Merge branch 'for-rmk' of git://git.pengutronix.de/git/imx/linux-2.6 into devel

Conflicts:

	arch/arm/mach-at91/gpio.c

23 files changed, 147 insertions, 100 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 89096e811a4..772901e41ec 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	 */
 	do {
 		v = readl(aaci->base + AACI_SLFR);
-	} while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--);
+	} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
 
 	if (!timeout)
 		dev_err(&aaci->dev->dev,
@@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	 */
 	do {
 		v = readl(aaci->base + AACI_SLFR);
-	} while ((v & SLFR_1TXB) && timeout--);
+	} while ((v & SLFR_1TXB) && --timeout);
 
 	if (!timeout) {
 		dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
@@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	do {
 		cond_resched();
 		v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
-	} while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--);
+	} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
 
 	if (!timeout) {
 		dev_err(&aaci->dev->dev, "timeout on RX valid\n");
diff --git a/sound/core/jack.c b/sound/core/jack.c
index dd4a12dc09a..077a85262c1 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device)
 	int err;
 
 	snprintf(jack->name, sizeof(jack->name), "%s %s",
-		 card->longname, jack->id);
+		 card->shortname, jack->id);
 	jack->input_dev->name = jack->name;
 
 	/* Default to the sound card device. */
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index a466443c4a2..2fa9299a440 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin,
 		while (dst_frames1 > 0) {
 			S1 = S2;
 			if (src_frames1-- > 0) {
-				S1 = *src;
+				S2 = *src;
 				src += src_step;
 			}
 			if (pos & ~R_MASK) {
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5b89c0883d6..48b64e6b267 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -706,7 +706,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
 	mtp_card->card = card;
 	mtp_card->irq = -1;
 	mtp_card->share_irq = 0;
-	mtp_card->inmidiport = 0xffffffff;
 	mtp_card->inmidistate = 0;
 	mtp_card->outmidihwport = 0xffffffff;
 	init_timer(&mtp_card->timer);
@@ -719,6 +718,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
 	if (err < 0)
 		goto __error;
 
+	mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST;
+
 	err = snd_mtpav_get_ISA(mtp_card);
 	if (err < 0)
 		goto __error;
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 57d9f154c88..38931f2f696 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -847,23 +847,23 @@ static int __init AtaIrqInit(void)
 	   of events. So all we need to keep the music playing is
 	   to provide the sound hardware with new data upon
 	   an interrupt from timer A. */
-	mfp.tim_ct_a = 0;	/* ++roman: Stop timer before programming! */
-	mfp.tim_dt_a = 1;	/* Cause interrupt after first event. */
-	mfp.tim_ct_a = 8;	/* Turn on event counting. */
+	st_mfp.tim_ct_a = 0;	/* ++roman: Stop timer before programming! */
+	st_mfp.tim_dt_a = 1;	/* Cause interrupt after first event. */
+	st_mfp.tim_ct_a = 8;	/* Turn on event counting. */
 	/* Register interrupt handler. */
 	if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound",
 			AtaInterrupt))
 		return 0;
-	mfp.int_en_a |= 0x20;	/* Turn interrupt on. */
-	mfp.int_mk_a |= 0x20;
+	st_mfp.int_en_a |= 0x20;	/* Turn interrupt on. */
+	st_mfp.int_mk_a |= 0x20;
 	return 1;
 }
 
 #ifdef MODULE
 static void AtaIrqCleanUp(void)
 {
-	mfp.tim_ct_a = 0;	/* stop timer */
-	mfp.int_en_a &= ~0x20;	/* turn interrupt off */
+	st_mfp.tim_ct_a = 0;		/* stop timer */
+	st_mfp.int_en_a &= ~0x20;	/* turn interrupt off */
 	free_irq(IRQ_MFP_TIMA, AtaInterrupt);
 }
 #endif /* MODULE */
@@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void)
 		is_falcon = 0;
 	    } else
 		return -ENODEV;
-	    if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0)
+	    if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0)
 		return dmasound_init();
 	    else {
 		printk("DMA sound driver: Timer A interrupt already in use\n");
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f..c7c54e7748e 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
 
 static struct pci_device_id snd_aw2_ids[] = {
-	{PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID,
+	{PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
 	 0, 0, 0},
 	{0}
 };
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d6..101a1c13a20 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
+	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
 	 .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]",
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0b708134d12..d03f99298be 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3088,6 +3088,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare);
 
