Merge branch 'master' of /pub/scm/linux/kernel/git/torvalds/linux-2.6

28 files changed, 210 insertions, 58 deletions

diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index a7bf7a4b1f85..fb64c890109b 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -22,6 +22,10 @@
 #include <sound/opl3.h>
 #include <sound/asound_fm.h>
 
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#define OPL3_SUPPORT_SYNTH
+#endif
+
 /*
  *    There is 18 possible 2 OP voices
  *      (9 in the left and 9 in the right).
@@ -155,9 +159,11 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
 #endif
 		return snd_opl3_set_connection(opl3, (int) arg);
 
+#ifdef OPL3_SUPPORT_SYNTH
 	case SNDRV_DM_FM_IOCTL_CLEAR_PATCHES:
 		snd_opl3_clear_patches(opl3);
 		return 0;
+#endif
 
 #ifdef CONFIG_SND_DEBUG
 	default:
@@ -178,6 +184,7 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file)
 	return 0;
 }
 
+#ifdef OPL3_SUPPORT_SYNTH
 /*
  * write the device - load patches
  */
@@ -341,6 +348,7 @@ void snd_opl3_clear_patches(struct snd_opl3 *opl3)
 	}
 	memset(opl3->patch_table, 0, sizeof(opl3->patch_table));
 }
+#endif /* OPL3_SUPPORT_SYNTH */
 
 /* ------------------------------ */
 
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index 6304c3a89ba0..fe03bb820532 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -277,7 +277,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
 	} else {
 		snd_sbdsp_command(chip, 256 - runtime->rate_den);
 	}
-	if (chip->capture_format != SB_DSP_OUTPUT) {
+	if (chip->capture_format != SB_DSP_INPUT) {
 		count--;
 		snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE);
 		snd_sbdsp_command(chip, count & 0xff);
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index c9a2421cf6f0..4ecdd635ed1d 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -681,15 +681,12 @@ static struct snd_kcontrol_new snd_bt87x_capture_source = {
 
 static int snd_bt87x_free(struct snd_bt87x *chip)
 {
-	if (chip->mmio) {
+	if (chip->mmio)
 		snd_bt87x_stop(chip);
-		if (chip->irq >= 0)
-			synchronize_irq(chip->irq);
-
-		iounmap(chip->mmio);
-	}
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
+	if (chip->mmio)
+		iounmap(chip->mmio);
 	pci_release_regions(chip->pci);
 	pci_disable_device(chip->pci);
 	kfree(chip);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26812dc2b7f2..37c413923db8 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1055,6 +1055,12 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	const char **s;
 	int err;
 
+	for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++)
+		;
+	if (!*s) {
+		snd_printdd("No slave found for %s\n", name);
+		return 0;
+	}
 	kctl = snd_ctl_make_virtual_master(name, tlv);
 	if (!kctl)
 		return -ENOMEM;
@@ -1197,8 +1203,8 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
 	struct hda_bind_ctls *c;
 	int err;
 
-	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	kcontrol->private_value = *c->values;
 	err = c->ops->info(kcontrol, uinfo);
 	kcontrol->private_value = (long)c;
@@ -1213,8 +1219,8 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
 	struct hda_bind_ctls *c;
 	int err;
 
-	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	kcontrol->private_value = *c->values;
 	err = c->ops->get(kcontrol, ucontrol);
 	kcontrol->private_value = (long)c;
@@ -1230,8 +1236,8 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
 	unsigned long *vals;
 	int err = 0, change = 0;
 
-	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	for (vals = c->values; *vals; vals++) {
 		kcontrol->private_value = *vals;
 		err = c->ops->put(kcontrol, ucontrol);
@@ -1251,8 +1257,8 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	struct hda_bind_ctls *c;
 	int err;
 
