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-rw-r--r--sound/i2c/other/tea575x-tuner.c21
-rw-r--r--sound/pci/es1968.c2
-rw-r--r--sound/pci/fm801.c4
-rw-r--r--sound/pci/hda/Kconfig13
-rw-r--r--sound/pci/hda/hda_auto_parser.c1
-rw-r--r--sound/pci/hda/hda_auto_parser.h10
-rw-r--r--sound/pci/hda/hda_codec.c50
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/pci/hda/patch_conexant.c6
-rw-r--r--sound/pci/hda/patch_realtek.c15
-rw-r--r--sound/pci/hda/patch_sigmatel.c7
-rw-r--r--sound/soc/codecs/wm8904.c26
-rw-r--r--sound/soc/codecs/wm8994.c3
-rw-r--r--sound/soc/codecs/wm8996.c8
-rw-r--r--sound/soc/pxa/pxa-ssp.c38
-rw-r--r--sound/soc/tegra/tegra_wm8903.c13
-rw-r--r--sound/usb/6fire/firmware.c2
-rw-r--r--sound/usb/mixer_maps.c8
-rw-r--r--sound/usb/pcm.c21
-rw-r--r--sound/usb/quirks-table.h30
21 files changed, 182 insertions, 100 deletions
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index 582aace20ea3..7eca25fae413 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -37,8 +37,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
MODULE_LICENSE("GPL");
-#define FREQ_LO (76U * 16000)
-#define FREQ_HI (108U * 16000)
+#define FREQ_LO ((tea->tea5759 ? 760 : 875) * 1600U)
+#define FREQ_HI ((tea->tea5759 ? 910 : 1080) * 1600U)
/*
* definitions
@@ -120,9 +120,9 @@ static u32 snd_tea575x_read(struct snd_tea575x *tea)
return data;
}
-static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
+static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val)
{
- u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK;
+ u32 freq = val & TEA575X_BIT_FREQ_MASK;
if (freq == 0)
return freq;
@@ -139,6 +139,11 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */
}
+static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
+{
+ return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea));
+}
+
static void snd_tea575x_set_freq(struct snd_tea575x *tea)
{
u32 freq = tea->freq;
@@ -156,6 +161,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea)
tea->val &= ~TEA575X_BIT_FREQ_MASK;
tea->val |= freq & TEA575X_BIT_FREQ_MASK;
snd_tea575x_write(tea, tea->val);
+ tea->freq = snd_tea575x_val_to_freq(tea, tea->val);
}
/*
@@ -317,7 +323,6 @@ static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl)
}
static const struct v4l2_file_operations tea575x_fops = {
- .owner = THIS_MODULE,
.unlocked_ioctl = video_ioctl2,
.open = v4l2_fh_open,
.release = v4l2_fh_release,
@@ -337,7 +342,6 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = {
};
static const struct video_device tea575x_radio = {
- .fops = &tea575x_fops,
.ioctl_ops = &tea575x_ioctl_ops,
.release = video_device_release_empty,
};
@@ -349,7 +353,7 @@ static const struct v4l2_ctrl_ops tea575x_ctrl_ops = {
/*
* initialize all the tea575x chips
*/
-int snd_tea575x_init(struct snd_tea575x *tea)
+int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner)
{
int retval;
@@ -374,6 +378,9 @@ int snd_tea575x_init(struct snd_tea575x *tea)
tea->vd.lock = &tea->mutex;
tea->vd.v4l2_dev = tea->v4l2_dev;
tea->vd.ctrl_handler = &tea->ctrl_handler;
+ tea->fops = tea575x_fops;
+ tea->fops.owner = owner;
+ tea->vd.fops = &tea->fops;
set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags);
/* disable hw_freq_seek if we can't use it */
if (tea->cannot_read_data)
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 67f47d891959..52b5c0bf90c1 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2769,7 +2769,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
chip->tea.ops = &snd_es1968_tea_ops;
strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card));
sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
- if (!snd_tea575x_init(&chip->tea))
+ if (!snd_tea575x_init(&chip->tea, THIS_MODULE))
printk(KERN_INFO "es1968: detected TEA575x radio\n");
#endif
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f69662322750..b32e8024ea86 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1254,7 +1254,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 &&
(tea575x_tuner & TUNER_TYPE_MASK) < 4) {
- if (snd_tea575x_init(&chip->tea)) {
+ if (snd_tea575x_init(&chip->tea, THIS_MODULE)) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
snd_fm801_free(chip);
return -ENODEV;
@@ -1263,7 +1263,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
/* autodetect tuner connection */
for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) {
chip->tea575x_tuner = tea575x_tuner;
- if (!snd_tea575x_init(&chip->tea)) {
+ if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) {
snd_printk(KERN_INFO "detected TEA575x radio type %s\n",
get_tea575x_gpio(chip)->name);
break;
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 163b6b5de3eb..d03079764189 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -97,19 +97,6 @@ config SND_HDA_CODEC_REALTEK
snd-hda-codec-realtek.
