Merge branch 'linus' into tracing/ftrace

9 files changed, 104 insertions, 48 deletions

diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index f07cc1ee1fe..1d661f795e8 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -24,7 +24,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 /* default timer freq for PC-Speaker: 18643 Hz */
 #define DIV_18KHZ 64
 #define MAX_DIV DIV_18KHZ
-#define CUR_DIV() (MAX_DIV >> chip->treble)
+#define CALC_DIV(d) (MAX_DIV >> (d))
+#define CUR_DIV() CALC_DIV(chip->treble)
 #define PCSP_MAX_TREBLE 1
 
 /* unfortunately, with hrtimers 37KHz does not work very well :( */
@@ -36,7 +37,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 #define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1)
 #define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV)
 #define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble))
-#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV())
+#define PCSP_CALC_RATE(i) (PIT_TICK_RATE / CALC_DIV(i))
+#define PCSP_RATE() PCSP_CALC_RATE(chip->treble)
 #define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE
 #define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE
 #define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1)
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 64a695fef74..caeb0f57fcc 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,7 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.enumerated.items = chip->max_treble + 1;
 	if (uinfo->value.enumerated.item > chip->max_treble)
 		uinfo->value.enumerated.item = chip->max_treble;
-	sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE());
+	sprintf(uinfo->value.enumerated.name, "%d",
+			PCSP_CALC_RATE(uinfo->value.enumerated.item));
 	return 0;
 }
 
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 2da89810ca1..1292dcee072 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1971,6 +1971,9 @@ static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd
 
 	val = ac97->regs[AC97_AD_MISC];
 	ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL);
+	if (ac97->spec.ad18xx.lo_as_master)
+		ucontrol->value.integer.value[0] =
+			!ucontrol->value.integer.value[0];
 	return 0;
 }
 
@@ -1979,8 +1982,10 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd
 	struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
 	unsigned short val;
 
-	val = !ucontrol->value.integer.value[0]
-		? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
+	val = !ucontrol->value.integer.value[0];
+	if (ac97->spec.ad18xx.lo_as_master)
+		val = !val;
+	val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
 	return snd_ac97_update_bits(ac97, AC97_AD_MISC,
 				    AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val);
 }
@@ -2031,7 +2036,7 @@ static void ad1888_update_jacks(struct snd_ac97 *ac97)
 {
 	unsigned short val = 0;
 	/* clear LODIS if shared jack is to be used for Surround out */
-	if (is_shared_linein(ac97))
+	if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97))
 		val |= (1 << 12);
 	/* clear CLDIS if shared jack is to be used for C/LFE out */
 	if (is_shared_micin(ac97))
@@ -2067,9 +2072,13 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
 
 static int patch_ad1888_specific(struct snd_ac97 *ac97)
 {
-	/* rename 0x04 as "Master" and 0x02 as "Master Surround" */
-	snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Master Surround Playback");
-	snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+	if (!ac97->spec.ad18xx.lo_as_master) {
+		/* rename 0x04 as "Master" and 0x02 as "Master Surround" */
+		snd_ac97_rename_vol_ctl(ac97, "Master Playback",
+					"Master Surround Playback");
+		snd_ac97_rename_vol_ctl(ac97, "Headphone Playback",
+					"Master Playback");
+	}
 	return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls));
 }
 
@@ -2088,16 +2097,27 @@ static int patch_ad1888(struct snd_ac97 * ac97)
 	
 	patch_ad1881(ac97);
 	ac97->build_ops = &patch_ad1888_build_ops;
-	/* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
-	/* it seems that most vendors connect line-out connector to headphone out of AC'97 */
+
+	/*
+	 * LO can be used as a real line-out on some devices,
+	 * and we need to revert the front/surround mixer switches
+	 */
+	if (ac97->subsystem_vendor == 0x1043 &&
+	    ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */
+		ac97->spec.ad18xx.lo_as_master = 1;
+
+	misc = snd_ac97_read(ac97, AC97_AD_MISC);
 	/* AD-compatible mode */
 	/* Stereo mutes enabled */
-	misc = snd_ac97_read(ac97, AC97_AD_MISC);
-	snd_ac97_write_cache(ac97, AC97_AD_MISC, misc |
-			     AC97_AD198X_LOSEL |
-			     AC97_AD198X_HPSEL |
-			     AC97_AD198X_MSPLT |
-			     AC97_AD198X_AC97NC);
+	misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC;
+	if (!ac97->spec.ad18xx.lo_as_master)
+		/* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
+		/* it seems that most vendors connect line-out connector to
+		 * headphone out of AC'97
+		 */
+		misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL;
+
+	snd_ac97_write_cache(ac97, AC97_AD_MISC, misc);
 	ac97->flags |= AC97_STEREO_MUTES;
 	return 0;
 }
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index abde5b90188..548c9cc81af 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1818,13 +1818,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	}
 	emu->port = pci_resource_start(pci, 0);
 
