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-rw-r--r--sound/pci/hda/hda_intel.c1
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_realtek.c26
-rw-r--r--sound/pci/hda/patch_sigmatel.c22
-rw-r--r--sound/pci/hda/patch_via.c76
-rw-r--r--sound/pci/sis7019.c64
-rw-r--r--sound/soc/atmel/Kconfig21
-rw-r--r--sound/soc/atmel/Makefile4
-rw-r--r--sound/soc/atmel/playpaq_wm8510.c473
-rw-r--r--sound/soc/codecs/ad1836.h2
-rw-r--r--sound/soc/codecs/cs4270.c10
-rw-r--r--sound/soc/codecs/cs42l51.c2
-rw-r--r--sound/soc/codecs/max9877.c10
-rw-r--r--sound/soc/codecs/wm8994.c7
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c24
-rw-r--r--sound/soc/samsung/smdk_wm8994.c1
-rw-r--r--sound/soc/samsung/speyside.c2
-rw-r--r--sound/soc/soc-core.c6
-rw-r--r--sound/usb/quirks-table.h31
19 files changed, 185 insertions, 599 deletions
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 096507d2ca9a..7d98240def0b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 7bd2a52f2bac..70a7abda7e22 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1278,7 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = {
[CS420X_MBP53] = "mbp53",
[CS420X_MBP55] = "mbp55",
[CS420X_IMAC27] = "imac27",
- [CS420X_IMAC27] = "apple",
+ [CS420X_APPLE] = "apple",
[CS420X_AUTO] = "auto",
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 06c0c12d4fec..a7d1bc4e0d09 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -277,6 +277,12 @@ static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur)
return false;
}
+static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx)
+{
+ return spec->capsrc_nids ?
+ spec->capsrc_nids[idx] : spec->adc_nids[idx];
+}
+
/* select the given imux item; either unmute exclusively or select the route */
static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
unsigned int idx, bool force)
@@ -291,6 +297,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
imux = &spec->input_mux[mux_idx];
if (!imux->num_items && mux_idx > 0)
imux = &spec->input_mux[0];
+ if (!imux->num_items)
+ return 0;
if (idx >= imux->num_items)
idx = imux->num_items - 1;
@@ -303,8 +311,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
adc_idx = spec->dyn_adc_idx[idx];
}
- nid = spec->capsrc_nids ?
- spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
+ nid = get_capsrc(spec, adc_idx);
/* no selection? */
num_conns = snd_hda_get_conn_list(codec, nid, NULL);
@@ -1058,8 +1065,7 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec)
hda_nid_t pin = spec->imux_pins[i];
int c;
for (c = 0; c < spec->num_adc_nids; c++) {
- hda_nid_t cap = spec->capsrc_nids ?
- spec->capsrc_nids[c] : spec->adc_nids[c];
+ hda_nid_t cap = get_capsrc(spec, c);
int idx = get_connection_index(codec, cap, pin);
if (idx >= 0) {
imux->items[i].index = idx;
@@ -1969,10 +1975,8 @@ static int alc_build_controls(struct hda_codec *codec)
if (!kctl)
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
- const hda_nid_t *nids = spec->capsrc_nids;
- if (!nids)
- nids = spec->adc_nids;
- err = snd_hda_add_nid(codec, kctl, i, nids[i]);
+ err = snd_hda_add_nid(codec, kctl, i,
+ get_capsrc(spec, i));
if (err < 0)
return err;
}
@@ -2759,8 +2763,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec)
}
for (c = 0; c < num_adcs; c++) {
- hda_nid_t cap = spec->capsrc_nids ?
- spec->capsrc_nids[c] : spec->adc_nids[c];
+ hda_nid_t cap = get_capsrc(spec, c);
idx = get_connection_index(codec, cap, pin);
if (idx >= 0) {
spec->imux_pins[imux->num_items] = pin;
@@ -3706,8 +3709,7 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
if (!pin)
return 0;
for (i = 0; i < spec->num_adc_nids; i++) {
- hda_nid_t cap = spec->capsrc_nids ?
