Merge branch 'devel-stable' into devel

Conflicts:
	MAINTAINERS
	arch/arm/mm/fault.c

26 files changed, 164 insertions, 73 deletions

diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 5ca2220..1dd0c28 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -182,6 +182,10 @@ static int pmf_set_notify(struct gpio_runtime *rt,
 	if (!old && notify) {
 		irq_client = kzalloc(sizeof(struct pmf_irq_client),
 				     GFP_KERNEL);
+		if (!irq_client) {
+			err = -ENOMEM;
+			goto out_unlock;
+		}
 		irq_client->data = notif;
 		irq_client->handler = pmf_handle_notify_irq;
 		irq_client->owner = THIS_MODULE;
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 108b643..6205f37 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 
-	if (rtd && rtd->params)
+	if (rtd && rtd->params && rtd->params->drcmr)
 		*rtd->params->drcmr = 0;
 
 	snd_pcm_set_runtime_buffer(substream, NULL);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 333e4dd..72cfd47 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 		xrun(substream);
 		return -EPIPE;
 	}
+	if (xrun_debug(substream, 8)) {
+		char name[16];
+		pcm_debug_name(substream, name, sizeof(name));
+		snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, "
+			   "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+			   name, (unsigned int)pos,
+			   (unsigned int)runtime->period_size,
+			   (unsigned int)runtime->buffer_size,
+			   (unsigned long)old_hw_ptr,
+			   (unsigned long)runtime->hw_ptr_base,
+			   (unsigned long)runtime->hw_ptr_interrupt);
+	}
 	hw_base = runtime->hw_ptr_base;
 	new_hw_ptr = hw_base + pos;
 	hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
@@ -244,18 +256,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 			delta = new_hw_ptr - hw_ptr_interrupt;
 	}
 	if (delta < 0) {
-		delta += runtime->buffer_size;
+		if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr)
+			delta += runtime->buffer_size;
 		if (delta < 0) {
 			hw_ptr_error(substream, 
 				     "Unexpected hw_pointer value "
 				     "(stream=%i, pos=%ld, intr_ptr=%ld)\n",
 				     substream->stream, (long)pos,
 				     (long)hw_ptr_interrupt);
+#if 1
+			/* simply skipping the hwptr update seems more
+			 * robust in some cases, e.g. on VMware with
+			 * inaccurate timer source
+			 */
+			return 0; /* skip this update */
+#else
 			/* rebase to interrupt position */
 			hw_base = new_hw_ptr = hw_ptr_interrupt;
 			/* align hw_base to buffer_size */
 			hw_base -= hw_base % runtime->buffer_size;
 			delta = 0;
+#endif
 		} else {
 			hw_base += runtime->buffer_size;
 			if (hw_base >= runtime->boundary)
@@ -344,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
 		xrun(substream);
 		return -EPIPE;
 	}
+	if (xrun_debug(substream, 16)) {
+		char name[16];
+		pcm_debug_name(substream, name, sizeof(name));
+		snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, "
+			   "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+			   name, (unsigned int)pos,
+			   (unsigned int)runtime->period_size,
+			   (unsigned int)runtime->buffer_size,
+			   (unsigned long)old_hw_ptr,
+			   (unsigned long)runtime->hw_ptr_base,
+			   (unsigned long)runtime->hw_ptr_interrupt);
+	}
+
 	hw_base = runtime->hw_ptr_base;
 	new_hw_ptr = hw_base + pos;
 
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 1bcb360..941f64a 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -3,10 +3,6 @@
 # Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
 #
 
-ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-  obj-$(CONFIG_SND_SEQUENCER) += oss/
-endif
-
 snd-seq-device-objs := seq_device.o
 snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
                 seq_fifo.o seq_prioq.o seq_timer.o \
@@ -19,7 +15,8 @@ snd-seq-virmidi-objs := seq_virmidi.o
 
 obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
 ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+  obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+  obj-$(CONFIG_SND_SEQUENCER) += oss/
 endif
 obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
 
