Merge branch 'perf/urgent' into perf/core

Merge reason: We are going to queue up a dependent patch.

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

11 files changed, 71 insertions, 21 deletions

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/hda_codec.c b/sound/pci/hda/hda_codec.c
index c2c65f63bf06..684307372d73 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -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/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bc5a993d1146..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;
 	}
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a7a5733aa4d2..d29d6d377904 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -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 3647baa9bfed..22c73b78ac6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -802,7 +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 * const *texts;
 
@@ -1856,7 +1856,7 @@ static const char * const alc_slave_vols[] = {
 	"Headphone Playback Volume",
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
-	"Line-Out Playback Volume",
+	"Line Out Playback Volume",
 	"CLFE Playback Volume",
 	"Bass Speaker Playback Volume",
 	"PCM Playback Volume",
@@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = {
 	"Speaker Playback Switch",
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
-	"Line-Out Playback Switch",
+	"Line Out Playback Switch",
 	"CLFE Playback Switch",
 	"Bass Speaker Playback Switch",
 	"PCM Playback Switch",
@@ -2068,12 +2068,16 @@ static int alc_build_controls(struct hda_codec *codec)
  */
 
 static void alc_init_special_input_src(struct hda_codec *codec);
+static int alc269_fill_coef(struct hda_codec *codec);
 
 static int alc_init(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	unsigned int i;
 
+	if (codec->vendor_id == 0x10ec0269)
+		alc269_fill_coef(codec);
+
 	alc_fix_pll(codec);
 	alc_auto_init_amp(codec, spec->init_amp);
 
@@ -3797,7 +3801,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec)
 	else
 		nums = spec->num_adc_nids;
 	for (c = 0; c < nums; c++)
-		alc_mux_select(codec, 0, spec->cur_mux[c], true);
+		alc_mux_select(codec, c, spec->cur_mux[c], true);
 }
 
 /* add mic boosts if needed */
@@ -4367,6 +4371,7 @@ enum {
 	ALC882_FIXUP_PB_M5210,
 	ALC882_FIXUP_ACER_ASPIRE_7736,
 	ALC882_FIXUP_ASUS_W90V,
+	ALC889_FIXUP_CD,
 	ALC889_FIXUP_VAIO_TT,
 	ALC888_FIXUP_EEE1601,
 	ALC882_FIXUP_EAPD,
@@ -4494,6 +4499,13 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ }
 		}
 	},
+	[ALC889_FIXUP_CD] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x1c, 0x993301f0 }, /* CD */
+			{ }
+		}
+	},
 	[ALC889_FIXUP_VAIO_TT] = {
 		.type = ALC_FIXUP_PINS,
 		.v.pins = (const struct alc_pincfg[]) {
@@ -4650,6 +4662,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
@@ -5467,8 +5480,12 @@ static const struct alc_model_fixup alc269_fixup_models[] = {
 
 static int alc269_fill_coef(struct hda_codec *codec)
 {
+	struct alc_spec *spec = codec->spec;
 	int val;
 
+	if (spec->codec_variant != ALC269_TYPE_ALC269VB)
+		return 0;
+
 	if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
 		alc_write_coef_idx(codec, 0xf, 0x960b);
 		alc_write_coef_idx(codec, 0xe, 0x8817);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df131a00..9dbb5735d778 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4629,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
 		unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
 		if (no_hp_sensing(spec, i))
 			continue;
-		if (presence)
+		if (1 /*presence*/)
 			stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
 #if 0 /* FIXME */
 /* Resetting the pinctl like below may lead to (a sort of) regressions
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cc9f6c83d661..bc030a2088da 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
 
 	hw->ops.open = snd_hdspm_hwdep_dummy_op;
 	hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
+	hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl;
 	hw->ops.release = snd_hdspm_hwdep_dummy_op;
 
 	return 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 01d1f749cf02..b6adbed6e506 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* data on rising edge of bclk, frame high 1clk before data */
-		strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+		strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
 		break;
 	}
 
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c6012ff5bd3e..d23b19a59d83 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -367,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.platform_name = "samsung-audio",
 	.cpu_dai_name = "s3c24xx-iis",
 	.codec_dai_name = "wm8753-hifi",
-	.codec_name = "wm8753-codec.0-001a",
+	.codec_name = "wm8753.0-001a",
 	.init = neo1973_wm8753_init,
 	.ops = &neo1973_hifi_ops,
 },
@@ -376,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.stream_name = "Voice",
 	.cpu_dai_name = "dfbmcs320-pcm",
 	.codec_dai_name = "wm8753-voice",
-	.codec_name = "wm8753-codec.0-001a",
+	.codec_name = "wm8753.0-001a",
 	.ops = &neo1973_voice_ops,
 },
 };
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1f55ded4047f..1315663c1c09 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 	 * standby.
 	 */
 	if (powerdown) {
-		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+		if (dapm->bias_level == SND_SOC_BIAS_ON)
+			snd_soc_dapm_set_bias_level(dapm,
+						    SND_SOC_BIAS_PREPARE);
 		dapm_seq_run(dapm, &down_list, 0, false);
-		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+		if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+			snd_soc_dapm_set_bias_level(dapm,
+						    SND_SOC_BIAS_STANDBY);
 	}
 }
 
@@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
 
 	list_for_each_entry(codec, &card->codec_dev_list, list) {
 		soc_dapm_shutdown_codec(&codec->dapm);
-		snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+			snd_soc_dapm_set_bias_level(&codec->dapm,
+						    SND_SOC_BIAS_OFF);
 	}
 }