- spec->capsrc_nids[i] : spec->adc_nids[i];
+ hda_nid_t cap = get_capsrc(spec, i);
int idx;
idx = get_connection_index(codec, cap, pin);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e035cf6de278..ea1f157ca38b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4440,7 +4440,9 @@ static int stac92xx_init(struct hda_codec *codec)
int pinctl, def_conf;
/* power on when no jack detection is available */
- if (!spec->hp_detect) {
+ /* or when the VREF is used for controlling LED */
+ if (!spec->hp_detect ||
+ (spec->gpio_led > 8 && spec->gpio_led == nid)) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -5057,20 +5059,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
return 0;
}
-static int stac92xx_post_suspend(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
- if (spec->gpio_led > 8) {
- /* with vref-out pin used for mute led control
- * codec AFG is prevented from D3 state, but on
- * system suspend it can (and should) be used
- */
- snd_hda_codec_read(codec, codec->afg, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- }
- return 0;
-}
-
static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
@@ -5671,8 +5659,6 @@ again:
} else {
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
- codec->patch_ops.post_suspend =
- stac92xx_post_suspend;
}
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
@@ -5986,8 +5972,6 @@ again:
} else {
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
- codec->patch_ops.post_suspend =
- stac92xx_post_suspend;
}
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 431c0d417eeb..b5137629f8e9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -208,6 +208,7 @@ struct via_spec {
/* work to check hp jack state */
struct hda_codec *codec;
struct delayed_work vt1708_hp_work;
+ int hp_work_active;
int vt1708_jack_detect;
int vt1708_hp_present;
@@ -305,27 +306,35 @@ enum {
static void analog_low_current_mode(struct hda_codec *codec);
static bool is_aa_path_mute(struct hda_codec *codec);
-static void vt1708_start_hp_work(struct via_spec *spec)
+#define hp_detect_with_aa(codec) \
+ (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1 && \
+ !is_aa_path_mute(codec))
+
+static void vt1708_stop_hp_work(struct via_spec *spec)
{
if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
return;
- snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
- !spec->vt1708_jack_detect);
- if (!delayed_work_pending(&spec->vt1708_hp_work))
- schedule_delayed_work(&spec->vt1708_hp_work,
- msecs_to_jiffies(100));
+ if (spec->hp_work_active) {
+ snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 1);
+ cancel_delayed_work_sync(&spec->vt1708_hp_work);
+ spec->hp_work_active = 0;
+ }
}
-static void vt1708_stop_hp_work(struct via_spec *spec)
+static void vt1708_update_hp_work(struct via_spec *spec)
{
if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
return;
- if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1
- && !is_aa_path_mute(spec->codec))
- return;
- snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
- !spec->vt1708_jack_detect);
- cancel_delayed_work_sync(&spec->vt1708_hp_work);
+ if (spec->vt1708_jack_detect &&
+ (spec->active_streams || hp_detect_with_aa(spec->codec))) {
+ if (!spec->hp_work_active) {
+ snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 0);
+ schedule_delayed_work(&spec->vt1708_hp_work,
+ msecs_to_jiffies(100));
+ spec->hp_work_active = 1;
+ }
+ } else if (!hp_detect_with_aa(spec->codec))
+ vt1708_stop_hp_work(spec);
}
static void set_widgets_power_state(struct hda_codec *codec)
@@ -343,12 +352,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol,
set_widgets_power_state(codec);
analog_low_current_mode(snd_kcontrol_chip(kcontrol));
- if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) {
- if (is_aa_path_mute(codec))
- vt1708_start_hp_work(codec->spec);
- else
- vt1708_stop_hp_work(codec->spec);
- }
+ vt1708_update_hp_work(codec->spec);
return change;
}
@@ -1154,7 +1158,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
spec->cur_dac_stream_tag = stream_tag;
spec->cur_dac_format = format;
mutex_unlock(&spec->config_mutex);
- vt1708_start_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1174,7 +1178,7 @@ static int via_playback_hp_pcm_prepare(struct hda_pcm_stream *hinfo,
spec->cur_hp_stream_tag = stream_tag;
spec->cur_hp_format = format;