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index edb11ee..2dcf45b 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 		if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
 			continue;
 		/* load real volume - better precision */
-		spin_lock_irqsave(&gus->reg_lock, flags);
+		spin_lock(&gus->reg_lock);
 		snd_gf1_select_voice(gus, pvoice->number);
 		snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
 		vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
 		snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
 		pcmp->final_volume = 1;
-		spin_unlock_irqrestore(&gus->reg_lock, flags);
+		spin_unlock(&gus->reg_lock);
 	}
 	spin_unlock_irqrestore(&gus->voice_alloc, flags);
 	return change;
diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c
index 3ee9900..35b5912 100644
--- a/sound/oss/aedsp16.c
+++ b/sound/oss/aedsp16.c
@@ -325,8 +325,9 @@
 /*
  * Size of character arrays that store name and version of sound card
  */
-#define CARDNAMELEN 15		/* Size of the card's name in chars     */
-#define CARDVERLEN  2		/* Size of the card's version in chars  */
+#define CARDNAMELEN	15	/* Size of the card's name in chars     */
+#define CARDVERLEN	10	/* Size of the card's version in chars	*/
+#define CARDVERDIGITS	2	/* Number of digits in the version	*/
 
 #if defined(CONFIG_SC6600)
 /*
@@ -410,7 +411,7 @@
 
 static int      soft_cfg __initdata = 0;	/* bitmapped config */
 static int      soft_cfg_mss __initdata = 0;	/* bitmapped mss config */
-static int      ver[CARDVERLEN] __initdata = {0, 0};	/* DSP Ver:
+static int      ver[CARDVERDIGITS] __initdata = {0, 0};	/* DSP Ver:
 						   hi->ver[0] lo->ver[1] */
 
 #if defined(CONFIG_SC6600)
@@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port)
 	 * string is finished.
 	 */
 		ver[len++] = ret;
-	  } while (len < CARDVERLEN);
+	  } while (len < CARDVERDIGITS);
 	sprintf(DSPVersion, "%d.%d", ver[0], ver[1]);
 
 	DBG(("success.\n"));
diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c
index 1b2316f..734b8f9 100644
--- a/sound/oss/mpu401.c
+++ b/sound/oss/mpu401.c
@@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner)
 			sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name);
 		else
 			sprintf(mpu_synth_info[m].name,
-				"MPU-401 %d.%d%c Midi interface #%d",
+				"MPU-401 %d.%d%c MIDI #%d",
 				(int) (devc->version & 0xf0) >> 4,
 				devc->version & 0x0f,
 				revision_char,
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index f24bf1e..15e4138 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
 	.rate_max =		192000,
 	.channels_min =		2,
 	.channels_max =		2,
-	.buffer_bytes_max =	((65536 - 64) * 8),
+	.buffer_bytes_max =	65536 - 128,
 	.period_bytes_min =	64,
-	.period_bytes_max =	(65536 - 64),
+	.period_bytes_max =	32768 - 64,
 	.periods_min =		2,
 	.periods_max =		2,
 	.fifo_size =		0,
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a1db51b3..a7f4a67 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,
 
 	/* Allocate mem for amixer resource */
 	amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
-	if (NULL == amixer) {
-		err = -ENOMEM;
-		return err;
-	}
+	if (!amixer)
+		return -ENOMEM;
 
 	/* Check whether there are sufficient
 	 * amixer resources to meet request. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,
 
 	/* Allocate mem for sum resource */
 	sum = kzalloc(sizeof(*sum), GFP_KERNEL);
-	if (NULL == sum) {
-		err = -ENOMEM;
-		return err;
-	}
+	if (!sum)
+		return -ENOMEM;
 
 	/* Check whether there are sufficient sum resources to meet request. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 082e35c..deb6cfa 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
 
 struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
 	[LINEO1] = {.left = 0x40, .right = 0x41},
-	[LINEO2] = {.left = 0x70, .right = 0x71},
+	[LINEO2] = {.left = 0x60, .right = 0x61},
 	[LINEO3] = {.left = 0x50, .right = 0x51},
-	[LINEO4] = {.left = 0x60, .right = 0x61},
+	[LINEO4] = {.left = 0x70, .right = 0x71},
 	[LINEIM] = {.left = 0x45, .right = 0xc5},
 	[SPDIFOO] = {.left = 0x00, .right = 0x01},
 	[SPDIFIO] = {.left = 0x05, .right = 0x85},
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index e1c145d..df43a5c 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,
 
