Merge branch 'fix/misc' into topic/misc

22 files changed, 188 insertions, 63 deletions

diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index c0adc14c91f0..70d6f25ba526 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -248,7 +248,8 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice,
 	list_for_each_entry(substream, &s->substreams, list) {
 		if (substream->opened) {
 			if (stream == SNDRV_RAWMIDI_STREAM_INPUT ||
-			    !(mode & SNDRV_RAWMIDI_LFLG_APPEND))
+			    !(mode & SNDRV_RAWMIDI_LFLG_APPEND) ||
+			    !substream->append)
 				continue;
 		}
 		if (subdevice < 0 || subdevice == substream->number) {
@@ -266,17 +267,21 @@ static int open_substream(struct snd_rawmidi *rmidi,
 {
 	int err;
 
-	err = snd_rawmidi_runtime_create(substream);
-	if (err < 0)
-		return err;
-	err = substream->ops->open(substream);
-	if (err < 0)
-		return err;
-	substream->opened = 1;
-	if (substream->use_count++ == 0)
+	if (substream->use_count == 0) {
+		err = snd_rawmidi_runtime_create(substream);
+		if (err < 0)
+			return err;
+		err = substream->ops->open(substream);
+		if (err < 0) {
+			snd_rawmidi_runtime_free(substream);
+			return err;
+		}
+		substream->opened = 1;
 		substream->active_sensing = 0;
-	if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
-		substream->append = 1;
+		if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
+			substream->append = 1;
+	}
+	substream->use_count++;
 	rmidi->streams[substream->stream].substream_opened++;
 	return 0;
 }
@@ -297,27 +302,27 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode,
 				       SNDRV_RAWMIDI_STREAM_INPUT,
 				       mode, &sinput);
 		if (err < 0)
-			goto __error;
+			return err;
 	}
 	if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
 		err = assign_substream(rmidi, subdevice,
 				       SNDRV_RAWMIDI_STREAM_OUTPUT,
 				       mode, &soutput);
 		if (err < 0)
-			goto __error;
+			return err;
 	}
 
 	if (sinput) {
 		err = open_substream(rmidi, sinput, mode);
 		if (err < 0)
-			goto __error;
+			return err;
 	}
 	if (soutput) {
 		err = open_substream(rmidi, soutput, mode);
 		if (err < 0) {
 			if (sinput)
 				close_substream(rmidi, sinput, 0);
-			goto __error;
+			return err;
 		}
 	}
 
@@ -325,13 +330,6 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode,
 	rfile->input = sinput;
 	rfile->output = soutput;
 	return 0;
-
-      __error:
-	if (sinput && sinput->runtime)
-		snd_rawmidi_runtime_free(sinput);
-	if (soutput && soutput->runtime)
-		snd_rawmidi_runtime_free(soutput);
-	return err;
 }
 
 /* called from sound/core/seq/seq_midi.c */
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 146ef00f94a3..252e04ce602f 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -165,7 +165,7 @@ MODULE_PARM_DESC(enable, "Enable this dummy soundcard.");
 module_param_array(pcm_devs, int, NULL, 0444);
 MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver.");
 module_param_array(pcm_substreams, int, NULL, 0444);
-MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver.");
+MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver.");
 //module_param_array(midi_devs, int, NULL, 0444);
 //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver.");
 module_param(fake_buffer, bool, 0444);
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index 793b7f478433..3f3c3f71db4b 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -219,7 +219,9 @@ static int shared_resources_initialised;
      *  Mid level stuff
      */
 
-struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED };
+struct sound_settings dmasound = {
+	.lock = __SPIN_LOCK_UNLOCKED(dmasound.lock)
+};
 
 static inline void sound_silence(void)
 {
diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c
index 77d0e5efda76..ce4db49291f7 100644
--- a/sound/oss/sb_common.c
+++ b/sound/oss/sb_common.c
@@ -157,7 +157,7 @@ static void sb_intr (sb_devc *devc)
 				break;
 
 			default:
-				/* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */
+				/* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */
 				;
 		}
 	}
@@ -177,7 +177,7 @@ static void sb_intr (sb_devc *devc)
 				break;
 
 			default:
-				/* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */
+				/* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */
 				;
 		}
 	}
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c
index 180e95c87e3e..51a3d381a59e 100644
--- a/sound/oss/sb_ess.c
+++ b/sound/oss/sb_ess.c
@@ -782,7 +782,7 @@ printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode);
 			break;
 
 		default:;
-			/* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */
+			/* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */
 	}
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c9ad182e1b4b..6517f589d01d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -722,9 +722,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		   chip->last_cmd[addr]);
 	chip->single_cmd = 1;
 	bus->response_reset = 0;
-	/* re-initialize CORB/RIRB */
+	/* release CORB/RIRB */
 	azx_free_cmd_io(chip);
-	azx_init_cmd_io(chip);
+	/* disable unsolicited responses */
+	azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL);
 	return -1;
 }
 