This module is automatically loaded at probing.
-config SND_HDA_ENABLE_REALTEK_QUIRKS
- bool "Build static quirks for Realtek codecs"
- depends on SND_HDA_CODEC_REALTEK
- default y
- help
- Say Y here to build the static quirks codes for Realtek codecs.
- If you need the "model" preset that the default BIOS auto-parser
- can't handle, turn this option on.
-
- If your device works with model=auto option, basically you don't
- need the quirk code. By turning this off, you can reduce the
- module size quite a lot.
-
config SND_HDA_CODEC_ANALOG
bool "Build Analog Device HD-audio codec support"
default y
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 6e9ef3e25093..f7520b9f909c 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -618,7 +618,6 @@ int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
const struct hda_verb *list)
{
const struct hda_verb **v;
- snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
v = snd_array_new(&spec->verbs);
if (!v)
return -ENOMEM;
diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h
index 2a7889dfbd1b..632ad0ad3007 100644
--- a/sound/pci/hda/hda_auto_parser.h
+++ b/sound/pci/hda/hda_auto_parser.h
@@ -157,4 +157,14 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
const struct snd_pci_quirk *quirk,
const struct hda_fixup *fixlist);
+static inline void snd_hda_gen_init(struct hda_gen_spec *spec)
+{
+ snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
+}
+
+static inline void snd_hda_gen_free(struct hda_gen_spec *spec)
+{
+ snd_array_free(&spec->verbs);
+}
+
#endif /* __SOUND_HDA_AUTO_PARSER_H */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 41ca803a1fff..51cb2a2e4fce 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1184,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
{
if (!codec)
return;
+ snd_hda_jack_tbl_clear(codec);
restore_init_pincfgs(codec);
#ifdef CONFIG_SND_HDA_POWER_SAVE
cancel_delayed_work(&codec->power_work);
@@ -1192,6 +1193,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
list_del(&codec->list);
snd_array_free(&codec->mixers);
snd_array_free(&codec->nids);
+ snd_array_free(&codec->cvt_setups);
snd_array_free(&codec->conn_lists);
snd_array_free(&codec->spdif_out);
codec->bus->caddr_tbl[codec->addr] = NULL;
@@ -2333,6 +2335,8 @@ int snd_hda_codec_reset(struct hda_codec *codec)
/* free only driver_pins so that init_pins + user_pins are restored */
snd_array_free(&codec->driver_pins);
restore_pincfgs(codec);
+ snd_array_free(&codec->cvt_setups);
+ snd_array_free(&codec->spdif_out);
codec->num_pcms = 0;
codec->pcm_info = NULL;
codec->preset = NULL;
@@ -4393,20 +4397,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec)
codec->power_jiffies += delta;
}
-/**
- * snd_hda_power_up - Power-up the codec
- * @codec: HD-audio codec
- *
- * Increment the power-up counter and power up the hardware really when
- * not turned on yet.
- */
-void snd_hda_power_up(struct hda_codec *codec)
+/* Transition to powered up, if wait_power_down then wait for a pending
+ * transition to D3 to complete. A pending D3 transition is indicated
+ * with power_transition == -1. */
+static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
{
struct hda_bus *bus = codec->bus;
spin_lock(&codec->power_lock);
codec->power_count++;
- if (codec->power_on || codec->power_transition > 0) {
+ /* Return if power_on or transitioning to power_on, unless currently
+ * powering down. */
+ if ((codec->power_on || codec->power_transition > 0) &&
+ !(wait_power_down && codec->power_transition < 0)) {
spin_unlock(&codec->power_lock);
return;
}
@@ -4430,8 +4433,37 @@ void snd_hda_power_up(struct hda_codec *codec)
codec->power_transition = 0;
spin_unlock(&codec->power_lock);
}
+
+/**
+ * snd_hda_power_up - Power-up the codec
+ * @codec: HD-audio codec
+ *
+ * Increment the power-up counter and power up the hardware really when
+ * not turned on yet.
+ */
+void snd_hda_power_up(struct hda_codec *codec)
+{
+ __snd_hda_power_up(codec, false);
+}
EXPORT_SYMBOL_HDA(snd_hda_power_up);
+/**
+ * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending
+ * D3 transition to complete. This differs from snd_hda_power_up() when
+ * power_transition == -1. snd_hda_power_up sees this case as a nop,
+ * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers
+ * back up.
