Merge branch 'master' into for-2.6.33

38 files changed, 362 insertions, 134 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1f0f8213e2d..6c160a038b2 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -504,6 +504,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
 	int err;
 
 	aaci_pcm_hw_free(substream);
+	if (aacirun->pcm_open) {
+		snd_ac97_pcm_close(aacirun->pcm);
+		aacirun->pcm_open = 0;
+	}
 
 	err = devdma_hw_alloc(NULL, substream,
 			      params_buffer_bytes(params));
@@ -517,7 +521,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
 	else
 		err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
 					params_channels(params),
-					aacirun->pcm->r[1].slots);
+					aacirun->pcm->r[0].slots);
 
 	if (err)
 		goto out;
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 0c1440121c2..c69c60b2a48 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -953,11 +953,12 @@ static int snd_pcm_dev_register(struct snd_device *device)
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_notify *notify;
 	char str[16];
-	struct snd_pcm *pcm = device->device_data;
+	struct snd_pcm *pcm;
 	struct device *dev;
 
-	if (snd_BUG_ON(!pcm || !device))
+	if (snd_BUG_ON(!device || !device->device_data))
 		return -ENXIO;
+	pcm = device->device_data;
 	mutex_lock(&register_mutex);
 	err = snd_pcm_add(pcm);
 	if (err) {
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index c0adc14c91f..70d6f25ba52 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 6ba066c41d2..252e04ce602 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);
@@ -808,8 +808,6 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy)
 	unsigned int idx;
 	int err;
 
-	if (snd_BUG_ON(!dummy))
-		return -EINVAL;
 	spin_lock_init(&dummy->mixer_lock);
 	strcpy(card->mixername, "Dummy Mixer");
 
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 84cc2658c05..e1145ac6e90 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -39,25 +39,20 @@ static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
 /* write the port and returns the next expire time in ns;
  * called at the trigger-start and in hrtimer callback
  */
-static unsigned long pcsp_timer_update(struct hrtimer *handle)
+static u64 pcsp_timer_update(struct snd_pcsp *chip)
 {
 	unsigned char timer_cnt, val;
 	u64 ns;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_runtime *runtime;
-	struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
 	unsigned long flags;
 
 	if (chip->thalf) {
 		outb(chip->val61, 0x61);
 		chip->thalf = 0;
-		if (!atomic_read(&chip->timer_active))
-			return 0;
 		return chip->ns_rem;
 	}
 
-	if (!atomic_read(&chip->timer_active))
-		return 0;
 	substream = chip->playback_substream;
 	if (!substream)
 		return 0;
@@ -88,24 +83,17 @@ static unsigned long pcsp_timer_update(struct hrtimer *handle)
 	return ns;
 }
 
-enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+static void pcsp_pointer_update(struct snd_pcsp *chip)
 {
-	struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
 	struct snd_pcm_substream *substream;
-	int periods_elapsed, pointer_update;
 	size_t period_bytes, buffer_bytes;
-	unsigned long ns;
+	int periods_elapsed;
 	unsigned long flags;
 
-	pointer_update = !chip->thalf;
-	ns = pcsp_timer_update(handle);
-	if (!ns)
-		return HRTIMER_NORESTART;
-
 	/* update the playback position */
 	substream = chip->playback_substream;
 	if (!substream)
-		return HRTIMER_NORESTART;
+		return;
 
 	period_bytes = snd_pcm_lib_period_bytes(substream);
 	buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
@@ -134,6 +122,26 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 
 	if (periods_elapsed)
 		tasklet_schedule(&pcsp_pcm_tasklet);
+}
+
+enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+{
+	struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
+	int pointer_update;
+	u64 ns;
+
+	if (!atomic_read(&chip->timer_active) || !chip->playback_substream)
+		return HRTIMER_NORESTART;
+
+	pointer_update = !chip->thalf;
+	ns = pcsp_timer_update(chip);
+	if (!ns) {
+		printk(KERN_WARNING "PCSP: unexpected stop\n");
+		return HRTIMER_NORESTART;
+	}
+
+	if (pointer_update)
+		pcsp_pointer_update(chip);
 
 	hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
 
@@ -142,8 +150,6 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 
 static int pcsp_start_playing(struct snd_pcsp *chip)
 {
-	unsigned long ns;
-
 #if PCSP_DEBUG
 	printk(KERN_INFO "PCSP: start_playing called\n");
 #endif
@@ -159,11 +165,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip)
 	atomic_set(&chip->timer_active, 1);
 	chip->thalf = 0;
 