 	/* Allocate mem for SRCIMP resource */
 	srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
-	if (NULL == srcimp) {
-		err = -ENOMEM;
-		return err;
-	}
+	if (!srcimp)
+		return -ENOMEM;
 
 	/* Check whether there are sufficient SRCIMP resources. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d255d..88480c0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 						  AC_VERB_GET_CONNECT_LIST, i);
 		range_val = !!(parm & (1 << (shift-1))); /* ranges */
 		val = parm & mask;
+		if (val == 0) {
+			snd_printk(KERN_WARNING "hda_codec: "
+				   "invalid CONNECT_LIST verb %x[%i]:%x\n",
+				    nid, i, parm);
+			return 0;
+		}
 		parm >>= shift;
 		if (range_val) {
 			/* ranges between the previous and this one */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index fcad5ec..9446a5a 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -508,7 +508,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
 	char name[64];
 	char *sname;
 	long long val;
-	int n;
+	unsigned int n;
 
 	while (!snd_info_get_line(buffer, line, sizeof(line))) {
 		if (sscanf(line, "%s %llx", name, &val) != 2)
@@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
 				sname++;
 				n = 10 * n + name[4] - '0';
 			}
-			if (n < 0 || n > 31) /* double the CEA limit */
+			if (n >= ELD_MAX_SAD)
 				continue;
 			if (!strcmp(sname, "_coding_type"))
 				e->sad[n].format = val;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index be7d25f..3da85ca 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3754,7 +3754,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
 	int mute = (!ucontrol->value.integer.value[0] &&
 		    !ucontrol->value.integer.value[1]);
 	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
 			    mute ? 0x02 : 0x0);
 	return ret;
 }
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bbb9b42..b95df5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,13 +275,13 @@ struct alc_spec {
 						 */
 	unsigned int num_init_verbs;
 
-	char stream_name_analog[16];	/* analog PCM stream */
+	char stream_name_analog[32];	/* analog PCM stream */
 	struct hda_pcm_stream *stream_analog_playback;
 	struct hda_pcm_stream *stream_analog_capture;
 	struct hda_pcm_stream *stream_analog_alt_playback;
 	struct hda_pcm_stream *stream_analog_alt_capture;
 
-	char stream_name_digital[16];	/* digital PCM stream */
+	char stream_name_digital[32];	/* digital PCM stream */
 	struct hda_pcm_stream *stream_digital_playback;
 	struct hda_pcm_stream *stream_digital_capture;
 
@@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 					      &dig_nid, 1);
 		if (err < 0)
 			continue;
+		if (dig_nid > 0x7f) {
+			printk(KERN_ERR "alc880_auto: invalid dig_nid "
+				"connection 0x%x for NID 0x%x\n", dig_nid,
+				spec->autocfg.dig_out_pins[i]);
+			continue;
+		}
 		if (!i)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
@@ -10625,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
 	alc262_lenovo_3000_automute(codec, 1);
 }
 
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+				  int dir, int idx, long *valp)
+{
+	int i, change = 0;
+
+	for (i = 0; i < 2; i++, valp++)
+		change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+						   HDA_AMP_MUTE,
+						   *valp ? 0 : HDA_AMP_MUTE);
+	return change;
+}
+
 /* bind hp and internal speaker mute (with plug check) */
 static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
@@ -10633,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-	change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-
+	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+	change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
 	if (change)
 		alc262_fujitsu_automute(codec, 0);
 	return change;
@@ -10674,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-
+	change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
 	if (change)
 		alc262_lenovo_3000_automute(codec, 0);
 	return change;
@@ -11848,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-					  HDA_AMP_MUTE,
-					  valp[0] ? 0 : HDA_AMP_MUTE);
-	change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-					   HDA_AMP_MUTE,
-					   valp[1] ? 0 : HDA_AMP_MUTE);
+	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
 	if (change)
 		alc268_acer_automute(codec, 0);
 	return change;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a..5383d8c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1809,6 +1809,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 1537", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
 				"Dell Studio 17", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
+				"Dell Studio 1555", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
@@ -2378,6 +2380,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
 		      "Dell Vostro 1500", STAC_9205_DELL_M42),
 	/* Gateway */
+	SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
 	SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
 	{} /* terminator */
 };
@@ -4065,7 +4068,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
 	jack->nid = nid;
 	jack->type = type;
 