-	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+	c = (struct hda_bind_ctls *)kcontrol->private_value;
 	kcontrol->private_value = *c->values;
 	err = c->ops->tlv(kcontrol, op_flag, size, tlv);
 	kcontrol->private_value = (long)c;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 56f8a3050751..4be36c84b36c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -275,6 +275,11 @@ enum {
 #define NVIDIA_HDA_TRANSREG_ADDR      0x4e
 #define NVIDIA_HDA_ENABLE_COHBITS     0x0f
 
+/* Defines for Intel SCH HDA snoop control */
+#define INTEL_SCH_HDA_DEVC      0x78
+#define INTEL_SCH_HDA_DEVC_NOSNOOP       (0x1<<11)
+
+
 /*
  */
 
@@ -868,6 +873,8 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg,
 
 static void azx_init_pci(struct azx *chip)
 {
+	unsigned short snoop;
+
 	/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
 	 * TCSEL == Traffic Class Select Register, which sets PCI express QOS
 	 * Ensuring these bits are 0 clears playback static on some HD Audio
@@ -888,6 +895,19 @@ static void azx_init_pci(struct azx *chip)
 				NVIDIA_HDA_TRANSREG_ADDR,
 				0x0f, NVIDIA_HDA_ENABLE_COHBITS);
 		break;
+	case AZX_DRIVER_SCH:
+		pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
+		if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
+			pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, \
+				snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP));
+			pci_read_config_word(chip->pci,
+				INTEL_SCH_HDA_DEVC, &snoop);
+			snd_printdd("HDA snoop disabled, enabling ... %s\n",\
+				(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) \
+				? "Failed" : "OK");
+		}
+		break;
+
         }
 }
 
@@ -1040,6 +1060,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 
 static unsigned int azx_max_codecs[] __devinitdata = {
 	[AZX_DRIVER_ICH] = 3,
+	[AZX_DRIVER_SCH] = 3,
 	[AZX_DRIVER_ATI] = 4,
 	[AZX_DRIVER_ATIHDMI] = 4,
 	[AZX_DRIVER_VIA] = 3,		/* FIXME: correct? */
@@ -1797,7 +1818,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 		 */
 		chip->playback_streams = (gcap & (0xF << 12)) >> 12;
 		chip->capture_streams = (gcap & (0xF << 8)) >> 8;
-		chip->playback_index_offset = (gcap & (0xF << 12)) >> 12;
+		chip->playback_index_offset = chip->capture_streams;
 		chip->capture_index_offset = 0;
 	} else {
 		/* gcap didn't give any info, switching to old method */
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 35a630d1770f..5633f77f8f3b 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -584,7 +584,8 @@ static void print_codec_info(struct snd_info_entry *entry,
 			print_amp_caps(buffer, codec, nid, HDA_INPUT);
 			snd_iprintf(buffer, "  Amp-In vals: ");
 			print_amp_vals(buffer, codec, nid, HDA_INPUT,
-				       wid_caps & AC_WCAP_STEREO, conn_len);
+				       wid_caps & AC_WCAP_STEREO,
+				       wid_type == AC_WID_PIN ? 1 : conn_len);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 19f08846d6fc..c8649282c2cf 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1778,9 +1778,9 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 static struct hda_input_mux ad1988_6stack_capture_source = {
 	.num_items = 5,
 	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mic", 0x4 },
+		{ "Front Mic", 0x1 },	/* port-B */
+		{ "Line", 0x2 },	/* port-C */
+		{ "Mic", 0x4 },		/* port-E */
 		{ "CD", 0x5 },
 		{ "Mix", 0x9 },
 	},
@@ -1789,7 +1789,7 @@ static struct hda_input_mux ad1988_6stack_capture_source = {
 static struct hda_input_mux ad1988_laptop_capture_source = {
 	.num_items = 3,
 	.items = {
-		{ "Mic/Line", 0x0 },
+		{ "Mic/Line", 0x1 },	/* port-B */
 		{ "CD", 0x5 },
 		{ "Mix", 0x9 },
 	},
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f6dd51cda7b2..7206b30cbf94 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -488,7 +488,7 @@ static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol,
 static hda_nid_t cxt5045_dac_nids[1] = { 0x19 };
 static hda_nid_t cxt5045_adc_nids[1] = { 0x1a };
 static hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a };
-#define CXT5045_SPDIF_OUT	0x13
+#define CXT5045_SPDIF_OUT	0x18
 