-	if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
-			"EMU10K1", emu)) {
-		err = -EBUSY;
-		goto error;
-	}
-	emu->irq = pci->irq;
-
 	emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT;
 	if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
 				32 * 1024, &emu->ptb_pages) < 0) {
@@ -1887,6 +1880,14 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 	emu->fx8010.etram_pages.area = NULL;
 	emu->fx8010.etram_pages.bytes = 0;
 
+	/* irq handler must be registered after I/O ports are activated */
+	if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
+			"EMU10K1", emu)) {
+		err = -EBUSY;
+		goto error;
+	}
+	emu->irq = pci->irq;
+
 	/*
 	 *  Init to 0x02109204 :
 	 *  Clock accuracy    = 0     (1000ppm)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605adde4..a99e86d7427 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = {
 static struct snd_pci_quirk ad1988_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
 	{}
 };
 
@@ -3643,33 +3644,17 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	{ } /* end */
 };
 
-static struct hda_input_mux ad1884a_mobile_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x1 }, /* port-C */
-		{ "Mix", 0x3 },
-	},
-};
-
 static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -3686,14 +3671,31 @@ static void ad1884a_hp_automute(struct hda_codec *codec)
 			    present ? 0x00 : 0x02);
 }
 
+/* switch to external mic if plugged */
+static void ad1884a_hp_automic(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
+			    present ? 0 : 1);
+}
+
 #define AD1884A_HP_EVENT		0x37
+#define AD1884A_MIC_EVENT		0x36
 
 /* unsolicited event for HP jack sensing */
 static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1884a_hp_automute(codec);
+	switch (res >> 26) {
+	case AD1884A_HP_EVENT:
+		ad1884a_hp_automute(codec);
+		break;
+	case AD1884A_MIC_EVENT:
+		ad1884a_hp_automic(codec);
+		break;
+	}
 }
 
 /* initialize jack-sensing, too */
@@ -3701,6 +3703,7 @@ static int ad1884a_hp_init(struct hda_codec *codec)
 {
 	ad198x_init(codec);
 	ad1884a_hp_automute(codec);
+	ad1884a_hp_automic(codec);
 	return 0;
 }
 
@@ -3714,10 +3717,15 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
 	/* Port-F pin */
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Port-C pin - internal mic-in */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
 	/* analog mix */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	/* unsolicited event for pin-sense */
 	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
 	{ } /* end */
 };
 
@@ -3877,7 +3885,6 @@ static int patch_ad1884a(struct hda_codec *codec)
 		spec->mixers[0] = ad1884a_mobile_mixers;
 		spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
 		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1884a_mobile_capture_source;
 		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
 		codec->patch_ops.init = ad1884a_hp_init;
 		break;
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index c73ce074a6e..6ef57fbfb6e 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = {
 
 static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
+	SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
 	SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
 	{} /* terminator */
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 864b2f598c3..b0a2a262ece 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -853,6 +853,7 @@ do_sku:
 		case 0x10ec0269:
 		case 0x10ec0862:
 		case 0x10ec0662:	
+		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x14, 0,
 					    AC_VERB_SET_EAPD_BTLENABLE, 2);
 			snd_hda_codec_write(codec, 0x15, 0,
@@ -877,6 +878,7 @@ do_sku:
 		case 0x10ec0883:
 		case 0x10ec0885:
 		case 0x10ec0888:
+		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_COEF_INDEX, 7);
 			tmp = snd_hda_codec_read(codec, 0x20, 0,
@@ -940,7 +942,6 @@ do_sku:
 			    AC_VERB_SET_UNSOLICITED_ENABLE,
 			    AC_USRSP_EN | ALC880_HP_EVENT);
 	spec->unsol_event = alc_sku_unsol_event;
-	spec->init_hook = alc_sku_automute;	
 }
 
 /*
@@ -7743,6 +7744,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
@@ -8640,6 +8642,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
 
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{}
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
@@ -10510,6 +10513,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
 	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
 	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 393f7fd2b1b..a4f44a00bae 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
 	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
 	{ } /* end */
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 52b1d81a26f..e7e43524f8c 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
 	},
 };
 
+static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = 0x10, /* NID to query formats and rates */
+	/* We got noisy outputs on the right channel on VT1708 when
+	 * 24bit samples are used.  Until any workaround is found,
+	 * disable the 24bit format, so far.
+	 */
+	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.ops = {
+		.open = via_playback_pcm_open,
+		.prepare = via_playback_pcm_prepare,
+		.cleanup = via_playback_pcm_cleanup
+	},
+};
+
 static struct hda_pcm_stream vt1708_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
@@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec)
 	
 	spec->stream_name_analog = "VT1708 Analog";
 	spec->stream_analog_playback = &vt1708_pcm_analog_playback;
+	/* disable 32bit format on VT1708 */
+	if (codec->vendor_id == 0x11061708)
+		spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
 	spec->stream_analog_capture = &vt1708_pcm_analog_capture;
 
 	spec->stream_name_digital = "VT1708 Digital";