (*valp ? 0 : 0x10);
-	chipio_write(codec, REG_CODEC_MUTE, data);
+	err = chipio_write(codec, REG_CODEC_MUTE, data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	spec->curr_speaker_switch = *valp;
 
+ exit:
 	snd_hda_power_down(codec);
-	return 1;
+	return err < 0 ? err : 1;
 }
 
 static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol,
@@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol,
 
 	err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	val = 31 - left_vol;
 	data = (data & 0xe0) | val;
-	chipio_write(codec, REG_CODEC_HP_VOL_L, data);
+	err = chipio_write(codec, REG_CODEC_HP_VOL_L, data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	val = 31 - right_vol;
 	data = (data & 0xe0) | val;
-	chipio_write(codec, REG_CODEC_HP_VOL_R, data);
+	err = chipio_write(codec, REG_CODEC_HP_VOL_R, data);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	spec->curr_hp_volume[0] = left_vol;
 	spec->curr_hp_volume[1] = right_vol;
 
+ exit:
 	snd_hda_power_down(codec);
-	return 1;
+	return err < 0 ? err : 1;
 }
 
 static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid)
@@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		err = add_in_volume(codec, spec->dig_in, "IEC958");
+		if (err < 0)
+			return err;
 	}
 	return 0;
 }
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 0e99357e822c..c83ccdba1e5a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
 		"Front Speaker", "Surround Speaker", "Bass Speaker"
 	};
 	static const char * const line_outs[] = {
-		"Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+		"Front Line Out", "Surround Line Out", "Bass Line Out"
 	};
 
 	fix_volume_caps(codec, dac);
@@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
 		if (num_ctls > 1)
 			name = line_outs[idx];
 		else
-			name = "Line-Out";
+			name = "Line Out";
 		break;
 	}
 
@@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec)
 			change_cur_input(codec, !spec->automic_idx, 0);
 	} else {
 		if (present) {
-			spec->last_input = spec->cur_input;
-			spec->cur_input = spec->automic_idx;
+			if (spec->cur_input != spec->automic_idx) {
+				spec->last_input = spec->cur_input;
+				spec->cur_input = spec->automic_idx;
+			}
 		} else  {
 			spec->cur_input = spec->last_input;
 		}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8a32a69c83c3..d29d6d377904 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
@@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
 		"Disabled", "Enabled"
 	};
 	static const char * const texts3[] = {
-		"Disabled", "Speaker Only", "Line-Out+Speaker"
+		"Disabled", "Speaker Only", "Line Out+Speaker"
 	};
 	const char * const *texts;
 
@@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 		err = snd_hda_ctl_add(codec, nid, kctl);
 		if (err < 0)
 			return err;
-		if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+		if (!(query_amp_caps(codec, nid, hda_dir) &
+		      (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
 			break;
 	}
 	return 0;
@@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = {
 	{}
 };
 
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+	static hda_nid_t out_nids[] = {
+		0x10, 0x11, 0
+	};
+	hda_nid_t *p;
+
+	for (p = out_nids; *p; p++)
+		snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+					  AC_AMPCAP_MIN_MUTE |
+					  query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
 static int patch_conexant_auto(struct hda_codec *codec)
 {
 	struct conexant_spec *spec;
@@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	case 0x14f15045:
 		spec->single_adc_amp = 1;
 		break;
+	case 0x14f15051:
+		add_cx5051_fake_mutes(codec);
+		break;
 	}
 
 	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e82acf77c5a..22c73b78ac6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
 	ALC_AUTOMUTE_MIXER,	/* mute/unmute mixer widget AMP */
 };
 
+#define MAX_VOL_NIDS	0x40
+
 struct alc_spec {
 	/* codec parameterization */
 	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
 	const hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 	hda_nid_t mixer_nid;		/* analog-mixer NID */
-	DECLARE_BITMAP(vol_ctls, 0x20 << 1);
-	DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+	DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+	DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
 
 	/* capture setup for dynamic dual-adc switch */
 	hda_nid_t cur_adc;
@@ -177,6 +179,7 @@ struct alc_spec {
 	unsigned int detect_lo:1;	/* Line-out detection enabled */
 	unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */
 	unsigned int automute_lo_possible:1;	  /* there are line outs and HP */
+	unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */
 
 	/* other flags */
 	unsigned int no_analog :1; /* digital I/O only */
@@ -185,7 +188,6 @@ struct alc_spec {
 	unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
 	unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
 	unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
-	unsigned int use_jack_tbl:1; /* 1 for model=auto */
 
 	/* auto-mute control */
 	int automute_mode;
@@ -496,13 +498,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 
 	for (i = 0; i < num_pins; i++) {
 		hda_nid_t nid = pins[i];
+		unsigned int val;
 		if (!nid)
 			break;
 		switch (spec->automute_mode) {
 		case ALC_AUTOMUTE_PIN:
+			/* don't reset VREF value in case it's controlling
+			 * the amp (see alc861_fixup_asus_amp_vref_0f())
+			 */
+			if (spec->keep_vref_in_automute) {
+				val = snd_hda_codec_read(codec, nid, 0,
+					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+				val &= ~PIN_HP;
+			} else
+				val = 0;
+			val |= pin_bits;
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    pin_bits);
+					    val);
 			break;
 		case ALC_AUTOMUTE_AMP:
 			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -621,17 +634,10 @@ static void alc_mic_automute(struct hda_codec *codec)
 		alc_mux_select(codec, 0, spec->int_mic_idx, false);
 }
 
-/* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+/* handle the specified unsol action (ALC_XXX_EVENT) */
+static void alc_exec_unsol_event(struct hda_codec *codec, int action)
 {
-	struct alc_spec *spec = codec->spec;
-	if (codec->vendor_id == 0x10ec0880)
-		res >>= 28;
-	else
-		res >>= 26;
-	if (spec->use_jack_tbl)
-		res = snd_hda_jack_get_action(codec, res);
-	switch (res) {
+	switch (action) {
 	case ALC_HP_EVENT:
 		alc_hp_automute(codec);
 		break;
@@ -645,6 +651,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 	snd_hda_jack_report_sync(codec);
 }
 
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	if (codec->vendor_id == 0x10ec0880)
+		res >>= 28;
+	else
+		res >>= 26;
+	res = snd_hda_jack_get_action(codec, res);
+	alc_exec_unsol_event(codec, res);
+}
+
 /* call init functions of standard auto-mute helpers */
 static void alc_inithook(struct hda_codec *codec)
 {
@@ -785,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
 		"Disabled", "Enabled"
 	};
 	static const char * const texts3[] = {
-		"Disabled", "Speaker Only", "Line-Out+Speaker"
+		"Disabled", "Speaker Only", "Line Out+Speaker"
 	};
 	const char