 static struct hda_channel_mode cxt5045_modes[1] = {
 	{ 2, NULL },
@@ -658,6 +658,7 @@ static struct hda_verb cxt5045_init_verbs[] = {
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE,
 	 AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
 	/* SPDIF route: PCM */
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{ 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 },
 	/* EAPD */
 	{0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ 
@@ -683,6 +684,7 @@ static struct hda_verb cxt5045_benq_init_verbs[] = {
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE,
 	 AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
 	/* SPDIF route: PCM */
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
 	/* EAPD */
 	{0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
@@ -781,7 +783,8 @@ static struct hda_verb cxt5045_test_init_verbs[] = {
 	 * PCM format, copyright asserted, no pre-emphasis and no validity
 	 * control.
 	 */
-	{0x13, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
 
 	/* Start with output sum widgets muted and their output gains at min */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -1227,6 +1230,11 @@ static struct hda_verb cxt5047_toshiba_init_verbs[] = {
 static struct hda_verb cxt5047_hp_init_verbs[] = {
 	/* pin sensing on HP jack */
 	{0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+	/* 0x13 is actually shared by both HP and speaker;
+	 * setting the connection to 0 (=0x19) makes the master volume control
+	 * working mysteriouslly...
+	 */
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
 	/* Record selector: Ext Mic */
 	{0x12, AC_VERB_SET_CONNECT_SEL,0x03},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 586d98f1b63d..33282f9c01c7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3973,8 +3973,8 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
 	ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
 	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -4005,9 +4005,9 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
 	ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
-	HDA_CODEC_VOLUME_MONO("Mono Speaker Playback Volume", 0x0a, 1, 0x0,
+	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
 			      HDA_OUTPUT),
-	HDA_BIND_MUTE_MONO("Mono Speaker Playback Switch", 0x0a, 1, 2,
+	HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
 			   HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
@@ -5227,10 +5227,14 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol,
 	const struct hda_input_mux *imux = spec->input_mux;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
 	static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
-	hda_nid_t nid = capture_mixers[adc_idx];
+	hda_nid_t nid;
 	unsigned int *cur_val = &spec->cur_mux[adc_idx];
 	unsigned int i, idx;
 
+	if (spec->num_adc_nids < 3)
+		nid = capture_mixers[adc_idx + 1];
+	else
+		nid = capture_mixers[adc_idx];
 	idx = ucontrol->value.enumerated.item[0];
 	if (idx >= imux->num_items)
 		idx = imux->num_items - 1;
@@ -6457,7 +6461,7 @@ static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol,
 	struct alc_spec *spec = codec->spec;
 	const struct hda_input_mux *imux = spec->input_mux;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
+	static hda_nid_t capture_mixers[2] = { 0x23, 0x22 };
 	hda_nid_t nid = capture_mixers[adc_idx];
 	unsigned int *cur_val = &spec->cur_mux[adc_idx];
 	unsigned int i, idx;
@@ -7635,6 +7639,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
 	SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
 	{}
@@ -8098,7 +8103,7 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	   HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+	   HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
@@ -8120,7 +8125,7 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	   HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+	   HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
 	/*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	{ } /* end */
@@ -9234,6 +9239,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
 	SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
 	SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
@@ -12989,8 +12995,8 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
 static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 
-	HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -13003,8 +13009,8 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
 };
 
 static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
-	HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
index 9ab4a9f383cb..5a158b73dcaa 100644
--- a/sound/pci/ice1712/phase.c
+++ b/sound/pci/ice1712/phase.c
@@ -51,7 +51,7 @@
 struct phase28_spec {
 	unsigned short master[2];
 	unsigned short vol[8];
-} phase28;
+};
 