mutex_unlock(&spec->config_mutex);
- vt1708_start_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1188,7 +1192,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
spec->active_streams &= ~STREAM_MULTI_OUT;
mutex_unlock(&spec->config_mutex);
- vt1708_stop_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1203,7 +1207,7 @@ static int via_playback_hp_pcm_cleanup(struct hda_pcm_stream *hinfo,
snd_hda_codec_setup_stream(codec, spec->hp_dac_nid, 0, 0, 0);
spec->active_streams &= ~STREAM_INDEP_HP;
mutex_unlock(&spec->config_mutex);
- vt1708_stop_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1645,7 +1649,8 @@ static void via_hp_automute(struct hda_codec *codec)
int nums;
struct via_spec *spec = codec->spec;
- if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0])
+ if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0] &&
+ (spec->codec_type != VT1708 || spec->vt1708_jack_detect))
present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
if (spec->smart51_enabled)
@@ -2612,8 +2617,6 @@ static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol,
if (spec->codec_type != VT1708)
return 0;
- spec->vt1708_jack_detect =
- !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1);
ucontrol->value.integer.value[0] = spec->vt1708_jack_detect;
return 0;
}
@@ -2623,18 +2626,22 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- int change;
+ int val;
if (spec->codec_type != VT1708)
return 0;
- spec->vt1708_jack_detect = ucontrol->value.integer.value[0];
- change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8))
- == !spec->vt1708_jack_detect;
- if (spec->vt1708_jack_detect) {
+ val = !!ucontrol->value.integer.value[0];
+ if (spec->vt1708_jack_detect == val)
+ return 0;
+ spec->vt1708_jack_detect = val;
+ if (spec->vt1708_jack_detect &&
+ snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") != 1) {
mute_aa_path(codec, 1);
notify_aa_path_ctls(codec);
}
- return change;
+ via_hp_automute(codec);
+ vt1708_update_hp_work(spec);
+ return 1;
}
static const struct snd_kcontrol_new vt1708_jack_detect_ctl = {
@@ -2771,6 +2778,7 @@ static int via_init(struct hda_codec *codec)
via_auto_init_unsol_event(codec);
via_hp_automute(codec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -2787,7 +2795,9 @@ static void vt1708_update_hp_jack_state(struct work_struct *work)
spec->vt1708_hp_present ^= 1;
via_hp_automute(spec->codec);
}
- vt1708_start_hp_work(spec);
+ if (spec->vt1708_jack_detect)
+ schedule_delayed_work(&spec->vt1708_hp_work,
+ msecs_to_jiffies(100));
}
static int get_mux_nids(struct hda_codec *codec)
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index a391e622a192..28dfafb56dd1 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}");
static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
static int enable = 1;
+static int codecs = 1;
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator.");
@@ -48,6 +49,8 @@ module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator.");
module_param(enable, bool, 0444);
MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator.");
+module_param(codecs, int, 0444);
+MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)");
static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = {
{ PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) },
@@ -140,6 +143,9 @@ struct sis7019 {
dma_addr_t silence_dma_addr;
};
+/* These values are also used by the module param 'codecs' to indicate
+ * which codecs should be present.
+ */
#define SIS_PRIMARY_CODEC_PRESENT 0x0001
#define SIS_SECONDARY_CODEC_PRESENT 0x0002
#define SIS_TERTIARY_CODEC_PRESENT 0x0004
@@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis)
{
unsigned long io = sis->ioport;
void __iomem *ioaddr = sis->ioaddr;
+ unsigned long timeout;
u16 status;
int count;
int i;
@@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis)
while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
udelay(1);
+ /* Command complete, we can let go of the semaphore now.
+ */
+ outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
+ if (!count)
+ return -EIO;
+
/* Now that we've finished the reset, find out what's attached.
+ * There are some codec/board combinations that take an extremely
+ * long time to come up. 350+ ms has been observed in the field,
+ * so we'll give them up to 500ms.
