Merge branch 'tracing/urgent' into tracing/core

Merge reason: Pick up fixes and move base from -rc1 to -rc5.

Signed-off-by: Ingo Molnar <mingo@elte.hu>

14 files changed, 178 insertions, 68 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index dc78272..1f0f821 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
 	struct snd_ac97 *ac97;
 	int ret;
 
+	writel(0, aaci->base + AC97_POWERDOWN);
 	/*
 	 * Assert AACIRESET for 2us
 	 */
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 6e7d09a..7d722a0 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
 
 extern int use_internal_drums;
 
+static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
+				     struct snd_midi_channel *chan);
 /*
  * The next table looks magical, but it certainly is not. Its values have
  * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception
@@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data)
 	int again = 0;
 	int i;
 
-	spin_lock_irqsave(&opl3->sys_timer_lock, flags);
+	spin_lock_irqsave(&opl3->voice_lock, flags);
 	for (i = 0; i < opl3->max_voices; i++) {
 		struct snd_opl3_voice *vp = &opl3->voices[i];
 		if (vp->state > 0 && vp->note_off_check) {
 			if (vp->note_off == jiffies)
-				snd_opl3_note_off(opl3, vp->note, 0, vp->chan);
+				snd_opl3_note_off_unsafe(opl3, vp->note, 0,
+							 vp->chan);
 			else
 				again++;
 		}
 	}
+	spin_unlock_irqrestore(&opl3->voice_lock, flags);
+
+	spin_lock_irqsave(&opl3->sys_timer_lock, flags);
 	if (again) {
 		opl3->tlist.expires = jiffies + 1;	/* invoke again */
 		add_timer(&opl3->tlist);
@@ -658,15 +664,14 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice)
 /*
  * Release a note in response to a midi note off.
  */
-void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan)
+static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
+				     struct snd_midi_channel *chan)
 {
   	struct snd_opl3 *opl3;
 
 	int voice;
 	struct snd_opl3_voice *vp;
 
-	unsigned long flags;
-
 	opl3 = p;
 
 #ifdef DEBUG_MIDI
@@ -674,12 +679,9 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
 		   chan->number, chan->midi_program, note);
 #endif
 
-	spin_lock_irqsave(&opl3->voice_lock, flags);
-
 	if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
 		if (chan->drum_channel && use_internal_drums) {
 			snd_opl3_drum_switch(opl3, note, vel, 0, chan);
-			spin_unlock_irqrestore(&opl3->voice_lock, flags);
 			return;
 		}
 		/* this loop will hopefully kill all extra voices, because
@@ -697,6 +699,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
 			snd_opl3_kill_voice(opl3, voice);
 		}
 	}
+}
+
+void snd_opl3_note_off(void *p, int note, int vel,
+		       struct snd_midi_channel *chan)
+{
+	struct snd_opl3 *opl3 = p;
+	unsigned long flags;
+
+	spin_lock_irqsave(&opl3->voice_lock, flags);
+	snd_opl3_note_off_unsafe(p, note, vel, chan);
 	spin_unlock_irqrestore(&opl3->voice_lock, flags);
 }
 
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 24585c6..4e2b925 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = {
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
 	/* Leadtek Winfast tv 2000xp delux */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
+	/* Pinnacle PCTV */
+	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC),
 	/* Voodoo TV 200 */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
 	/* Askey Computer Corp. MagicTView'99 */
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index c8435c9..9fb6027 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -29,6 +29,9 @@
 #include "hda_codec.h"
 #include "hda_local.h"
 
+/* define below to restrict the supported rates and formats */
+/* #define LIMITED_RATE_FMT_SUPPORT */
+
 struct nvhdmi_spec {
 	struct hda_multi_out multiout;
 
@@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = {
 	{} /* terminator */
 };
 
