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authorLinus Torvalds <torvalds@linux-foundation.org>2009-11-02 09:50:22 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2009-11-02 09:50:22 -0800
commit20107f84b29536887b958e38b20474cccc619322 (patch)
tree457a84cafe89bf63ba243edb3f8074955c478838
parent3fe866ca6cc0495510a1d37c03a4abdc55174e83 (diff)
parent8fd6959de180f727b28813741468c9cb2a7028ec (diff)
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()
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt2
-rw-r--r--sound/core/pcm.c5
-rw-r--r--sound/drivers/dummy.c4
-rw-r--r--sound/drivers/pcsp/pcsp_lib.c65
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c2
-rw-r--r--sound/parisc/harmony.c6
-rw-r--r--sound/pci/Kconfig1
-rw-r--r--sound/pci/ali5451/ali5451.c2
-rw-r--r--sound/pci/hda/hda_intel.c1
-rw-r--r--sound/pci/hda/patch_realtek.c7
-rw-r--r--sound/pci/via82xx.c59
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.c21
-rw-r--r--sound/pcmcia/vx/vxpocket.c21
-rw-r--r--sound/soc/omap/Kconfig13
-rw-r--r--sound/soc/soc-core.c11
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/usb/caiaq/audio.c16
-rw-r--r--sound/usb/caiaq/device.c2
18 files changed, 163 insertions, 77 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 1c8eb4518ce0..fd9a2f67edf2 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -522,7 +522,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
pcm_devs - Number of PCM devices assigned to each card
(default = 1, up to 4)
pcm_substreams - Number of PCM substreams assigned to each PCM
- (default = 8, up to 16)
+ (default = 8, up to 128)
hrtimer - Use hrtimer (=1, default) or system timer (=0)
fake_buffer - Fake buffer allocations (default = 1)
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},"