+ * @codec: HD-audio codec
+ *
+ * Cancel any power down operation hapenning on the work queue, then power up.
+ */
+void snd_hda_power_up_d3wait(struct hda_codec *codec)
+{
+ /* This will cancel and wait for pending power_work to complete. */
+ __snd_hda_power_up(codec, true);
+}
+EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait);
+
#define power_save(codec) \
((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 4fc3960c8591..2fdaadbb4326 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg);
*/
#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_up_d3wait(struct hda_codec *codec);
void snd_hda_power_down(struct hda_codec *codec);
void snd_hda_update_power_acct(struct hda_codec *codec);
#else
static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {}
static inline void snd_hda_power_down(struct hda_codec *codec) {}
#endif
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 02763827dde0..7757536b9d5f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
buff_step);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
buff_step);
- snd_hda_power_up(apcm->codec);
+ snd_hda_power_up_d3wait(apcm->codec);
err = hinfo->ops.open(hinfo, apcm->codec, substream);
if (err < 0) {
azx_release_device(azx_dev);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 172370b3793b..2bf99fc1cbf2 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -445,8 +445,10 @@ static int conexant_init(struct hda_codec *codec)
static void conexant_free(struct hda_codec *codec)
{
+ struct conexant_spec *spec = codec->spec;
+ snd_hda_gen_free(&spec->gen);
snd_hda_detach_beep_device(codec);
- kfree(codec->spec);
+ kfree(spec);
}
static const struct snd_kcontrol_new cxt_capture_mixers[] = {
@@ -4466,6 +4468,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
{}
};
@@ -4497,6 +4500,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
if (!spec)
return -ENOMEM;
codec->spec = spec;
+ snd_hda_gen_init(&spec->gen);
switch (codec->vendor_id) {
case 0x14f15045:
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f8f4906e498d..5ccf10a4d593 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2289,6 +2289,7 @@ static void alc_free(struct hda_codec *codec)
alc_shutup(codec);
alc_free_kctls(codec);
alc_free_bind_ctls(codec);
+ snd_hda_gen_free(&spec->gen);
kfree(spec);
snd_hda_detach_beep_device(codec);
}
@@ -4253,6 +4254,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
return -ENOMEM;
codec->spec = spec;
spec->mixer_nid = mixer_nid;
+ snd_hda_gen_init(&spec->gen);
err = alc_codec_rename_from_preset(codec);
if (err < 0) {
@@ -6705,6 +6707,12 @@ static int patch_alc662(struct hda_codec *codec)
alc_fix_pll_init(codec, 0x20, 0x04, 15);
+ alc_pick_fixup(codec, alc662_fixup_models,
+ alc662_fixup_tbl, alc662_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
+ alc_auto_parse_customize_define(codec);
+
if ((alc_get_coef0(codec) & (1 << 14)) &&
codec->bus->pci->subsystem_vendor == 0x1025 &&
spec->cdefine.platform_type == 1) {
@@ -6712,12 +6720,6 @@ static int patch_alc662(struct hda_codec *codec)
goto error;
}
- alc_pick_fixup(codec, alc662_fixup_models,
- alc662_fixup_tbl, alc662_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
-
- alc_auto_parse_customize_define(codec);
-
/* automatic parse from the BIOS config */
err = alc662_parse_auto_config(codec);
if (err < 0)
@@ -6800,6 +6802,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
+ { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
.patch = patch_alc861 },
{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7db8228f1b88..07675282015a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4367,7 +4367,7 @@ static int stac92xx_init(struct hda_codec *codec)
AC_PINCTL_IN_EN);
for (i = 0; i < spec->num_pwrs; i++) {
hda_nid_t nid = spec->pwr_nids[i];
- int pinctl, def_conf;
+ unsigned int pinctl, def_conf;
def_conf = snd_hda_codec_get_pincfg(codec, nid);
def_conf = get_defcfg_connect(def_conf);
@@ -4376,6 +4376,11 @@ static int stac92xx_init(struct hda_codec *codec)
stac_toggle_power_map(codec, nid, 0);
continue;
}
+ if (def_conf == AC_JACK_PORT_FIXED) {
+ /* no need for jack detection for fixed pins */
+ stac_toggle_power_map(codec, nid, 1);
+ continue;
+ }
/* power on when no jack detection is available */
/* or when the VREF is used for controlling LED */
if (!spec->hp_detect ||
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 65d525d74c54..812acd83fb48 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
+ regcache_cache_only(wm8904->regmap, false);
regcache_sync(wm8904->regmap);
/* Enable bias */
@@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_BIAS_ENA, 0);
-#ifdef CONFIG_REGULATOR
- /* Post 2.6.34 we will be able to get a callback when
- * the regulators are disabled which we can use but
- * for now just assume that the power will be cut if
- * the regulator API is in use.