+#ifdef LIMITED_RATE_FMT_SUPPORT
+/* support only the safe format and rate */
+#define SUPPORTED_RATES		SNDRV_PCM_RATE_48000
+#define SUPPORTED_MAXBPS	16
+#define SUPPORTED_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+#else
+/* support all rates and formats */
+#define SUPPORTED_RATES \
+	(SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+	SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+	 SNDRV_PCM_RATE_192000)
+#define SUPPORTED_MAXBPS	24
+#define SUPPORTED_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+#endif
+
 /*
  * Controls
  */
@@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = {
 	.channels_min = 2,
 	.channels_max = 8,
 	.nid = Nv_Master_Convert_nid,
-	.rates = SNDRV_PCM_RATE_48000,
-	.maxbps = 16,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SUPPORTED_RATES,
+	.maxbps = SUPPORTED_MAXBPS,
+	.formats = SUPPORTED_FORMATS,
 	.ops = {
 		.open = nvhdmi_dig_playback_pcm_open,
 		.close = nvhdmi_dig_playback_pcm_close_8ch,
@@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.nid = Nv_Master_Convert_nid,
-	.rates = SNDRV_PCM_RATE_48000,
-	.maxbps = 16,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SUPPORTED_RATES,
+	.maxbps = SUPPORTED_MAXBPS,
+	.formats = SUPPORTED_FORMATS,
 	.ops = {
 		.open = nvhdmi_dig_playback_pcm_open,
 		.close = nvhdmi_dig_playback_pcm_close_2ch,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7810d3d..c08ca66 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,7 +275,7 @@ struct alc_spec {
 	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
-	const struct hda_verb *init_verbs[5];	/* initialization verbs
+	const struct hda_verb *init_verbs[10];	/* initialization verbs
 						 * don't forget NULL
 						 * termination!
 						 */
@@ -1332,15 +1332,20 @@ do_sku:
 	 *	        when the external headphone out jack is plugged"
 	 */
 	if (!spec->autocfg.hp_pins[0]) {
+		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
-			spec->autocfg.hp_pins[0] = porta;
+			nid = porta;
 		else if (tmp == 1)
-			spec->autocfg.hp_pins[0] = porte;
+			nid = porte;
 		else if (tmp == 2)
-			spec->autocfg.hp_pins[0] = portd;
+			nid = portd;
 		else
 			return 1;
+		for (i = 0; i < spec->autocfg.line_outs; i++)
+			if (spec->autocfg.line_out_pins[i] == nid)
+				return 1;
+		spec->autocfg.hp_pins[0] = nid;
 	}
 
 	alc_init_auto_hp(codec);
@@ -1362,7 +1367,7 @@ static void alc_ssid_check(struct hda_codec *codec,
 }
 
 /*
- * Fix-up pin default configurations
+ * Fix-up pin default configurations and add default verbs
  */
 
 struct alc_pincfg {
@@ -1370,9 +1375,14 @@ struct alc_pincfg {
 	u32 val;
 };
 
-static void alc_fix_pincfg(struct hda_codec *codec,
+struct alc_fixup {
+	const struct alc_pincfg *pins;
+	const struct hda_verb *verbs;
+};
+
+static void alc_pick_fixup(struct hda_codec *codec,
 			   const struct snd_pci_quirk *quirk,
-			   const struct alc_pincfg **pinfix)
+			   const struct alc_fixup *fix)
 {
 	const struct alc_pincfg *cfg;
 
@@ -1380,9 +1390,14 @@ static void alc_fix_pincfg(struct hda_codec *codec,
 	if (!quirk)
 		return;
 
-	cfg = pinfix[quirk->value];
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+	fix += quirk->value;
+	cfg = fix->pins;
+	if (cfg) {
+		for (; cfg->nid; cfg++)
+			snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+	}
+	if (fix->verbs)
+		add_verb(codec->spec, fix->verbs);
 }
 
 /*
@@ -9593,11 +9608,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
 	{ }
 };
 
-static const struct alc_pincfg *alc882_pin_fixes[] = {
-	[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+static const struct alc_fixup alc882_fixups[] = {
+	[PINFIX_ABIT_AW9D_MAX] = {
+		.pins = alc882_abit_aw9d_pinfix
+	},
 };
 
-static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+static struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	{}
 };
@@ -9869,7 +9886,7 @@ static int patch_alc882(struct hda_codec *codec)
 		board_config = ALC882_AUTO;
 	}
 
-	alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+	alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
 
 	if (board_config == ALC882_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -12842,12 +12859,15 @@ static int patch_alc268(struct hda_codec *codec)
 		unsigned int wcap = get_wcaps(codec, 0x07);
 		int i;
 