-	ns = pcsp_timer_update(&pcsp_chip.timer);
-	if (!ns)
-		return -EIO;
-
-	hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL);
+	hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
 	return 0;
 }
 
@@ -232,21 +234,22 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream)
 static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+	pcsp_sync_stop(chip);
+	chip->playback_ptr = 0;
+	chip->period_ptr = 0;
+	chip->fmt_size =
+		snd_pcm_format_physical_width(substream->runtime->format) >> 3;
+	chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
 #if PCSP_DEBUG
 	printk(KERN_INFO "PCSP: prepare called, "
-			"size=%zi psize=%zi f=%zi f1=%i\n",
+			"size=%zi psize=%zi f=%zi f1=%i fsize=%i\n",
 			snd_pcm_lib_buffer_bytes(substream),
 			snd_pcm_lib_period_bytes(substream),
 			snd_pcm_lib_buffer_bytes(substream) /
 			snd_pcm_lib_period_bytes(substream),
-			substream->runtime->periods);
+			substream->runtime->periods,
+			chip->fmt_size);
 #endif
-	pcsp_sync_stop(chip);
-	chip->playback_ptr = 0;
-	chip->period_ptr = 0;
-	chip->fmt_size =
-		snd_pcm_format_physical_width(substream->runtime->format) >> 3;
-	chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
 	return 0;
 }
 
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 199b0337714..903bc846763 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -72,7 +72,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol,
 	if (treble != chip->treble) {
 		chip->treble = treble;
 #if PCSP_DEBUG
-		printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE());
+		printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE());
 #endif
 		changed = 1;
 	}
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index 793b7f47843..3f3c3f71db4 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/hex2hex.c b/sound/oss/hex2hex.c
index 5460faae98c..041ef5c52bc 100644
--- a/sound/oss/hex2hex.c
+++ b/sound/oss/hex2hex.c
@@ -12,7 +12,7 @@
 #define MAX_SIZE (256*1024)
 unsigned char buf[MAX_SIZE];
 
-int loadhex(FILE *inf, unsigned char *buf)
+static int loadhex(FILE *inf, unsigned char *buf)
 {
 	int l=0, c, i;
 
diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c
index 77d0e5efda7..ce4db49291f 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 180e95c87e3..51a3d381a59 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/parisc/harmony.c b/sound/parisc/harmony.c
index e924492df21..f47f9e226b0 100644
--- a/sound/parisc/harmony.c
+++ b/sound/parisc/harmony.c
@@ -624,6 +624,9 @@ snd_harmony_pcm_init(struct snd_harmony *h)
 	struct snd_pcm *pcm;
 	int err;
 
+	if (snd_BUG_ON(!h))
+		return -EINVAL;
+
 	harmony_disable_interrupts(h);
 	
    	err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm);
@@ -865,11 +868,12 @@ snd_harmony_mixer_reset(struct snd_harmony *h)
 static int __devinit
 snd_harmony_mixer_init(struct snd_harmony *h)
 {
-	struct snd_card *card = h->card;
+	struct snd_card *card;
 	int idx, err;
 
 	if (snd_BUG_ON(!h))
 		return -EINVAL;
+	card = h->card;
 	strcpy(card->mixername, "Harmony Gain control interface");
 
 	for (idx = 0; idx < HARMONY_CONTROLS; idx++) {
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index fb5ee3cc396..75c602b5b13 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -259,7 +259,6 @@ config SND_CS5530
 