+int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
+				  struct hda_multi_out *mout)
+{
+	mutex_lock(&codec->spdif_mutex);
+	cleanup_dig_out_stream(codec, mout->dig_out_nid);
+	mutex_unlock(&codec->spdif_mutex);
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup);
+
 /*
  * release the digital out
  */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 300ab407cf4..4ae51dcb81a 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -175,7 +175,7 @@ static int reconfig_codec(struct hda_codec *codec)
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
 		return err;
-	return 0;
+	return snd_card_register(codec->bus->card);
 }
 
 /*
@@ -277,18 +277,19 @@ static ssize_t init_verbs_store(struct device *dev,
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
-	char *p;
-	struct hda_verb verb, *v;
+	struct hda_verb *v;
+	int nid, verb, param;
 
-	verb.nid = simple_strtoul(buf, &p, 0);
-	verb.verb = simple_strtoul(p, &p, 0);
-	verb.param = simple_strtoul(p, &p, 0);
-	if (!verb.nid || !verb.verb || !verb.param)
+	if (sscanf(buf, "%i %i %i", &nid, &verb, &param) != 3)
+		return -EINVAL;
+	if (!nid || !verb)
 		return -EINVAL;
 	v = snd_array_new(&codec->init_verbs);
 	if (!v)
 		return -ENOMEM;
-	*v = verb;
+	v->nid = nid;
+	v->verb = verb;
+	v->param = param;
 	return count;
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 11e791b965f..5e909e0da04 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1947,16 +1947,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 	return 0;
 }
 
-static int azx_resume_early(struct pci_dev *pci)
-{
-	return pci_restore_state(pci);
-}
-
 static int azx_resume(struct pci_dev *pci)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct azx *chip = card->private_data;
 
+	pci_set_power_state(pci, PCI_D0);
+	pci_restore_state(pci);
 	if (pci_enable_device(pci) < 0) {
 		printk(KERN_ERR "hda-intel: pci_enable_device failed, "
 		       "disabling device\n");
@@ -2098,6 +2095,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
 	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
 	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
+	/* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */
+	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03),
 	{}
 };
 
@@ -2468,7 +2467,6 @@ static struct pci_driver driver = {
 	.remove = __devexit_p(azx_remove),
 #ifdef CONFIG_PM
 	.suspend = azx_suspend,
-	.resume_early = azx_resume_early,
 	.resume = azx_resume,
 #endif
 };
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 1dd8716c387..44f189cb97a 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -251,6 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 				  unsigned int stream_tag,
 				  unsigned int format,
 				  struct snd_pcm_substream *substream);
+int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
+				  struct hda_multi_out *mout);
 int snd_hda_multi_out_analog_open(struct hda_codec *codec,
 				  struct hda_multi_out *mout,
 				  struct snd_pcm_substream *substream,
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2e7371ec2e2..e48612323aa 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -275,6 +275,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     format, substream);
 }
 
+static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					   struct hda_codec *codec,
+					   struct snd_pcm_substream *substream)
+{
+	struct ad198x_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 /*
  * Analog capture
  */
@@ -333,7 +341,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = {
 	.ops = {
 		.open = ad198x_dig_playback_pcm_open,
 		.close = ad198x_dig_playback_pcm_close,
-		.prepare = ad198x_dig_playback_pcm_prepare
+		.prepare = ad198x_dig_playback_pcm_prepare,
+		.cleanup = ad198x_dig_playback_pcm_cleanup
 	},
 };
 
@@ -1885,8 +1894,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 #define AD1988_SPDIF_OUT_HDMI	0x0b
 #define AD1988_SPDIF_IN		0x07
 
-static hda_nid_t ad1989b_slave_dig_outs[2] = {
-	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI
+static hda_nid_t ad1989b_slave_dig_outs[] = {
+	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
 };
 
 static struct hda_input_mux ad1988_6stack_capture_source = {
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 3564f4e4b74..fcc77fec448 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = {
 	{} /* terminator */
 };
 
-static struct hda_verb pinout_disable_verb[] = {
-	{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00},
-	{}
-};
-
 static struct hda_verb unsolicited_response_verb[] = {
 	{PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
 						  INTEL_HDMI_EVENT_TAG},
@@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
 
 static void hdmi_enable_output(struct hda_codec *codec)
 {
-	/* Enable Audio InfoFrame Transmission */
-	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
-	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
-						AC_DIPXMIT_BEST);
 	/* Unmute */
 	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
 		snd_hda_codec_write(codec, PIN_NID, 0,
@@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec)
 	snd_hda_sequence_write(codec, pinout_enable_verb);
 }
 