*/
- status = inl(io + SIS_AC97_STATUS);
- if (status & SIS_AC97_STATUS_CODEC_READY)
- sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
- if (status & SIS_AC97_STATUS_CODEC2_READY)
- sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
- if (status & SIS_AC97_STATUS_CODEC3_READY)
- sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
-
- /* All done, let go of the semaphore, and check for errors
+ sis->codecs_present = 0;
+ timeout = msecs_to_jiffies(500) + jiffies;
+ while (time_before_eq(jiffies, timeout)) {
+ status = inl(io + SIS_AC97_STATUS);
+ if (status & SIS_AC97_STATUS_CODEC_READY)
+ sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
+ if (status & SIS_AC97_STATUS_CODEC2_READY)
+ sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
+ if (status & SIS_AC97_STATUS_CODEC3_READY)
+ sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
+
+ if (sis->codecs_present == codecs)
+ break;
+
+ msleep(1);
+ }
+
+ /* All done, check for errors.
*/
- outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
- if (!sis->codecs_present || !count)
+ if (!sis->codecs_present) {
+ printk(KERN_ERR "sis7019: could not find any codecs\n");
return -EIO;
+ }
+
+ if (sis->codecs_present != codecs) {
+ printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n",
+ sis->codecs_present, codecs);
+ }
/* Let the hardware know that the audio driver is alive,
* and enable PCM slots on the AC-link for L/R playback (3 & 4) and
@@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci,
if (!enable)
goto error_out;
+ /* The user can specify which codecs should be present so that we
+ * can wait for them to show up if they are slow to recover from
+ * the AC97 cold reset. We default to a single codec, the primary.
+ *
+ * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2.
+ */
+ codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT |
+ SIS_TERTIARY_CODEC_PRESENT;
+ if (!codecs)
+ codecs = SIS_PRIMARY_CODEC_PRESENT;
+
rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card);
if (rc < 0)
goto error_out;
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index bee3c94f58b0..d1fcc816ce97 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
config SND_ATMEL_SOC
tristate "SoC Audio for the Atmel System-on-Chip"
- depends on ARCH_AT91 || AVR32
+ depends on ARCH_AT91
help
Say Y or M if you want to add support for codecs attached to
the ATMEL SSC interface. You will also need
@@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731
Say Y if you want to add support for SoC audio on WM8731-based
AT91sam9g20 evaluation board.
-config SND_AT32_SOC_PLAYPAQ
- tristate "SoC Audio support for PlayPaq with WM8510"
- depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
- select SND_ATMEL_SOC_SSC
- select SND_SOC_WM8510
- help
- Say Y or M here if you want to add support for SoC audio
- on the LRS PlayPaq.
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
- bool "Run CODEC on PlayPaq in slave mode"
- depends on SND_AT32_SOC_PLAYPAQ
- default n
- help
- Say Y if you want to run with the AT32 SSC generating the BCLK
- and FRAME signals on the PlayPaq. Unless you want to play
- with the AT32 as the SSC master, you probably want to say N here,
- as this will give you better sound quality.