 config SND_CS5535AUDIO
 	tristate "CS5535/CS5536 Audio"
-	depends on X86 && !X86_64
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index b458d208720..aaf4da68969 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec,
 	void *private_data;
 
 	snd_ali_printk("free_voice: channel=%d\n",pvoice->number);
-	if (pvoice == NULL || !pvoice->use)
+	if (!pvoice->use)
 		return;
 	snd_ali_clear_voices(codec, pvoice->number, pvoice->number);
 	spin_lock_irq(&codec->voice_alloc);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c9ad182e1b4..6517f589d01 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 3fbbc8c01e7..905859d4f4d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -110,6 +110,7 @@ struct conexant_spec {
 
 	unsigned int dell_automute;
 	unsigned int port_d_mode;
+	unsigned char ext_mic_bias;
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
 static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
 #define CXT5066_SPDIF_OUT	0x21
 
+/* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal with
+ * the DC input range of the codec. */
+#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
+
 static struct hda_channel_mode cxt5066_modes[1] = {
 	{ 2, NULL },
 };
@@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5066_automic(struct hda_codec *codec)
 {
-	static struct hda_verb ext_mic_present[] = {
+	struct conexant_spec *spec = codec->spec;
+	struct hda_verb ext_mic_present[] = {
 		/* enable external mic, port B */
-		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
 
 		/* switch to external mic input */
 		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
 	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
 
 	/* Port B: external microphone */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
 
 	/* Port C: internal microphone */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2325,6 +2332,7 @@ 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),
 	{}
 };
 
@@ -2352,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->input_mux = &cxt5066_capture_source;
 
 	spec->port_d_mode = PIN_HP;
+	spec->ext_mic_bias = PIN_VREF80;
 
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2383,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->port_d_mode = 0;
+		spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 9fb60276f5c..6afdab09bab 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -397,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
 static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 	{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+	{ .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -406,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 
 MODULE_ALIAS("snd-hda-codec-id:10de0002");
 MODULE_ALIAS("snd-hda-codec-id:10de0003");
+MODULE_ALIAS("snd-hda-codec-id:10de0005");
 MODULE_ALIAS("snd-hda-codec-id:10de0006");
 MODULE_ALIAS("snd-hda-codec-id:10de0007");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c08ca660dab..70583719282 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);
@@ -4682,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
 		}
 	}
 	if (spec->autocfg.dig_in_pin)
@@ -6247,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
 	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
 	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
@@ -8909,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
 	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
-	/* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
-	 * no perfect solution yet
+	/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+	 * so apparently no perfect solution yet
 	 */
 	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
 	{} /* terminator */
 };
 
@@ -9811,9 +9814,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
 		}
 	}
 	if (spec->autocfg.dig_in_pin)
@@ -11458,6 +11461,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
 	SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+	SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+	SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
 			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
@@ -12602,7 +12607,8 @@ 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 +17380,7 @@ 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 66c0876bf73..86de305fc9f 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>
@@ -1589,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 17", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
 				"Dell Studio 1555", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
+				"Dell Studio 1557", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
@@ -1693,6 +1696,8 @@ 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 +4670,26 @@ 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 +4779,11 @@ 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 +5273,7 @@ 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 +5457,45 @@ 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/ice1712.h b/sound/pci/ice1712/ice1712.h
index 9da2dae64c5..d063149e704 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -382,8 +382,8 @@ struct snd_ice1712 {
 #ifdef CONFIG_PM
 	int (*pm_suspend)(struct snd_ice1712 *);
 	int (*pm_resume)(struct snd_ice1712 *);
-	int pm_suspend_enabled:1;
-	int pm_saved_is_spdif_master:1;
+	unsigned int pm_suspend_enabled:1;
+	unsigned int pm_saved_is_spdif_master:1;
 	unsigned int pm_saved_spdif_ctrl;
 	unsigned char pm_saved_spdif_cfg;
 	unsigned int pm_saved_route;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index c75515f5be6..6a9fee3ee78 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 754867ed478..aac20fb4aad 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/pci/via82xx.c b/sound/pci/via82xx.c
index 91683a34903..8a332d2f615 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -386,6 +386,7 @@ struct via82xx {
 
 	struct snd_pcm *pcms[2];
 	struct snd_rawmidi *rmidi;
+	struct snd_kcontrol *dxs_controls[4];
 
 	struct snd_ac97_bus *ac97_bus;
 	struct snd_ac97 *ac97;
@@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
 