@@ -865,7 +866,9 @@ static int azx_reset(struct azx *chip)
 	}
 
 	/* Accept unsolicited responses */
-	azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL);
+	if (!chip->single_cmd)
+		azx_writel(chip, GCTL, azx_readl(chip, GCTL) |
+			   ICH6_GCTL_UNSOL);
 
 	/* detect codecs */
 	if (!chip->codec_mask) {
@@ -980,7 +983,8 @@ static void azx_init_chip(struct azx *chip)
 	azx_int_enable(chip);
 
 	/* initialize the codec command I/O */
-	azx_init_cmd_io(chip);
+	if (!chip->single_cmd)
+		azx_init_cmd_io(chip);
 
 	/* program the position buffer */
 	azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
@@ -2674,6 +2678,7 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3fbbc8c01e70..6479e65858d3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2325,6 +2325,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 		      CXT5066_LAPTOP),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
 		      CXT5066_DELL_LAPTOP),
+	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c08ca660daba..ff20048504b6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -965,6 +965,8 @@ static void alc_automute_pin(struct hda_codec *codec)
 	unsigned int nid = spec->autocfg.hp_pins[0];
 	int i;
 
+	if (!nid)
+		return;
 	pincap = snd_hda_query_pin_caps(codec, nid);
 	if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
 		snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
@@ -12602,7 +12604,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
 						ALC268_ACER_ASPIRE_ONE),
 	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
-	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
+	SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
+			"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
 	/* almost compatible with toshiba but with optional digital outs;
 	 * auto-probing seems working fine
 	 */
@@ -17374,7 +17377,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 
 /* create playback/capture controls for input pins */
 #define alc662_auto_create_input_ctls \
-	alc880_auto_create_input_ctls
+	alc882_auto_create_input_ctls
 
 static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid, int pin_type,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 66c0876bf734..8eb6508cd991 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -28,6 +28,7 @@
 #include <linux/delay.h>
 #include <linux/slab.h>
 #include <linux/pci.h>
+#include <linux/dmi.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
 #include <sound/jack.h>
@@ -1693,6 +1694,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD71BXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
 		      "HP dv4-1222nr", STAC_HP_DV4_1222NR),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720,
+			  "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
 		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
@@ -4665,6 +4668,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
+static int hp_bseries_system(u32 subsystem_id)
+{
+	switch (subsystem_id) {
+	case 0x103c307e:
+	case 0x103c307f:
+	case 0x103c3080:
+	case 0x103c3081:
+	case 0x103c1722:
+	case 0x103c1723:
+	case 0x103c1724:
+	case 0x103c1725:
+	case 0x103c1726:
+	case 0x103c1727:
+	case 0x103c1728:
+	case 0x103c1729:
+		return 1;
+	}
+	return 0;
+}
+
 #ifdef CONFIG_PROC_FS
 static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
 			       struct hda_codec *codec, hda_nid_t nid)
@@ -4754,6 +4777,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 		else
 			spec->gpio_data |= spec->gpio_led; /* white */
 
+		if (hp_bseries_system(codec->subsystem_id)) {
+			/* LED state is inverted on these systems */
+			spec->gpio_data ^= spec->gpio_led;
+		}
+
 		stac_gpio_set(codec, spec->gpio_mask,
 			      spec->gpio_dir,
 			      spec->gpio_data);
@@ -5243,6 +5271,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
 	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
+	unsigned int pin_cfg;
 	int err = 0;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5426,6 +5455,45 @@ again:
 		break;
 	}
 