-
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index e7ea56bd5f82..a5c0bf19da78 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
# AT91 Machine Support
snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
deleted file mode 100644
index 73ae99ad4578..000000000000
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ /dev/null
@@ -1,473 +0,0 @@
-/* sound/soc/at32/playpaq_wm8510.c
- * ASoC machine driver for PlayPaq using WM8510 codec
- *
- * Copyright (C) 2008 Long Range Systems
- * Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
- *
- * NOTE: If you don't have the AT32 enhanced portmux configured (which
- * isn't currently in the mainline or Atmel patched kernel), you will
- * need to set the MCLK pin (PA30) to peripheral A in your board initialization
- * code. Something like:
- * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
- *
- */
-
-/* #define DEBUG */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/errno.h>
-#include <linux/clk.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/at32ap700x.h>
-#include <mach/portmux.h>
-
-#include "../codecs/wm8510.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-
-/*-------------------------------------------------------------------------*\
- * constants
-\*-------------------------------------------------------------------------*/
-#define MCLK_PIN GPIO_PIN_PA(30)
-#define MCLK_PERIPH GPIO_PERIPH_A
-
-
-/*-------------------------------------------------------------------------*\
- * data types
-\*-------------------------------------------------------------------------*/
-/* SSC clocking data */
-struct ssc_clock_data {
- /* CMR div */
- unsigned int cmr_div;
-
- /* Frame period (as needed by xCMR.PERIOD) */
- unsigned int period;
-
- /* The SSC clock rate these settings where calculated for */
- unsigned long ssc_rate;
-};
-
-
-/*-------------------------------------------------------------------------*\
- * module data
-\*-------------------------------------------------------------------------*/
-static struct clk *_gclk0;
-static struct clk *_pll0;
-
-#define CODEC_CLK (_gclk0)
-
-
-/*-------------------------------------------------------------------------*\
- * Sound SOC operations
-\*-------------------------------------------------------------------------*/
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
-{
- struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
- struct ssc_device *ssc = ssc_p->ssc;
- struct ssc_clock_data cd;
- unsigned int rate, width_bits, channels;
- unsigned int bitrate, ssc_div;
- unsigned actual_rate;
-
-
- /*
- * Figure out required bitrate
- */
- rate = params_rate(params);
- channels = params_channels(params);
- width_bits = snd_pcm_format_physical_width(params_format(params));
- bitrate = rate * width_bits * channels;
-
-
- /*
- * Figure out required SSC divider and period for required bitrate
- */
- cd.ssc_rate = clk_get_rate(ssc->clk);
- ssc_div = cd.ssc_rate / bitrate;
- cd.cmr_div = ssc_div / 2;
- if (ssc_div & 1) {
- /* round cmr_div up */
- cd.cmr_div++;
- }
- cd.period = width_bits - 1;
-
-
- /*
- * Find actual rate, compare to requested rate
- */
- actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
- pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
- rate, actual_rate);
-
-
- return cd;
-}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-
-static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
- struct ssc_device *ssc = ssc_p->ssc;
- unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
- int ret;
-
-
- /* Due to difficulties with getting the correct clocks from the AT32's
- * PLL0, we're going to let the CODEC be in charge of all the clocks
- */
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
- const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
-#else
- struct ssc_clock_data cd;
- const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
-#endif
-
- if (ssc == NULL) {
- pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
- return -EINVAL;
- }
-
-
- /*
- * Figure out PLL and BCLK dividers for WM8510
- */
- switch (params_rate(params)) {
- case 48000:
- pll_out = 24576000;
- mclk_div = WM8510_MCLKDIV_2;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 44100:
- pll_out = 22579200;
- mclk_div = WM8510_MCLKDIV_2;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 22050:
- pll_out = 22579200;
- mclk_div = WM8510_MCLKDIV_4;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 16000:
- pll_out = 24576000;
- mclk_div = WM8510_MCLKDIV_6;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 11025:
- pll_out = 22579200;
- mclk_div = WM8510_MCLKDIV_8;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 8000:
- pll_out = 24576000;
- mclk_div = WM8510_MCLKDIV_12;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- default:
- pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
- params_rate(params));
- return -EINVAL;
- }
-
-
- /*
- * set CPU and CODEC DAI configuration
- */