-static void hdmi_disable_output(struct hda_codec *codec)
+/*
+ * Enable Audio InfoFrame Transmission
+ */
+static void hdmi_start_infoframe_trans(struct hda_codec *codec)
 {
-	snd_hda_sequence_write(codec, pinout_disable_verb);
-	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
-		snd_hda_codec_write(codec, PIN_NID, 0,
-				AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+						AC_DIPXMIT_BEST);
+}
 
-	/*
-	 * FIXME: noises may arise when playing music after reloading the
-	 * kernel module, until the next X restart or monitor repower.
-	 */
+/*
+ * Disable Audio InfoFrame Transmission
+ */
+static void hdmi_stop_infoframe_trans(struct hda_codec *codec)
+{
+	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+						AC_DIPXMIT_DISABLE);
 }
 
 static int hdmi_get_channel_count(struct hda_codec *codec)
@@ -368,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
 					struct hdmi_audio_infoframe *ai)
 {
 	u8 *params = (u8 *)ai;
+	u8 sum = 0;
 	int i;
 
 	hdmi_debug_dip_size(codec);
 	hdmi_clear_dip_buffers(codec); /* be paranoid */
 
+	for (i = 0; i < sizeof(ai); i++)
+		sum += params[i];
+	ai->checksum = - sum;
+
 	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
 	for (i = 0; i < sizeof(ai); i++)
 		hdmi_write_dip_byte(codec, PIN_NID, params[i]);
@@ -419,14 +422,18 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
 	/*
 	 * CA defaults to 0 for basic stereo audio
 	 */
-	if (!eld->eld_ver)
-		return 0;
-	if (!eld->spk_alloc)
-		return 0;
 	if (channels <= 2)
 		return 0;
 
 	/*
+	 * HDMI sink's ELD info cannot always be retrieved for now, e.g.
+	 * in console or for audio devices. Assume the highest speakers
+	 * configuration, to _not_ prohibit multi-channel audio playback.
+	 */
+	if (!eld->spk_alloc)
+		eld->spk_alloc = 0xffff;
+
+	/*
 	 * expand ELD's speaker allocation mask
 	 *
 	 * ELD tells the speaker mask in a compact(paired) form,
@@ -485,6 +492,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 	hdmi_setup_channel_mapping(codec, &ai);
 
 	hdmi_fill_audio_infoframe(codec, &ai);
+	hdmi_start_infoframe_trans(codec);
 }
 
 
@@ -562,7 +570,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo,
 {
 	struct intel_hdmi_spec *spec = codec->spec;
 
-	hdmi_disable_output(codec);
+	hdmi_stop_infoframe_trans(codec);
 
 	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
@@ -582,8 +590,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 
 	hdmi_setup_audio_infoframe(codec, substream);
 
-	hdmi_enable_output(codec);
-
 	return 0;
 }
 
@@ -628,8 +634,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec)
 
 static int intel_hdmi_init(struct hda_codec *codec)
 {
-	/* disable audio output as early as possible */
-	hdmi_disable_output(codec);
+	hdmi_enable_output(codec);
 
 	snd_hda_sequence_write(codec, unsolicited_response_verb);
 
@@ -679,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
 	{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
 	{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
 	{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
+	{ .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
 	{ .id = 0x10951392, .name = "SiI1392 HDMI",     .patch = patch_intel_hdmi },
 	{} /* terminator */
 };
@@ -687,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb");
 MODULE_ALIAS("snd-hda-codec-id:80862801");
 MODULE_ALIAS("snd-hda-codec-id:80862802");
 MODULE_ALIAS("snd-hda-codec-id:80862803");
+MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:10951392");
 