+	if (hp_bseries_system(codec->subsystem_id)) {
+		pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
+		if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+			get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER  ||
+			get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
+			/* It was changed in the BIOS to just satisfy MS DTM.
+			 * Lets turn it back into slaved HP
+			 */
+			pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
+					| (AC_JACK_HP_OUT <<
+						AC_DEFCFG_DEVICE_SHIFT);
+			pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
+							| AC_DEFCFG_SEQUENCE)))
+								| 0x1f;
+			snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
+		}
+	}
+
+	if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
+		const struct dmi_device *dev = NULL;
+		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
+					      NULL, dev))) {
+			if (strcmp(dev->name, "HP_Mute_LED_1")) {
+				switch (codec->vendor_id) {
+				case 0x111d7608:
+					spec->gpio_led = 0x01;
+					break;
+				case 0x111d7600:
+				case 0x111d7601:
+				case 0x111d7602:
+				case 0x111d7603:
+					spec->gpio_led = 0x08;
+					break;
+				}
+				break;
+			}
+		}
+	}
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (spec->gpio_led) {
 		spec->gpio_mask |= spec->gpio_led;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index c75515f5be6f..6a9fee3ee78f 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice)
 }
 
 #ifdef CONFIG_PM
-static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice)
+static int prodigy_hd2_resume(struct snd_ice1712 *ice)
 {
 	/* initialize ak4396 codec and restore previous mixer volumes */
 	struct prodigy_hifi_spec *spec = ice->spec;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 754867ed4785..aac20fb4aad2 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1950,6 +1950,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	},
 	{
 		.subvendor = 0x104d,
+		.subdevice = 0x8144,
+		.name = "Sony",
+		.type = AC97_TUNE_INV_EAPD
+	},
+	{
+		.subvendor = 0x104d,
 		.subdevice = 0x8197,
 		.name = "Sony S1XP",
 		.type = AC97_TUNE_INV_EAPD
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 7dea74b71cf1..64b859925c0b 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -217,20 +217,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link)
  * configuration callback
  */
 
-#define CS_CHECK(fn, ret) \
-do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0)
-
 static int pdacf_config(struct pcmcia_device *link)
 {
 	struct snd_pdacf *pdacf = link->priv;
-	int last_fn, last_ret;
+	int ret;
 
 	snd_printdd(KERN_DEBUG "pdacf_config called\n");
 	link->conf.ConfigIndex = 0x5;
 
-	CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io));
-	CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq));
-	CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf));
+	ret = pcmcia_request_io(link, &link->io);
+	if (ret)
+		goto failed;
+
+	ret = pcmcia_request_irq(link, &link->irq);
+	if (ret)
+		goto failed;
+
+	ret = pcmcia_request_configuration(link, &link->conf);
+	if (ret)
+		goto failed;
 
 	if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0)
 		goto failed;
@@ -238,8 +243,6 @@ static int pdacf_config(struct pcmcia_device *link)
 	link->dev_node = &pdacf->node;
 	return 0;
 
-cs_failed:
-	cs_error(link, last_fn, last_ret);
 failed:
 	pcmcia_disable_device(link);
 	return -ENODEV;
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 7445cc8a47d3..1492744ad67f 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -213,14 +213,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq
  * configuration callback
  */
 
-#define CS_CHECK(fn, ret) \
-do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0)
-
 static int vxpocket_config(struct pcmcia_device *link)
 {
 	struct vx_core *chip = link->priv;
 	struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip;
-	int last_fn, last_ret;
+	int ret;
 
 	snd_printdd(KERN_DEBUG "vxpocket_config called\n");
 
@@ -235,9 +232,17 @@ static int vxpocket_config(struct pcmcia_device *link)
 		strcpy(chip->card->driver, vxp440_hw.name);
 	}
 
-	CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io));
-	CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq));
-	CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf));
+	ret = pcmcia_request_io(link, &link->io);
+	if (ret)
+		goto failed;
+
+	ret = pcmcia_request_irq(link, &link->irq);
+	if (ret)
+		goto failed;
+
+	ret = pcmcia_request_configuration(link, &link->conf);
+	if (ret)
+		goto failed;
 
 	chip->dev = &handle_to_dev(link);
 	snd_card_set_dev(chip->card, chip->dev);
@@ -248,8 +253,6 @@ static int vxpocket_config(struct pcmcia_device *link)
 	link->dev_node = &vxp->node;
 	return 0;
 
-cs_failed:
-	cs_error(link, last_fn, last_ret);
 failed:
 	pcmcia_disable_device(link);
 	return -ENODEV;
diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig
index bd2338ab2ced..0519c60f5be1 100644
--- a/sound/ppc/Kconfig
+++ b/sound/ppc/Kconfig
@@ -2,7 +2,7 @@
 
 menuconfig SND_PPC
 	bool "PowerPC sound devices"
-	depends on PPC64 || PPC32
+	depends on PPC
 	default y
 	help
 	  Support for sound devices specific to PowerPC architectures.
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 583a3693df75..a0df401ebb9f 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -49,6 +49,7 @@ MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
 MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}");
+MODULE_FIRMWARE("aica_firmware.bin");
 