- ret = snd_soc_dai_set_fmt(codec_dai, fmt);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: "
- "Failed to set CODEC DAI format (%d)\n",
- ret);
- return ret;
- }
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: "
- "Failed to set CPU DAI format (%d)\n",
- ret);
- return ret;
- }
-
-
- /*
- * Set CPU clock configuration
- */
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
- cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
- pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
- cd.cmr_div, cd.period);
- ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
- ret);
- return ret;
- }
- ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
- cd.period);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: "
- "Failed to set CPU transmit period (%d)\n",
- ret);
- return ret;
- }
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
- /*
- * Set CODEC clock configuration
- */
- pr_debug("playpaq_wm8510: "
- "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
- clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
-
-
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
- if (ret < 0) {
- pr_warning
- ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
- ret);
- return ret;
- }
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
- ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
- clk_get_rate(CODEC_CLK), pll_out);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
- ret);
- return ret;
- }
-
-
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
- ret);
- return ret;
- }
-
-
- return 0;
-}
-
-
-
-static struct snd_soc_ops playpaq_wm8510_ops = {
- .hw_params = playpaq_wm8510_hw_params,
-};
-
-
-
-static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Int Mic", NULL),
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
-};
-
-
-
-static const struct snd_soc_dapm_route intercon[] = {
- /* speaker connected to SPKOUT */
- {"Ext Spk", NULL, "SPKOUTP"},
- {"Ext Spk", NULL, "SPKOUTN"},
-
- {"Mic Bias", NULL, "Int Mic"},
- {"MICN", NULL, "Mic Bias"},
- {"MICP", NULL, "Mic Bias"},
-};
-
-
-
-static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int i;
-
- /*
- * Add DAPM widgets
- */
- for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
- snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
-
-
-
- /*
- * Setup audio path interconnects
- */
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-
-
- /* always connected pins */
- snd_soc_dapm_enable_pin(dapm, "Int Mic");
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
-
-
- /* Make CSB show PLL rate */
- snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV,
- WM8510_OPCLKDIV_1 | 4);
-
- return 0;
-}
-
-
-
-static struct snd_soc_dai_link playpaq_wm8510_dai = {
- .name = "WM8510",
- .stream_name = "WM8510 PCM",
- .cpu_dai_name= "atmel-ssc-dai.0",
- .platform_name = "atmel-pcm-audio",
- .codec_name = "wm8510-codec.0-0x1a",
- .codec_dai_name = "wm8510-hifi",
- .init = playpaq_wm8510_init,
- .ops = &playpaq_wm8510_ops,
-};
-
-
-
-static struct snd_soc_card snd_soc_playpaq = {
- .name = "LRS_PlayPaq_WM8510",
- .dai_link = &playpaq_wm8510_dai,
- .num_links = 1,
-};
-
-static struct platform_device *playpaq_snd_device;
-
-
-static int __init playpaq_asoc_init(void)
-{
- int ret = 0;
-
- /*
- * Configure MCLK for WM8510
- */
- _gclk0 = clk_get(NULL, "gclk0");
- if (IS_ERR(_gclk0)) {
- _gclk0 = NULL;
- ret = PTR_ERR(_gclk0);
- goto err_gclk0;
- }
- _pll0 = clk_get(NULL, "pll0");
- if (IS_ERR(_pll0)) {
- _pll0 = NULL;
- ret = PTR_ERR(_pll0);
- goto err_pll0;
- }
- ret = clk_set_parent(_gclk0, _pll0);
- if (ret) {
- pr_warning("snd-soc-playpaq: "
- "Failed to set PLL0 as parent for DAC clock\n");
- goto err_set_clk;
- }
- clk_set_rate(CODEC_CLK, 12000000);
- clk_enable(CODEC_CLK);
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
- at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
-#endif
-
-
- /*
- * Create and register platform device
- */
- playpaq_snd_device = platform_device_alloc("soc-audio", 0);
- if (playpaq_snd_device == NULL) {
- ret = -ENOMEM;
- goto err_device_alloc;
- }
-
- platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq);
-
- ret = platform_device_add(playpaq_snd_device);
- if (ret) {
- pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
- ret);
- goto err_device_add;
- }
-
- return 0;
-
-
-err_device_add:
- if (playpaq_snd_device != NULL) {
- platform_device_put(playpaq_snd_device);
- playpaq_snd_device = NULL;
- }
-err_device_alloc:
-err_set_clk:
- if (_pll0 != NULL) {
- clk_put(_pll0);
- _pll0 = NULL;
- }
-err_pll0:
- if (_gclk0 != NULL) {
- clk_put(_gclk0);
- _gclk0 = NULL;
- }
- return ret;
-}
-
-
-static void __exit playpaq_asoc_exit(void)
-{
- if (_gclk0 != NULL) {
- clk_put(_gclk0);
- _gclk0 = NULL;
- }
- if (_pll0 != NULL) {
- clk_put(_pll0);
- _pll0 = NULL;
- }
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
- at32_free_pin(MCLK_PIN);
-#endif
-
- platform_device_unregister(playpaq_snd_device);
- playpaq_snd_device = NULL;
-}
-
-module_init(playpaq_asoc_init);