 
 /*
- * open callback for playback on via686 and via823x DSX
+ * open callback for playback on via686
  */
-static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
+static int snd_via686_playback_open(struct snd_pcm_substream *substream)
 {
 	struct via82xx *chip = snd_pcm_substream_chip(substream);
 	struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number];
@@ -1230,6 +1231,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
 }
 
 /*
+ * open callback for playback on via823x DXS
+ */
+static int snd_via8233_playback_open(struct snd_pcm_substream *substream)
+{
+	struct via82xx *chip = snd_pcm_substream_chip(substream);
+	struct viadev *viadev;
+	unsigned int stream;
+	int err;
+
+	viadev = &chip->devs[chip->playback_devno + substream->number];
+	if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
+		return err;
+	stream = viadev->reg_offset / 0x10;
+	if (chip->dxs_controls[stream]) {
+		chip->playback_volume[stream][0] = 0;
+		chip->playback_volume[stream][1] = 0;
+		chip->dxs_controls[stream]->vd[0].access &=
+			~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+			       SNDRV_CTL_EVENT_MASK_INFO,
+			       &chip->dxs_controls[stream]->id);
+	}
+	return 0;
+}
+
+/*
  * open callback for playback on via823x multi-channel
  */
 static int snd_via8233_multi_open(struct snd_pcm_substream *substream)
@@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
+static int snd_via8233_playback_close(struct snd_pcm_substream *substream)
+{
+	struct via82xx *chip = snd_pcm_substream_chip(substream);
+	struct viadev *viadev = substream->runtime->private_data;
+	unsigned int stream;
+
+	stream = viadev->reg_offset / 0x10;
+	if (chip->dxs_controls[stream]) {
+		chip->dxs_controls[stream]->vd[0].access |=
+			SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO,
+			       &chip->dxs_controls[stream]->id);
+	}
+	return snd_via82xx_pcm_close(substream);
+}
+
 
 /* via686 playback callbacks */
 static struct snd_pcm_ops snd_via686_playback_ops = {
-	.open =		snd_via82xx_playback_open,
+	.open =		snd_via686_playback_open,
 	.close =	snd_via82xx_pcm_close,
 	.ioctl =	snd_pcm_lib_ioctl,
 	.hw_params =	snd_via82xx_hw_params,
@@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = {
 
 /* via823x DSX playback callbacks */
 static struct snd_pcm_ops snd_via8233_playback_ops = {
-	.open =		snd_via82xx_playback_open,
-	.close =	snd_via82xx_pcm_close,
+	.open =		snd_via8233_playback_open,
+	.close =	snd_via8233_playback_close,
 	.ioctl =	snd_pcm_lib_ioctl,
 	.hw_params =	snd_via82xx_hw_params,
 	.hw_free =	snd_via82xx_hw_free,
@@ -1709,8 +1752,9 @@ static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
 	.device = 0,
 	/* .subdevice set later */
 	.name = "PCM Playback Volume",
-	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
-		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		  SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+		  SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.info = snd_via8233_dxs_volume_info,
 	.get = snd_via8233_dxs_volume_get,
 	.put = snd_via8233_dxs_volume_put,
@@ -1948,6 +1992,7 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
 				err = snd_ctl_add(chip->card, kctl);
 				if (err < 0)
 					return err;
+				chip->dxs_controls[i] = kctl;
 			}
 		}
 	}
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 7dea74b71cf..64b859925c0 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 7445cc8a47d..1492744ad67 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/sh/aica.c b/sound/sh/aica.c
index 583a3693df7..a0df401ebb9 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/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 0b8dcb5cd72..90a0264f753 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = {
 #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11)        | (1<<15))
 static const unsigned short sr_valid_mask[] = {
 	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 0*/
-	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
 	LOWER_GROUP,			/* Usb, bosr - 0*/
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
 	UPPER_GROUP,			/* Usb, bosr - 1*/
 };
 /*
@@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	int i;
 	u16 reg;
 
