Merge branch 'board-specific' of git://github.com/hzhuang1/linux into next/boards

* 'board-specific' of git://github.com/hzhuang1/linux: (5 commits)
  ARM: pxa: add dummy clock for pxa25x and pxa27x
  ARM: mmp: append irq name of gpio device
  pxa/hx4700: Fix PXA_GPIO_IRQ_BASE/IRQ_NUM values
  pxa/hx4700: Add ASIC3 LED support
  pxa/hx4700: Correct StrataFlash block size discovery

(update to v3.3-rc5)

11 files changed, 79 insertions, 31 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9350f3c3bdf..3647baa9bfe 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;
@@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 static inline unsigned int get_ctl_pos(unsigned int data)
 {
 	hda_nid_t nid = get_amp_nid_(data);
-	unsigned int dir = get_amp_direction_(data);
+	unsigned int dir;
+	if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+		return 0;
+	dir = get_amp_direction_(data);
 	return (nid << 1) | dir;
 }
 
@@ -4374,6 +4379,7 @@ enum {
 	ALC882_FIXUP_ACER_ASPIRE_8930G,
 	ALC882_FIXUP_ASPIRE_8930G_VERBS,
 	ALC885_FIXUP_MACPRO_GPIO,
+	ALC889_FIXUP_DAC_ROUTE,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -4427,6 +4433,31 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec,
 	alc882_gpio_mute(codec, 1, 0);
 }
 
+/* Fix the connection of some pins for ALC889:
+ * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't
+ * work correctly (bko#42740)
+ */
+static void alc889_fixup_dac_route(struct hda_codec *codec,
+				   const struct alc_fixup *fix, int action)
+{
+	if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+		/* fake the connections during parsing the tree */
+		hda_nid_t conn1[2] = { 0x0c, 0x0d };
+		hda_nid_t conn2[2] = { 0x0e, 0x0f };
+		snd_hda_override_conn_list(codec, 0x14, 2, conn1);
+		snd_hda_override_conn_list(codec, 0x15, 2, conn1);
+		snd_hda_override_conn_list(codec, 0x18, 2, conn2);
+		snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+	} else if (action == ALC_FIXUP_ACT_PROBE) {
+		/* restore the connections */
+		hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+		snd_hda_override_conn_list(codec, 0x14, 5, conn);
+		snd_hda_override_conn_list(codec, 0x15, 5, conn);
+		snd_hda_override_conn_list(codec, 0x18, 5, conn);
+		snd_hda_override_conn_list(codec, 0x1a, 5, conn);
+	}
+}
+
 static const struct alc_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
 		.type = ALC_FIXUP_PINS,
@@ -4574,6 +4605,10 @@ static const struct alc_fixup alc882_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc885_fixup_macpro_gpio,
 	},
+	[ALC889_FIXUP_DAC_ROUTE] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc889_fixup_dac_route,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -4598,6 +4633,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
 		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+	SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
 	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 948f0be2f4f..6345df131a0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5078,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
 				spec->gpio_dir, spec->gpio_data);
 	} else {
 		notmtd_lvl = spec->gpio_led_polarity ?
-				AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD;
+				AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
 		muted_lvl = spec->gpio_led_polarity ?
-				AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
+				AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50;
 		spec->vref_led = muted ? muted_lvl : notmtd_lvl;
 		stac_vrefout_set(codec,	spec->vref_mute_led_nid,
 				 spec->vref_led);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 284e311040f..dff9a00ee8f 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
 	/* init input-src */
 	for (i = 0; i < spec->num_adc_nids; i++) {
 		int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx;
+		/* secondary ADCs must have the unique MUX */
+		if (i > 0 && !spec->mux_nids[i])
+			break;
 		if (spec->mux_nids[adc_idx]) {
 			int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx;
 			snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 9f3b01bb72c..e0a4263baa2 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	},
 	{
 		.subvendor = 0x161f,
+		.subdevice = 0x202f,
+		.name = "Gateway M520",
+		.type = AC97_TUNE_INV_EAPD
+	},
+	{
+		.subvendor = 0x161f,
 		.subdevice = 0x203a,
 		.name = "Gateway 4525GZ",		/* AD1981B */
 		.type = AC97_TUNE_INV_EAPD
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e..278c0a0575f 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 
 	SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
 			 0, 0xFF, 1, out_tlv),
-
-	SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
 };
 