 /* WM8770 registers */
 #define WM_DAC_ATTEN		0x00	/* DAC1-8 analog attenuation */
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index ddd5fc8d4fe1..301bf929acd9 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -36,7 +36,7 @@
 struct revo51_spec {
 	struct snd_i2c_device *dev;
 	struct snd_pt2258 *pt2258;
-} revo51;
+};
 
 static void revo_i2s_mclk_changed(struct snd_ice1712 *ice)
 {
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 061072c7db03..c52abd0bf22e 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1708,6 +1708,12 @@ static struct ac97_pcm ac97_pcm_defs[] __devinitdata = {
 };
 
 static struct ac97_quirk ac97_quirks[] __devinitdata = {
+        {
+		.subvendor = 0x0e11,
+		.subdevice = 0x000e,
+		.name = "Compaq Deskpro EN",	/* AD1885 */
+		.type = AC97_TUNE_HP_ONLY
+        },
 	{
 		.subvendor = 0x0e11,
 		.subdevice = 0x008a,
@@ -1740,6 +1746,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	},
 	{
 		.subvendor = 0x1025,
+		.subdevice = 0x0082,
+		.name = "Acer Travelmate 2310",
+		.type = AC97_TUNE_HP_ONLY
+	},
+	{
+		.subvendor = 0x1025,
 		.subdevice = 0x0083,
 		.name = "Acer Aspire 3003LCi",
 		.type = AC97_TUNE_HP_ONLY
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 3ea1f05228a1..666f69a3312e 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -150,6 +150,7 @@ static const struct oxygen_model model_hifier = {
 	.shortname = "C-Media CMI8787",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8788",
+	.owner = THIS_MODULE,
 	.init = hifier_init,
 	.control_filter = hifier_control_filter,
 	.mixer_init = hifier_mixer_init,
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index f31a0eb409b0..9a9941bb0460 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -28,7 +28,9 @@
  * GPIO 1 -> DFS1 of AK5385
  */
 
+#include <linux/mutex.h>
 #include <linux/pci.h>
+#include <sound/ac97_codec.h>
 #include <sound/control.h>
 #include <sound/core.h>
 #include <sound/initval.h>
@@ -37,6 +39,7 @@
 #include <sound/tlv.h>
 #include "oxygen.h"
 #include "ak4396.h"
+#include "cm9780.h"
 
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 driver");
@@ -75,6 +78,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids);
 #define GPIO_AK5385_DFS_DOUBLE	0x0001
 #define GPIO_AK5385_DFS_QUAD	0x0002
 
+#define GPIO_LINE_MUTE		CM9780_GPO0
+
 #define WM8785_R0	0
 #define WM8785_R1	1
 #define WM8785_R2	2
@@ -180,16 +185,23 @@ static void wm8785_init(struct oxygen *chip)
 	snd_component_add(chip->card, "WM8785");
 }
 
+static void cmi9780_init(struct oxygen *chip)
+{
+	oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE);
+}
+
 static void generic_init(struct oxygen *chip)
 {
 	ak4396_init(chip);
 	wm8785_init(chip);
+	cmi9780_init(chip);
 }
 
 static void meridian_init(struct oxygen *chip)
 {
 	ak4396_init(chip);
 	ak5385_init(chip);
+	cmi9780_init(chip);
 }
 
 static void generic_cleanup(struct oxygen *chip)
@@ -285,6 +297,27 @@ static void set_ak5385_params(struct oxygen *chip,
 			      value, GPIO_AK5385_DFS_MASK);
 }
 
