Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Don't check invalid HP pin
  ALSA: dummy - Fix descriptions of pcm_substreams parameter
  ALSA: pcmcia: use dynamic debug infrastructure, deprecate CS_CHECK (sound)
  ALSA: hda: Use quirk mask for Dell Inspiron Mini9/Vostro A90 using ALC268
  sound: via82xx: deactivate DXS controls of inactive streams
  ALSA: snd-usb-caiaq: Bump version number to 1.3.20
  ALSA: snd-usb-caiaq: Lock on stream start/unpause
  ALSA: snd-usb-caiaq: Missing lock around use of buffer positions
  ALSA: sound/parisc: Move dereference after NULL test
  ALSA: sound: Move dereference after NULL test and drop unnecessary NULL tests
  ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller
  ALSA: pcsp - Fix nforce workaround
  ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency
  ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text
  ASoC: Fix possible codec_dai->ops NULL pointer problems
  ALSA: hda - Fix capture source checks for ALC662/663 codecs
  ASoC: Serialize access to dapm_power_widgets()

17 files changed, 162 insertions, 76 deletions

diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 0c14401..c69c60b 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/drivers/dummy.c b/sound/drivers/dummy.c
index 6ba066c..252e04c 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 84cc265..e1145ac 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 199b033..903bc84 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/parisc/harmony.c b/sound/parisc/harmony.c
index e924492..f47f9e2 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 fb5ee3c..75c602b 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 b458d20..aaf4da6 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 c9ad182..e340792 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2674,6 +2674,7 @@ 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_realtek.c b/sound/pci/hda/patch_realtek.c
index c08ca66..ff200485 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/via82xx.c b/sound/pci/via82xx.c
index 91683a3..8a332d2 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 7dea74b..64b8599 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 7445cc8..1492744 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/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee983..653a362 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/soc-core.c b/sound/soc/soc-core.c
index 7ff04ad..0a1b2f6 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 8de6f9d..d89f6dc 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/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 121af06..86b2c3b 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 83e6c13..a3f02dd 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},"