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
-	SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+	SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
 
 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
 	SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HPOUTR"),
 	SND_SOC_DAPM_OUTPUT("LINEOUT"),
 
-	SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
-			   &ak4642_hpout_mixer_controls[0],
-			   ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+	SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+	SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+			    &ak4642_headphone_control),
 
-	SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
-			   &ak4642_hpout_mixer_controls[0],
-			   ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+	SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
 			   &ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 static const struct snd_soc_dapm_route ak4642_intercon[] = {
 
 	/* Outputs */
-	{"HPOUTL", NULL, "HPOUTL Mixer"},
-	{"HPOUTR", NULL, "HPOUTR Mixer"},
+	{"HPOUTL", NULL, "HPL Out"},
+	{"HPOUTR", NULL, "HPR Out"},
 	{"LINEOUT", NULL, "LINEOUT Mixer"},
 
-	{"HPOUTL Mixer", "DACH", "DAC"},
-	{"HPOUTR Mixer", "DACH", "DAC"},
+	{"HPL Out", NULL, "Headphone Enable"},
+	{"HPR Out", NULL, "Headphone Enable"},
+
+	{"Headphone Enable", "Switch", "DACH"},
+
+	{"DACH", NULL, "DAC"},
+
 	{"LINEOUT Mixer", "DACL", "DAC"},
 };
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 29c4b02c479..0ac228b7dc0 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
 
 static const struct soc_enum str_enum =
 	SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index db6c89a28bd..ea4a82d0116 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
-	int samples_pos = io->buff_sample_pos - 1;
 
-	if (samples_pos < 0)
-		samples_pos = 0;
-
-	return fsi_sample2frame(fsi, samples_pos);
+	return fsi_sample2frame(fsi, io->buff_sample_pos);
 }
 
 static struct snd_pcm_ops fsi_pcm_ops = {
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 2cf87f5afed..fde9a7a29cb 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
 
 	spin_lock(&dev->spinlock);
 
-	if (dev->input_panic || dev->output_panic)
+	if (dev->input_panic || dev->output_panic) {
 		ptr = SNDRV_PCM_POS_XRUN;
+		goto unlock;
+	}
 
 	if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
 		ptr = bytes_to_frames(sub->runtime,
 					dev->audio_in_buf_pos[index]);
 
+unlock:
 	spin_unlock(&dev->spinlock);
 	return ptr;
 }
diff --git a/sound/usb/card.h b/sound/usb/card.h
index a39edcc32a9..da5fa1ac4ed 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -1,6 +1,7 @@
 #ifndef __USBAUDIO_CARD_H
 #define __USBAUDIO_CARD_H
 
+#define MAX_NR_RATES	1024
 #define MAX_PACKS	20
 #define MAX_PACKS_HS	(MAX_PACKS * 8)	/* in high speed mode */
 #define MAX_URBS	8
diff --git a/sound/usb/format.c b/sound/usb/format.c
index e09aba19375..ddfef57c4c9 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 	return 0;
 }
 
-#define MAX_UAC2_NR_RATES 1024
-
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
@@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
 			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
 
 			nr_rates++;
-			if (nr_rates >= MAX_UAC2_NR_RATES) {
+			if (nr_rates >= MAX_NR_RATES) {
 				snd_printk(KERN_ERR "invalid uac2 rates\n");
 				break;
 			}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a3ddac0deff..27817266867 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	unsigned *rate_table = NULL;
 
 	fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
-	if (! fp) {
+	if (!fp) {
 		snd_printk(KERN_ERR "cannot memdup\n");
 		return -ENOMEM;
 	}
+	if (fp->nr_rates > MAX_NR_RATES) {
+		kfree(fp);
+		return -EINVAL;
+	}
 	if (fp->nr_rates > 0) {
 		rate_table = kmemdup(fp->rate_table,
 				     sizeof(int) * fp->nr_rates, GFP_KERNEL);