-	sprintf(name, "%s at %s %s Jack",
+	snprintf(name, sizeof(name), "%s at %s %s Jack",
 		snd_hda_get_jack_type(def_conf),
 		snd_hda_get_jack_connectivity(def_conf),
 		snd_hda_get_jack_location(def_conf));
@@ -5854,6 +5857,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
 };
 
 static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+	SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
+			   "Sony VAIO F/S", STAC_9872_VAIO),
 	{} /* terminator */
 };
 
@@ -5866,6 +5871,8 @@ static int patch_stac9872(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 	codec->spec = spec;
+	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+	spec->pin_nids = stac9872_pin_nids;
 
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
 							stac9872_models,
@@ -5877,8 +5884,6 @@ static int patch_stac9872(struct hda_codec *codec)
 		stac92xx_set_config_regs(codec,
 					 stac9872_brd_tbl[spec->board_config]);
 
-	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
-	spec->pin_nids = stac9872_pin_nids;
 	spec->multiout.dac_nids = spec->dac_nids;
 	spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
 	spec->adc_nids = stac9872_adc_nids;
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 235a71e..b5ca02e 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2197,9 +2197,12 @@ static int __init alsa_card_riptide_init(void)
 	if (err < 0)
 		return err;
 #if defined(SUPPORT_JOYSTICK)
-	pci_register_driver(&joystick_driver);
+	err = pci_register_driver(&joystick_driver);
+	/* On failure unregister formerly registered audio driver */
+	if (err < 0)
+		pci_unregister_driver(&driver);
 #endif
-	return 0;
+	return err;
 }
 
 static void __exit alsa_card_riptide_exit(void)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ab099f4..cb0d1bf 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
 	u16 pll_d = 1;
+	u8 reg;
 
 	/* select data word length */
 	data =
@@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 		pll_q &= 0xf;
 		aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
 		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
-	} else
+		/* disable PLL if it is bypassed */
+		reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+		aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE);
+
+	} else {
 		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+		/* enable PLL when it is used */
+		reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+		aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE);
+	}
 
 	/* Route Left DAC to left channel input and
 	 * right DAC to right channel input */
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index c05f718..8c0fdf8 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -1037,14 +1037,14 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi)
 	codec->control_data = spi;
 	codec->dev = &spi->dev;
 
-	spi->dev.driver_data = wm8988;
+	dev_set_drvdata(&spi->dev, wm8988);
 
 	return wm8988_register(wm8988);
 }
 
 static int __devexit wm8988_spi_remove(struct spi_device *spi)
 {
-	struct wm8988_priv *wm8988 = spi->dev.driver_data;
+	struct wm8988_priv *wm8988 = dev_get_drvdata(&spi->dev);
 
 	wm8988_unregister(wm8988);
 
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
index a96dcad..e96f941 100644
--- a/sound/soc/s3c24xx/s3c24xx-ac97.h
+++ b/sound/soc/s3c24xx/s3c24xx-ac97.h
@@ -20,12 +20,6 @@
 #define AC_CMD_ADDR(x) (x << 16)
 #define AC_CMD_DATA(x) (x & 0xffff)
 
-#ifdef CONFIG_CPU_S3C2440
-#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97
-#else
-#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97
-#endif
-
 extern struct snd_soc_dai s3c2443_ac97_dai[];
 