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ae5c8a0d147..6c26afcb826 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7017,6 +7017,7 @@ static int patch_alc882(struct hda_codec *codec)
 		case 0x106b3e00: /* iMac 24 Aluminium */
 			board_config = ALC885_IMAC24;
 			break;
+		case 0x106b00a0: /* MacBookPro3,1 - Another revision */
 		case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
 		case 0x106b00a4: /* MacbookPro4,1 */
 		case 0x106b2c00: /* Macbook Pro rev3 */
@@ -8469,6 +8470,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
+		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
@@ -8478,6 +8481,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
@@ -10553,6 +10557,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
 	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
 	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 38428e22428..6094344fb22 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1207,7 +1207,7 @@ static const char *slave_vols[] = {
 	"LFE Playback Volume",
 	"Side Playback Volume",
 	"Headphone Playback Volume",
-	"Headphone Playback Volume",
+	"Headphone2 Playback Volume",
 	"Speaker Playback Volume",
 	"External Speaker Playback Volume",
 	"Speaker2 Playback Volume",
@@ -1221,7 +1221,7 @@ static const char *slave_sws[] = {
 	"LFE Playback Switch",
 	"Side Playback Switch",
 	"Headphone Playback Switch",
-	"Headphone Playback Switch",
+	"Headphone2 Playback Switch",
 	"Speaker Playback Switch",
 	"External Speaker Playback Switch",
 	"Speaker2 Playback Switch",
@@ -1799,7 +1799,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
 		      "HP dv5", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
-		      "HP dv7", STAC_HP_M4),
+		      "HP dv7", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
 		      "HP dv4", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
@@ -2442,6 +2442,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     stream_tag, format, substream);
 }
 
+static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 
 /*
  * Analog capture callbacks
@@ -2486,7 +2494,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
 	.ops = {
 		.open = stac92xx_dig_playback_pcm_open,
 		.close = stac92xx_dig_playback_pcm_close,
-		.prepare = stac92xx_dig_playback_pcm_prepare
+		.prepare = stac92xx_dig_playback_pcm_prepare,
+		.cleanup = stac92xx_dig_playback_pcm_cleanup
 	},
 };
 
@@ -3507,6 +3516,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (! spec->autocfg.line_outs)
 		return 0; /* can't find valid pin config */
 
+#if 0 /* FIXME: temporarily disabled */
 	/* If we have no real line-out pin and multiple hp-outs, HPs should
 	 * be set up as multi-channel outputs.
 	 */
@@ -3526,6 +3536,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
 		spec->autocfg.hp_outs = 0;
 	}
+#endif /* FIXME: temporarily disabled */
 	if (spec->autocfg.mono_out_pin) {
 		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
 			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
@@ -4980,7 +4991,7 @@ again:
 	case STAC_DELL_M4_3:
 		spec->num_dmics = 1;
 		spec->num_smuxes = 0;
-		spec->num_dmuxes = 0;
+		spec->num_dmuxes = 1;
 		break;
 	default:
 		spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 18c7c91786b..6c870c12a17 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -26,7 +26,7 @@
  * SPI 0 -> 1st PCM1796 (front)
  * SPI 1 -> 2nd PCM1796 (surround)
  * SPI 2 -> 3rd PCM1796 (center/LFE)
- * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!)
+ * SPI 4 -> 4th PCM1796 (back)
  *
  * GPIO 2 -> M0 of CS5381
  * GPIO 3 -> M1 of CS5381
@@ -207,12 +207,6 @@ static void xonar_gpio_changed(struct oxygen *chip);
 static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
 				     u8 reg, u8 value)
 {
-	/*
-	 * We don't want to do writes on SPI 4 because the EEPROM, which shares
-	 * the same pin, might get confused and broken.  We'd better take care
-	 * that the driver works with the default register values ...
-	 */
-#if 0
 	/* maps ALSA channel pair number to SPI output */
 	static const u8 codec_map[4] = {
 		0, 1, 2, 4
@@ -223,7 +217,6 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
 			 (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
 			 OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
 			 (reg << 8) | value);
-#endif
 }
 
 static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec,
@@ -757,9 +750,6 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
 