 /* module parameters */
 #define CARD_NAME "AICA"
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee9839be86..653a362425df 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_CX20442
 	help
-	  Say Y if you want to add support for SoC audio on Amstrad Delta.
+	  Say Y  if you want to add support  for SoC audio device  connected to
+	  a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
+
+	  Note that in order to get those devices fully supported,  you have to
+	  build  the kernel  with  standard  serial port  driver  included  and
+	  configured for at least 4 ports.  Then, from userspace, you must load
+	  a line discipline #19 on the modem (ttyS3) serial line.  The simplest
+	  way to achieve this is to install util-linux-ng  and use the included
+	  ldattach  utility.  This  can be  started  automatically  from  udev,
+	  a simple rule like this one should do the trick (it does for me):
+	  	ACTION=="add", KERNEL=="controlC0", \
+				RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
 
 config SND_OMAP_SOC_OSK5912
 	tristate "SoC Audio support for omap osk5912"
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5735945788bf..6a829eef2a4f 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	else
 		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
 
-	omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
-	omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+	if (!(cpu_class_is_omap1())) {
+		omap_set_dma_src_burst_mode(prtd->dma_ch,
+						OMAP_DMA_DATA_BURST_16);
+		omap_set_dma_dest_burst_mode(prtd->dma_ch,
+						OMAP_DMA_DATA_BURST_16);
+	}
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 5cbbdc80fde3..1f35c6fcf5fd 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	dma_addr_t pos = prtd->dma_pos;
+	unsigned int limit;
 	int ret;
 
 	pr_debug("Entered %s\n", __func__);
 
-	while (prtd->dma_loaded < prtd->dma_limit) {
+	if (s3c_dma_has_circular()) {
+		limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period;
+	} else
+		limit = prtd->dma_limit;
+
+	pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit);
+
+	while (prtd->dma_loaded < limit) {
 		unsigned long len = prtd->dma_period;
 
 		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
@@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
 		snd_pcm_period_elapsed(substream);
 
 	spin_lock(&prtd->lock);
-	if (prtd->state & ST_RUNNING) {
+	if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) {
 		prtd->dma_loaded--;
 		s3c24xx_pcm_enqueue(substream);
 	}
@@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 			printk(KERN_ERR "failed to get dma channel\n");
 			return ret;
 		}
+
+		/* use the circular buffering if we have it available. */
+		if (s3c_dma_has_circular())
+			s3c2410_dma_setflags(prtd->params->channel,
+					     S3C2410_DMAF_CIRCULAR);
 	}
 
 	s3c2410_dma_set_buffdone_fn(prtd->params->channel,
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 3c06c401d0fb..105a77eeded0 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
 		goto err;
 	}
 
+	clk_enable(i2s->iis_cclk);
+
 	ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
 	if (ret)
 		goto err_clk;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ff04ad2a97e..0a1b2f64bbee 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device);
 #define soc_resume	NULL
 #endif
 
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
 static void snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct platform_device *pdev = container_of(card->dev,
@@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 			ac97 = 1;
 	}
 
+	for (i = 0; i < card->num_links; i++) {
+		if (!card->dai_link[i].codec_dai->ops)
+			card->dai_link[i].codec_dai->ops = &null_dai_ops;
+	}
+
 	/* If we have AC97 in the system then don't wait for the
 	 * codec.  This will need revisiting if we have to handle
 	 * systems with mixed AC97 and non-AC97 parts.  Only check for
@@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 	return 0;
 }
 
-static struct snd_soc_dai_ops null_dai_ops = {
-};
-
 /**
  * snd_soc_register_dai - Register a DAI with the ASoC core
  *
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8de6f9dec4a2..d89f6dc00908 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2072,9 +2072,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 			}
 		}
 	}
-	mutex_unlock(&codec->mutex);
 
 	dapm_power_widgets(codec, event);
+	mutex_unlock(&codec->mutex);
 	dump_dapm(codec, __func__);
 	return 0;
 }
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 8e7f78941ba6..e9a3a9dca15c 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -210,7 +210,7 @@ struct snd_usb_midi_endpoint_info {
 /*
  */
 
-#define combine_word(s)    ((*s) | ((unsigned int)(s)[1] << 8))
+#define combine_word(s)    ((*(s)) | ((unsigned int)(s)[1] << 8))
 #define combine_triple(s)  (combine_word(s) | ((unsigned int)(s)[2] << 16))
 #define combine_quad(s)    (combine_triple(s) | ((unsigned int)(s)[3] << 24))