+static void cmi9780_switch_hook(struct oxygen *chip, unsigned int codec,
+				unsigned int reg, int mute)
+{
+	if (codec != 0)
+		return;
+	switch (reg) {
+	case AC97_LINE:
+		oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
+					 mute ? GPIO_LINE_MUTE : 0,
+					 GPIO_LINE_MUTE);
+		break;
+	case AC97_MIC:
+	case AC97_CD:
+	case AC97_AUX:
+		if (!mute)
+			oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_STATUS,
+					     GPIO_LINE_MUTE);
+		break;
+	}
+}
+
 static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
 
 static int ak4396_control_filter(struct snd_kcontrol_new *template)
@@ -308,6 +341,7 @@ static const struct oxygen_model model_generic = {
 	.set_adc_params = set_wm8785_params,
 	.update_dac_volume = update_ak4396_volume,
 	.update_dac_mute = update_ak4396_mute,
+	.ac97_switch_hook = cmi9780_switch_hook,
 	.model_data_size = sizeof(struct generic_data),
 	.dac_channels = 8,
 	.used_channels = OXYGEN_CHANNEL_A |
@@ -331,6 +365,7 @@ static const struct oxygen_model model_meridian = {
 	.set_adc_params = set_ak5385_params,
 	.update_dac_volume = update_ak4396_volume,
 	.update_dac_mute = update_ak4396_mute,
+	.ac97_switch_hook = cmi9780_switch_hook,
 	.model_data_size = sizeof(struct generic_data),
 	.dac_channels = 8,
 	.used_channels = OXYGEN_CHANNEL_B |
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 6eb36dd11476..78c21155218e 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -204,7 +204,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry,
 	mutex_unlock(&chip->mutex);
 }
 
-static void __devinit oxygen_proc_init(struct oxygen *chip)
+static void oxygen_proc_init(struct oxygen *chip)
 {
 	struct snd_info_entry *entry;
 
@@ -215,7 +215,7 @@ static void __devinit oxygen_proc_init(struct oxygen *chip)
 #define oxygen_proc_init(chip)
 #endif
 
-static void __devinit oxygen_init(struct oxygen *chip)
+static void oxygen_init(struct oxygen *chip)
 {
 	unsigned int i;
 
@@ -399,8 +399,8 @@ static void oxygen_card_free(struct snd_card *card)
 	pci_disable_device(chip->pci);
 }
 
-int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
-			       int midi, const struct oxygen_model *model)
+int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
+		     int midi, const struct oxygen_model *model)
 {
 	struct snd_card *card;
 	struct oxygen *chip;
@@ -507,7 +507,7 @@ err_card:
 }
 EXPORT_SYMBOL(oxygen_pci_probe);
 
-void __devexit oxygen_pci_remove(struct pci_dev *pci)
+void oxygen_pci_remove(struct pci_dev *pci)
 {
 	snd_card_free(pci_get_drvdata(pci));
 	pci_set_drvdata(pci, NULL);
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index dfad3db35c82..b70046aca657 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -634,7 +634,7 @@ static void oxygen_pcm_free(struct snd_pcm *pcm)
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-int __devinit oxygen_pcm_init(struct oxygen *chip)
+int oxygen_pcm_init(struct oxygen *chip)
 {
 	struct snd_pcm *pcm;
 	int outs, ins;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 40e92f5cd69c..d163397b85cc 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -389,6 +389,7 @@ static const struct oxygen_model model_xonar = {
 	.shortname = "Asus AV200",
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
+	.owner = THIS_MODULE,
 	.init = xonar_init,
 	.control_filter = xonar_control_filter,
 	.mixer_init = xonar_mixer_init,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index c2bd4384316a..1be84f22d0de 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -745,7 +745,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp)
 
 
 #ifdef HDSP_FW_LOADER
-static int __devinit hdsp_request_fw_loader(struct hdsp *hdsp);
+static int hdsp_request_fw_loader(struct hdsp *hdsp);
 #endif
 
 static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand)
@@ -4688,8 +4688,7 @@ static struct snd_pcm_ops snd_hdsp_capture_ops = {
 	.copy =		snd_hdsp_capture_copy,
 };
 
-static int __devinit snd_hdsp_create_hwdep(struct snd_card *card,
-					   struct hdsp *hdsp)
+static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 {
 	struct snd_hwdep *hw;
 	int err;
@@ -4857,7 +4856,7 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp
 