 #endif /*S3C24XXAC97_H_*/
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 523aec1..73525c0 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -48,6 +48,7 @@ config SND_USB_CAIAQ
 	    * Native Instruments Kore Controller
 	    * Native Instruments Kore Controller 2
 	    * Native Instruments Audio Kontrol 1
+	    * Native Instruments Audio 2 DJ
 	    * Native Instruments Audio 4 DJ
 	    * Native Instruments Audio 8 DJ
 	    * Native Instruments Guitar Rig Session I/O
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 8f9b60c..121af06 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
 		dev->samplerates |= SNDRV_PCM_RATE_192000;
 		/* fall thru */
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
 		dev->samplerates |= SNDRV_PCM_RATE_88200;
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index de38108..83e6c13 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,13 +35,14 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
 			 "{Native Instruments, Kore Controller},"
 			 "{Native Instruments, Kore Controller 2},"
 			 "{Native Instruments, Audio Kontrol 1},"
+			 "{Native Instruments, Audio 2 DJ},"
 			 "{Native Instruments, Audio 4 DJ},"
 			 "{Native Instruments, Audio 8 DJ},"
 			 "{Native Instruments, Session I/O},"
@@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = {
 		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
 		.idProduct =    USB_PID_AUDIO4DJ
 	},
+	{
+		.match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
+		.idProduct =    USB_PID_AUDIO2DJ
+	},
 	{ /* terminator */ }
 };
 
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index ece7351..44e3edf 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -10,6 +10,7 @@
 #define USB_PID_KORECONTROLLER	0x4711
 #define USB_PID_KORECONTROLLER2	0x4712
 #define USB_PID_AK1		0x0815
+#define USB_PID_AUDIO2DJ	0x041c
 #define USB_PID_AUDIO4DJ	0x0839
 #define USB_PID_AUDIO8DJ	0x1978
 #define USB_PID_SESSIONIO	0x1915
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c7b9023..44b9cdc 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2661,7 +2661,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 	struct usb_interface_descriptor *altsd;
 	int i, altno, err, stream;
 	int format;
-	struct audioformat *fp;
+	struct audioformat *fp = NULL;
 	unsigned char *fmt, *csep;
 	int num;
 
@@ -2734,6 +2734,18 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 			continue;
 		}
 
+		/*
+		 * Blue Microphones workaround: The last altsetting is identical
+		 * with the previous one, except for a larger packet size, but
+		 * is actually a mislabeled two-channel setting; ignore it.
+		 */
+		if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
+		    fp && fp->altsetting == 1 && fp->channels == 1 &&
+		    fp->format == SNDRV_PCM_FORMAT_S16_LE &&
+		    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
+							fp->maxpacksize * 2)
+			continue;
+
 		csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
 		/* Creamware Noah has this descriptor after the 2nd endpoint */
 		if (!csep && altsd->bNumEndpoints >= 2)
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 4bd3a7a..ec9cdf9 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
 		break;
 	}
 
-	/* quirk for UDA1321/N101 */
-	/* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
-	/* is not very clear from datasheets */
-	/* I hope that the min value is -15360 for newer firmware --jk */
+	/* volume control quirks */
 	switch (state->chip->usb_id) {
 	case USB_ID(0x0471, 0x0101):
 	case USB_ID(0x0471, 0x0104):
 	case USB_ID(0x0471, 0x0105):
 	case USB_ID(0x0672, 0x1041):
+	/* quirk for UDA1321/N101.
+	 * note that detection between firmware 2.1.1.7 (N101)
+	 * and later 2.1.1.21 is not very clear from datasheets.
+	 * I hope that the min value is -15360 for newer firmware --jk
+	 */
 		if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
 		    cval->min == -15616) {
-			snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n");
+			snd_printk(KERN_INFO
+				 "set volume quirk for UDA1321/N101 chip\n");
 			cval->max = -256;
 		}
+		break;
+
+	case USB_ID(0x046d, 0x09a4):
+		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+			snd_printk(KERN_INFO
+				"set volume quirk for QuickCam E3500\n");
+			cval->min = 6080;
+			cval->max = 8768;
+			cval->res = 192;
+		}
+		break;
+
 	}
 
 	snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",