+		spec->capsrc_nids = alc268_capsrc_nids;
 		/* get type */
 		wcap = get_wcaps_type(wcap);
 		if (spec->auto_mic ||
 		    wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc268_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
+			if (spec->auto_mic)
+				fixup_automic_adc(codec);
 			if (spec->auto_mic || spec->input_mux->num_items == 1)
 				add_mixer(spec, alc268_capture_nosrc_mixer);
 			else
@@ -12857,7 +12877,6 @@ static int patch_alc268(struct hda_codec *codec)
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
 			add_mixer(spec, alc268_capture_mixer);
 		}
-		spec->capsrc_nids = alc268_capsrc_nids;
 		/* set default input source */
 		for (i = 0; i < spec->num_adc_nids; i++)
 			snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
@@ -14357,15 +14376,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec)
 static void alc861_auto_init_hp_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
 
-	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
+	if (spec->autocfg.hp_outs)
+		alc861_auto_set_output_and_unmute(codec,
+						  spec->autocfg.hp_pins[0],
+						  PIN_HP,
 						  spec->multiout.hp_nid);
-	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+	if (spec->autocfg.speaker_outs)
+		alc861_auto_set_output_and_unmute(codec,
+						  spec->autocfg.speaker_pins[0],
+						  PIN_OUT,
 						  spec->multiout.dac_nids[0]);
 }
 
@@ -15158,7 +15178,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
 	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
-	SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+	/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
 	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
 	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
@@ -15551,6 +15571,29 @@ static void alc861vd_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
+enum {
+	ALC660VD_FIX_ASUS_GPIO1
+};
+
+/* reset GPIO1 */
+static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = {
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	{ }
+};
+
+static const struct alc_fixup alc861vd_fixups[] = {
+	[ALC660VD_FIX_ASUS_GPIO1] = {
+		.verbs = alc660vd_fix_asus_gpio1_verbs,
+	},
+};
+
+static struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
+	{}
+};
+
 static int patch_alc861vd(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
@@ -15572,6 +15615,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 		board_config = ALC861VD_AUTO;
 	}
 
+	alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+
 	if (board_config == ALC861VD_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc861vd_parse_auto_config(codec);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a9b2682..66c0876 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@ enum {
 	STAC_D965_5ST_NO_FP,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_927X_VOLKNOB,
 	STAC_927X_MODELS
 };
 
@@ -907,6 +908,16 @@ static struct hda_verb d965_core_init[] = {
 	{}
 };
 
+static struct hda_verb dell_3st_core_init[] = {
+	/* don't set delta bit */
+	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+	/* unmute node 0x1b */
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* select node 0x03 as DAC */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{}
+};
+
 static struct hda_verb stac927x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -915,6 +926,14 @@ static struct hda_verb stac927x_core_init[] = {
 	{}
 };
 
+static struct hda_verb stac927x_volknob_core_init[] = {
+	/* don't set delta bit */
+	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+	/* enable analog pc beep path */
+	{0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
+	{}
+};
+
 static struct hda_verb stac9205_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -1999,6 +2018,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 	[STAC_D965_5ST_NO_FP]  = d965_5st_no_fp_pin_configs,
 	[STAC_DELL_3ST]  = dell_3st_pin_configs,
 	[STAC_DELL_BIOS] = NULL,
+	[STAC_927X_VOLKNOB] = NULL,
 };
 
 static const char *stac927x_models[STAC_927X_MODELS] = {
@@ -2010,6 +2030,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
 	[STAC_D965_5ST_NO_FP]	= "5stack-no-fp",
 	[STAC_DELL_3ST]		= "dell-3stack",
 	[STAC_DELL_BIOS]	= "dell-bios",
+	[STAC_927X_VOLKNOB]	= "volknob",
 };
 
 static struct snd_pci_quirk stac927x_cfg_tbl[] = {
@@ -2045,6 +2066,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 			   "Intel D965", STAC_D965_5ST),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
 			   "Intel D965", STAC_D965_5ST),
+	/* volume-knob fixes */
+	SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB),
 	{} /* terminator */
 };
 
@@ -5612,10 +5635,14 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->dmic_nids = stac927x_dmic_nids;
 		spec->num_dmics = STAC927X_NUM_DMICS;
 
-		spec->init = d965_core_init;
+		spec->init = dell_3st_core_init;
 		spec->dmux_nids = stac927x_dmux_nids;
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
+	case STAC_927X_VOLKNOB:
+		spec->num_dmics = 0;
+		spec->init = stac927x_volknob_core_init;
+		break;
 	default:
 		spec->num_dmics = 0;
 		spec->init = stac927x_core_init;
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index 3756430..6da21a2 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice)
 