 #ifdef HDSP_FW_LOADER
 /* load firmware via hotplug fw loader */
-static int __devinit hdsp_request_fw_loader(struct hdsp *hdsp)
+static int hdsp_request_fw_loader(struct hdsp *hdsp)
 {
 	const char *fwfile;
 	const struct firmware *fw;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 710e0287ef8c..569ecaca0e8b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -681,8 +681,8 @@ static const struct aic3x_rate_divs aic3x_divs[] = {
 	{22579200, 48000, 48000, 0x0, 8, 7075},
 	{33868800, 48000, 48000, 0x0, 5, 8049},
 	/* 64k */
-	{22579200, 96000, 96000, 0x1, 8, 7075},
-	{33868800, 96000, 96000, 0x1, 5, 8049},
+	{22579200, 64000, 96000, 0x1, 8, 7075},
+	{33868800, 64000, 96000, 0x1, 5, 8049},
 	/* 88.2k */
 	{22579200, 88200, 88200, 0x0, 8, 0},
 	{33868800, 88200, 88200, 0x0, 5, 3333},
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 590baea3c4c3..524f7450804f 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -176,7 +176,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec)
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path.
  */
-static int mixer_event (struct snd_soc_dapm_widget *w, int event)
+static int mixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
 {
 	u16 l, r, beep, line, phone, mic, pcm, aux;
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index f26c4b2e8b6e..a00aac7a71f1 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -315,7 +315,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
-	} else if (strcasecmp(sprop, "rj-master") == 0) {
+	} else if (strcasecmp(sprop, "rj-slave") == 0) {
 		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 3f34e531bebf..1a70a6ac98ce 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -215,7 +215,8 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+static int corgi_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
 		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
@@ -225,7 +226,8 @@ static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
 	return 0;
 }
 
-static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event)
+static int corgi_mic_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
 		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 5ae59bd309a3..4fbf8bba9627 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -196,7 +196,8 @@ static int poodle_set_spk(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event)
+static int poodle_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
 		locomo_gpio_write(&poodle_locomo_device.dev,
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d56709e15435..ecca39033fcc 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -215,7 +215,8 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event)
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
 {
 	if (machine_is_borzoi() || machine_is_spitz()) {
 		if (SND_SOC_DAPM_EVENT_ON(event))
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index e4d40b528ca4..7346d7e5d066 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -135,7 +135,8 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol,
 }
 
 /* tosa dapm event handlers */
-static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event)
+static int tosa_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
 		set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index 58d25e4e7d6c..7c44a2c7f963 100644
--- a/sound/usb/caiaq/caiaq-device.c
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -245,7 +245,7 @@ int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev,
 					  tmp, sizeof(tmp));
 }
 
-static void setup_card(struct snd_usb_caiaqdev *dev)
+static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
 {
 	int ret;
 	char val[4];
@@ -359,7 +359,7 @@ static struct snd_card* create_card(struct usb_device* usb_dev)
 	return card;
 }
 
-static int init_card(struct snd_usb_caiaqdev *dev)
+static int __devinit init_card(struct snd_usb_caiaqdev *dev)
 {
 	char *c;
 	struct usb_device *usb_dev = dev->chip.dev;
@@ -428,7 +428,7 @@ static int init_card(struct snd_usb_caiaqdev *dev)
 	return 0;
 }
 
-static int snd_probe(struct usb_interface *intf, 
+static int __devinit snd_probe(struct usb_interface *intf, 
 		     const struct usb_device_id *id)
 {
 	int ret;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 8fa935665702..675672f313be 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -479,6 +479,33 @@ static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs,
 	return 0;
 }
 