 	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+	for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee9839be8..653a362425d 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 5735945788b..6a829eef2a4 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/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 9114c263077..13aa380de16 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit);
 
 MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
 MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index ad219aaf7cb..0cd06f5dd35 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
  *  |P| <--- TWL4030 <--------- Line In and MICs
  */
 static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
-	SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
 			   0, 0, NULL, 0, omap3pandora_hp_event,
 			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "CARKITR");
 	snd_soc_dapm_nc_pin(codec, "HFL");
 	snd_soc_dapm_nc_pin(codec, "HFR");
+	snd_soc_dapm_nc_pin(codec, "VIBRA");
 
 	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 5cbbdc80fde..1f35c6fcf5f 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 3c06c401d0f..105a77eeded 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 7ff04ad2a97..0a1b2f64bbe 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 8de6f9dec4a..66d4c165f99 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 			if (!w->power_check)
 				continue;
 
-			power = w->power_check(w);
-			if (power)
-				sys_power = 1;
+			/* If we're suspending then pull down all the 
+			 * power. */
+			switch (event) {
+			case SND_SOC_DAPM_STREAM_SUSPEND:
+				power = 0;
+				break;
+
+			default:
+				power = w->power_check(w);
+				if (power)
+					sys_power = 1;
+				break;
+			}
 
 			if (w->power == power)
 				continue;
@@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 		case SND_SOC_DAPM_STREAM_RESUME:
 			sys_power = 1;
 			break;
+		case SND_SOC_DAPM_STREAM_SUSPEND:
+			sys_power = 0;
+			break;
 		case SND_SOC_DAPM_STREAM_NOP:
 			sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+			break;
 		default:
 			break;
 		}
@@ -2072,9 +2086,9 @@ 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/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 121af0644fd..86b2c3b92df 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -62,10 +62,14 @@ static void
 activate_substream(struct snd_usb_caiaqdev *dev,
 	           struct snd_pcm_substream *sub)
 {
+	spin_lock(&dev->spinlock);
+
 	if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		dev->sub_playback[sub->number] = sub;
 	else
 		dev->sub_capture[sub->number] = sub;
+
+	spin_unlock(&dev->spinlock);
 }
 
 static void
@@ -269,16 +273,22 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
 {
 	int index = sub->number;
 	struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub);
+	snd_pcm_uframes_t ptr;
+
+	spin_lock(&dev->spinlock);
 
 	if (dev->input_panic || dev->output_panic)
-		return SNDRV_PCM_POS_XRUN;
+		ptr = SNDRV_PCM_POS_XRUN;
 
 	if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return bytes_to_frames(sub->runtime,
+		ptr = bytes_to_frames(sub->runtime,
 					dev->audio_out_buf_pos[index]);
 	else
-		return bytes_to_frames(sub->runtime,
+		ptr = bytes_to_frames(sub->runtime,
 					dev->audio_in_buf_pos[index]);
+
+	spin_unlock(&dev->spinlock);
+	return ptr;
 }
 
 /* operators for both playback and capture */
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 83e6c1312d4..a3f02dd9744 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 8e7f78941ba..e9a3a9dca15 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))
 
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 9efcfd08d74..c998220b99c 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig
 	channels = (ftr[0] - 7) / csize - 1;
 
 	master_bits = snd_usb_combine_bytes(ftr + 6, csize);
+	/* master configuration quirks */
+	switch (state->chip->usb_id) {
+	case USB_ID(0x08bb, 0x2702):
+		snd_printk(KERN_INFO
+			   "usbmixer: master volume quirk for PCM2702 chip\n");
+		/* disable non-functional volume control */
+		master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1));
+		break;
+	}
 	if (channels > 0)
 		first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize);
 	else