 	/* only use basic functionality for now */
 
-	ice->num_total_dacs = 2;	/* only PSDOUT0 is connected */
+	/* VT1616 6ch codec connected to PSDOUT0 using packed mode */
+	ice->num_total_dacs = 6;
 	ice->num_total_adcs = 2;
 
-	/* Chaintech AV-710 has another codecs, which need initialization */
-	/* initialize WM8728 codec */
+	/* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4
+	   (shared with the SPDIF output). Mixer control for this codec
+	   is not yet supported. */
 	if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) {
 		for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
 			wm_put(ice, wm_inits[i], wm_inits[i+1]);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index cecf1ff..d74033a 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info = snd_ice1712_pro_peak_info,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index af6e001..10fc92c 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
 	    (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
 		/* running? we cannot change the rate now... */
 		spin_unlock_irqrestore(&ice->reg_lock, flags);
-		return -EBUSY;
+		return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY;
 	}
 	if (!force && is_pro_rate_locked(ice)) {
 		spin_unlock_irqrestore(&ice->reg_lock, flags);
@@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device)
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(ice->pci),
-					      64*1024, 64*1024);
+					      256*1024, 256*1024);
 
 	ice->pcm = pcm;
 
@@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device)
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(ice->pci),
-					      64*1024, 64*1024);
+					      256*1024, 256*1024);
 
 	ice->pcm_ds = pcm;
 
@@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info = snd_vt1724_pro_peak_info,
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index acfa476..91683a3 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct via82xx *chip = snd_kcontrol_chip(kcontrol);
-	unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+	unsigned int idx = kcontrol->id.subdevice;
 
 	ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0];
 	ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1];
@@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct via82xx *chip = snd_kcontrol_chip(kcontrol);
-	unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+	unsigned int idx = kcontrol->id.subdevice;
 	unsigned long port = chip->port + 0x10 * idx;
 	unsigned char val;
 	int i, change = 0;
@@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata =
 };
 
 static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
-	.name = "VIA DXS Playback Volume",
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.device = 0,
+	/* .subdevice set later */
+	.name = "PCM Playback Volume",
 	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
 		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
-	.count = 4,
 	.info = snd_via8233_dxs_volume_info,
 	.get = snd_via8233_dxs_volume_get,
 	.put = snd_via8233_dxs_volume_put,
@@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
 		}
 		else /* Using DXS when PCM emulation is enabled is really weird */
 		{
-			/* Standalone DXS controls */
-			err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip));
-			if (err < 0)
-				return err;
+			for (i = 0; i < 4; ++i) {
+				struct snd_kcontrol *kctl;
+
+				kctl = snd_ctl_new1(
+					&snd_via8233_dxs_volume_control, chip);
+				if (!kctl)
+					return -ENOMEM;
+				kctl->id.subdevice = i;
+				err = snd_ctl_add(chip->card, kctl);
+				if (err < 0)
+					return err;
+			}
 		}
 	}
 	/* select spdif data slot 10/11 */
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373..593d5b9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
 	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
 			    WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
 	SOC_DAPM_SINGLE("PGA Capture Switch",
-			WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+			WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
 };
 
 /* Right Input Mixer */
@@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
 	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
 			    WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
 	SOC_DAPM_SINGLE("PGA Capture Switch",
-			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
 };
 
 /* Left Mic Mixer */
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index da97aae..1ef2454 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
 	codec->reg_cache = &wm8940->reg_cache;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
-	if (ret == 0) {
+	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
 	}
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2..ccdefe6 100644
--- a/sound/soc/imx/mxc-ssi.c
+++ b/sound/soc/imx/mxc-ssi.c
@@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		break;
 	}
 
-	/* sync */
-	if (!(fmt & SND_SOC_DAIFMT_ASYNC))
-		scr |= SSI_SCR_SYN;
-
-	/* tdm - only for stereo atm */
-	if (fmt & SND_SOC_DAIFMT_TDM)
-		scr |= SSI_SCR_NET;
-
 	if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
 		SSI1_STCR = stcr;
 		SSI1_SRCR = srcr;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f79711b..8de6f9d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 
 		/* connected jack or spk ? */
 		if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
-			widget->id == snd_soc_dapm_line)
+		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
 			return 1;
 	}
 
@@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 			return 1;
 
 		/* connected jack ? */
-		if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+		if (widget->id == snd_soc_dapm_mic ||
+		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
 			return 1;
 	}