+/*
+ * process after E-Mu 0202/0404 high speed playback sync complete
+ *
+ * These devices return the number of samples per packet instead of the number
+ * of samples per microframe.
+ */
+static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs,
+					   struct snd_pcm_runtime *runtime,
+					   struct urb *urb)
+{
+	unsigned int f;
+	unsigned long flags;
+
+	if (urb->iso_frame_desc[0].status == 0 &&
+	    urb->iso_frame_desc[0].actual_length == 4) {
+		f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff;
+		f >>= subs->datainterval;
+		if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+			spin_lock_irqsave(&subs->lock, flags);
+			subs->freqm = f;
+			spin_unlock_irqrestore(&subs->lock, flags);
+		}
+	}
+
+	return 0;
+}
+
 /* determine the number of frames in the next packet */
 static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
 {
@@ -2219,10 +2246,17 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo
 	subs->stream = as;
 	subs->direction = stream;
 	subs->dev = as->chip->dev;
-	if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+	if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) {
 		subs->ops = audio_urb_ops[stream];
-	else
+	} else {
 		subs->ops = audio_urb_ops_high_speed[stream];
+		switch (as->chip->usb_id) {
+		case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
+		case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
+			subs->ops.retire_sync = retire_playback_sync_urb_hs_emu;
+			break;
+		}
+	}
 	snd_pcm_set_ops(as->pcm, stream,
 			stream == SNDRV_PCM_STREAM_PLAYBACK ?
 			&snd_usb_playback_ops : &snd_usb_capture_ops);
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 750e929d5870..6676a177c99e 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -104,12 +104,14 @@ struct snd_usb_midi {
 	struct usb_protocol_ops* usb_protocol_ops;
 	struct list_head list;
 	struct timer_list error_timer;
+	spinlock_t disc_lock;
 
 	struct snd_usb_midi_endpoint {
 		struct snd_usb_midi_out_endpoint *out;
 		struct snd_usb_midi_in_endpoint *in;
 	} endpoints[MIDI_MAX_ENDPOINTS];
 	unsigned long input_triggered;
+	unsigned char disconnected;
 };
 
 struct snd_usb_midi_out_endpoint {
@@ -306,6 +308,11 @@ static void snd_usbmidi_error_timer(unsigned long data)
 	struct snd_usb_midi *umidi = (struct snd_usb_midi *)data;
 	int i;
 
+	spin_lock(&umidi->disc_lock);
+	if (umidi->disconnected) {
+		spin_unlock(&umidi->disc_lock);
+		return;
+	}
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		struct snd_usb_midi_in_endpoint *in = umidi->endpoints[i].in;
 		if (in && in->error_resubmit) {
@@ -316,6 +323,7 @@ static void snd_usbmidi_error_timer(unsigned long data)
 		if (umidi->endpoints[i].out)
 			snd_usbmidi_do_output(umidi->endpoints[i].out);
 	}
+	spin_unlock(&umidi->disc_lock);
 }
 
 /* helper function to send static data that may not DMA-able */
@@ -1049,7 +1057,14 @@ void snd_usbmidi_disconnect(struct list_head* p)
 	int i;
 
 	umidi = list_entry(p, struct snd_usb_midi, list);
-	del_timer_sync(&umidi->error_timer);
+	/*
+	 * an URB's completion handler may start the timer and
+	 * a timer may submit an URB. To reliably break the cycle
+	 * a flag under lock must be used
+	 */
+	spin_lock_irq(&umidi->disc_lock);
+	umidi->disconnected = 1;
+	spin_unlock_irq(&umidi->disc_lock);
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
 		if (ep->out)
@@ -1062,6 +1077,7 @@ void snd_usbmidi_disconnect(struct list_head* p)
 		if (ep->in)
 			usb_kill_urb(ep->in->urb);
 	}
+	del_timer_sync(&umidi->error_timer);
 }
 
 static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi)
@@ -1685,6 +1701,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip,
 	umidi->quirk = quirk;
 	umidi->usb_protocol_ops = &snd_usbmidi_standard_ops;
 	init_timer(&umidi->error_timer);
+	spin_lock_init(&umidi->disc_lock);
 	umidi->error_timer.function = snd_usbmidi_error_timer;
 	umidi->error_timer.data = (unsigned long)umidi;