 static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 {
-	if (!strncmp(template->name, "Master Playback ", 16))
-		/* disable volume/mute because they would require SPI writes */
-		return 1;
 	if (!strncmp(template->name, "CD Capture ", 11))
 		/* CD in is actually connected to the video in pin */
 		template->private_value ^= AC97_CD ^ AC97_VIDEO;
@@ -850,8 +840,9 @@ static const struct oxygen_model model_xonar_d2 = {
 	.dac_volume_min = 0x0f,
 	.dac_volume_max = 0xff,
 	.misc_flags = OXYGEN_MISC_MIDI,
-	.function_flags = OXYGEN_FUNCTION_SPI,
-	.dac_i2s_format = OXYGEN_I2S_FORMAT_I2S,
+	.function_flags = OXYGEN_FUNCTION_SPI |
+			  OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c9..69d87dee699 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -97,12 +97,12 @@ struct pcxhr_mgr {
 	int capture_chips;
 	int fw_file_set;
 	int firmware_num;
-	int is_hr_stereo:1;
-	int board_has_aes1:1;	/* if 1 board has AES1 plug and SRC */
-	int board_has_analog:1;	/* if 0 the board is digital only */
-	int board_has_mic:1;	/* if 1 the board has microphone input */
-	int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
-	int mono_capture:1;	/* if 1 the board does mono capture */
+	unsigned int is_hr_stereo:1;
+	unsigned int board_has_aes1:1;	/* if 1 board has AES1 plug and SRC */
+	unsigned int board_has_analog:1; /* if 0 the board is digital only */
+	unsigned int board_has_mic:1; /* if 1 the board has microphone input */
+	unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
+	unsigned int mono_capture:1; /* if 1 the board does mono capture */
 
 	struct snd_dma_buffer hostport;
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b47a749c5ea..aea0cb72d80 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value & 0xff;
-	int shift = (kcontrol->private_value >> 8) & 0x0f;
-	int mask = (kcontrol->private_value >> 16) & 0xff;
-	int invert = (kcontrol->private_value >> 24) & 0x01;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
 	unsigned short val, val_mask;
 	int ret;
 	struct snd_soc_dapm_path *path;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 1cbb7b9b51c..a5731faa150 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -176,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value & 0xff;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int reg = mc->reg;
 	int ret;
 	u16 val;
 
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index ad97836818b..e226fa75669 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -91,7 +91,7 @@ static struct snd_soc_dai_link sdp3430_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+static struct snd_soc_card snd_soc_sdp3430 = {
 	.name = "SDP3430",
 	.platform = &omap_soc_platform,
 	.dai_link = &sdp3430_dai,
@@ -100,7 +100,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = {
 
 /* Audio subsystem */
 static struct snd_soc_device sdp3430_snd_devdata = {
-	.machine = &snd_soc_machine_sdp3430,
+	.card = &snd_soc_sdp3430,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 55fdb4abb17..ec3f8bb4b51 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 
 	mutex_lock(&codec->mutex);
 #ifdef CONFIG_SND_SOC_AC97_BUS
-	if (ac97) {
+	/* Only instantiate AC97 if not already done by the adaptor
+	 * for the generic AC97 subsystem.
+	 */
+	if (ac97 && strcmp(codec->name, "AC97") != 0) {
 		ret = soc_ac97_dev_register(codec);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: AC97 device register failed\n");
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 2ab83129d9b..19e37451c21 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
 		 * build the rate table and bitmap flags
 		 */
 		int r, idx;
-		unsigned int nonzero_rates = 0;
 
 		fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
 		if (fp->rate_table == NULL) {
@@ -2532,24 +2531,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
 			return -1;
 		}
 
-		fp->nr_rates = nr_rates;
-		fp->rate_min = fp->rate_max = combine_triple(&fmt[8]);
+		fp->nr_rates = 0;
+		fp->rate_min = fp->rate_max = 0;
 		for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
 			unsigned int rate = combine_triple(&fmt[idx]);
+			if (!rate)
+				continue;
 			/* C-Media CM6501 mislabels its 96 kHz altsetting */
 			if (rate == 48000 && nr_rates == 1 &&
-			    chip->usb_id == USB_ID(0x0d8c, 0x0201) &&
+			    (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
+			     chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
 			    fp->altsetting == 5 && fp->maxpacksize == 392)
 				rate = 96000;
-			fp->rate_table[r] = rate;
-			nonzero_rates |= rate;
-			if (rate < fp->rate_min)
+			fp->rate_table[fp->nr_rates] = rate;
+			if (!fp->rate_min || rate < fp->rate_min)
 				fp->rate_min = rate;
-			else if (rate > fp->rate_max)
+			if (!fp->rate_max || rate > fp->rate_max)
 				fp->rate_max = rate;
 			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+			fp->nr_rates++;
 		}
-		if (!nonzero_rates) {
+		if (!fp->nr_rates) {
 			hwc_debug("All rates were zero. Skipping format!\n");
 			return -1;
 		}
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 320641ab5be..26bad373fe6 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
 	}
 
 	ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+	ep_info.out_interval = 0;
 	ep_info.out_cables = endpoint->out_cables & 0x5555;
 	err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
 	if (err < 0)