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 0c1440121c22..c69c60b2a48a 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 6ba066c41d2e..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);
@@ -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 84cc2658c05b..e1145ac6e908 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 199b03377142..903bc846763f 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 e924492df21d..f47f9e226b08 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 fb5ee3cc3968..75c602b5b132 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 b458d208720b..aaf4da68969c 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 c9ad182e1b4b..e340792f6cb3 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 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/via82xx.c b/sound/pci/via82xx.c
index 91683a349035..8a332d2f615c 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 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/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/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/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 121af0644fd9..86b2c3b92df5 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 83e6c1312d47..a3f02dd97440 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},"