-module_exit(playpaq_asoc_exit);
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 444747f0db26..dd7be0dbbc58 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -34,7 +34,7 @@
#define AD1836_ADC_CTRL2 13
#define AD1836_ADC_WORD_LEN_MASK 0x30
-#define AD1836_ADC_WORD_OFFSET 5
+#define AD1836_ADC_WORD_OFFSET 4
#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1f237ecec2a..73f46eb459f1 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
static int cs4270_soc_resume(struct snd_soc_codec *codec)
{
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *i2c_client = to_i2c_client(codec->dev);
int reg;
regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
ndelay(500);
/* first restore the entire register cache ... */
- for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
- u8 val = snd_soc_read(codec, reg);
-
- if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
- dev_err(codec->dev, "i2c write failed\n");
- return -EIO;
- }
- }
+ snd_soc_cache_sync(codec);
/* ... then disable the power-down bits */
reg = snd_soc_read(codec, CS4270_PWRCTL);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8c3c8205d19e..1ee66361f61b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
.probe = cs42l51_probe,
- .reg_cache_size = CS42L51_NUMREGS,
+ .reg_cache_size = CS42L51_NUMREGS + 1,
.reg_word_size = sizeof(u8),
};
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 9e7e964a5fa3..dcf6f2a1600a 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
unsigned int mask = mc->max;
unsigned int val = (ucontrol->value.integer.value[0] & mask);
unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
- unsigned int change = 1;
+ unsigned int change = 0;
- if (((max9877_regs[reg] >> shift) & mask) == val)
- change = 0;
+ if (((max9877_regs[reg] >> shift) & mask) != val)
+ change = 1;
- if (((max9877_regs[reg2] >> shift) & mask) == val2)
- change = 0;
+ if (((max9877_regs[reg2] >> shift) & mask) != val2)
+ change = 1;
if (change) {
max9877_regs[reg] &= ~(mask << shift);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e47eb99..6c2988549003 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
lrclk = bclk_rate / params_rate(params);
+ if (!lrclk) {
+ dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
+ bclk_rate);
+ return -EINVAL;
+ }
dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
lrclk, bclk_rate / lrclk);
@@ -3178,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
switch (wm8994->revision) {
case 0:
case 1:
+ case 2:
+ case 3:
wm8994->hubs.dcs_codes_l = -9;
wm8994->hubs.dcs_codes_r = -5;
break;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 31af405bda84..ae49f1c78c6d 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
if (strcasecmp(sprop, "i2s-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
@@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
machine_data->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "lj-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "lj-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "rj-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "rj-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "ac97-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "ac97-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else {
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index f75e43997d5b..ad9ac42522e2 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
#include "../codecs/wm8994.h"
#include <sound/pcm_params.h>
+#include <linux/module.h>
/*
* Default CFG switch settings to use this driver:
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 85bf541a771d..4b8e35410eb1 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card)
snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC");
snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC");
- snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker");
snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output");
snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685a5d38..a25fa63ce9a2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev)
struct snd_soc_card *card = dev_get_drvdata(dev);
int i, ac97_control = 0;
+ /* If the initialization of this soc device failed, there is no codec
+ * associated with it. Just bail out in this case.
+ */
+ if (list_empty(&card->codec_dev_list))
+ return 0;
+
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
* problem and may take a substantial amount of time to resume
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b61945f3af9e..32d2a21f2e3b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
{
+ /* Roland GAIA SH-01 */
+ USB_DEVICE(0x0582, 0x0111),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "GAIA",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &(const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
USB_DEVICE(0x0582, 0x0113),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
/* .vendor_name = "BOSS", */
generated by cgit v1.2.3 (git 2.39.0) at 2024-11-01 14:28:13 +0300