- */
- codec->cache_sync = 1;
-#endif
+ regcache_cache_only(wm8904->regmap, true);
+ regcache_mark_dirty(wm8904->regmap);
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
@@ -2084,10 +2079,8 @@ static int wm8904_probe(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
- u16 *reg_cache = codec->reg_cache;
int ret, i;
- codec->cache_sync = 1;
codec->control_data = wm8904->regmap;
switch (wm8904->devtype) {
@@ -2150,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec)
goto err_enable;
}
+ regcache_cache_only(wm8904->regmap, true);
/* Change some default settings - latch VU and enable ZC */
snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
WM8904_ADC_VU, WM8904_ADC_VU);
@@ -2180,14 +2174,18 @@ static int wm8904_probe(struct snd_soc_codec *codec)
if (!pdata->gpio_cfg[i])
continue;
- reg_cache[WM8904_GPIO_CONTROL_1 + i]
- = pdata->gpio_cfg[i] & 0xffff;
+ regmap_update_bits(wm8904->regmap,
+ WM8904_GPIO_CONTROL_1 + i,
+ 0xffff,
+ pdata->gpio_cfg[i]);
}
/* Zero is the default value for these anyway */
for (i = 0; i < WM8904_MIC_REGS; i++)
- reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i]
- = pdata->mic_cfg[i];
+ regmap_update_bits(wm8904->regmap,
+ WM8904_MIC_BIAS_CONTROL_0 + i,
+ 0xffff,
+ pdata->mic_cfg[i]);
}
/* Set Class W by default - this will be managed by the Class
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index aa8c98b628da..1436b6ce74d1 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -727,9 +727,6 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode)
if (!wm8994->jackdet || !wm8994->jack_cb)
return;
- if (!wm8994->jackdet || !wm8994->jack_cb)
- return;
-
if (wm8994->active_refcount)
mode = WM1811_JACKDET_MODE_AUDIO;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 8af422e38fd0..dc9b42b7fc4d 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
}
}
- regcache_cache_only(codec->control_data, true);
-
/* Apply platform data settings */
snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
@@ -3051,7 +3049,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
regulator_unregister_notifier(wm8996->supplies[i].consumer,
&wm8996->disable_nb[i]);
- regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
return 0;
}
@@ -3206,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
dev_info(&i2c->dev, "revision %c\n",
(reg & WM8996_CHIP_REV_MASK) + 'A');
- regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
-
ret = wm8996_reset(wm8996);
if (ret < 0) {
dev_err(&i2c->dev, "Failed to issue reset\n");
goto err_regmap;
}
+ regcache_cache_only(wm8996->regmap, true);
+ regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+
wm8996_init_gpio(wm8996);
ret = snd_soc_register_codec(&i2c->dev,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 1c2aa7fab3fd..4da5fc55c7ee 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -33,7 +33,6 @@
#include <mach/hardware.h>
#include <mach/dma.h>
-#include <mach/audio.h>
#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
@@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
{
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
- if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+ if (ssp->type == PXA25x_SSP) {
sscr0 &= ~0x0000ff00;
sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
} else {
@@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
u32 div;
- if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+ if (ssp->type == PXA25x_SSP)
div = ((sscr0 >> 8) & 0xff) * 2 + 2;
else
div = ((sscr0 >> 8) & 0xfff) + 1;
@@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
break;
case PXA_SSP_CLK_PLL:
/* Internal PLL is fixed */
- if (cpu_is_pxa25x())
+ if (ssp->type == PXA25x_SSP)
priv->sysclk = 1843200;
else
priv->sysclk = 13000000;
@@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
/* The SSP clock must be disabled when changing SSP clock mode
* on PXA2xx. On PXA3xx it must be enabled when doing so. */
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
clk_disable(ssp->clk);
val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
pxa_ssp_write_reg(ssp, SSCR0, val);
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
clk_enable(ssp->clk);
return 0;
@@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
case PXA_SSP_AUDIO_DIV_SCDB:
val = pxa_ssp_read_reg(ssp, SSACD);
val &= ~SSACD_SCDB;
-#if defined(CONFIG_PXA3xx)
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
val &= ~SSACD_SCDX8;
-#endif
switch (div) {
case PXA_SSP_CLK_SCDB_1:
val |= SSACD_SCDB;
break;
case PXA_SSP_CLK_SCDB_4:
break;
-#if defined(CONFIG_PXA3xx)
case PXA_SSP_CLK_SCDB_8:
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
val |= SSACD_SCDX8;
else
return -EINVAL;
break;
-#endif
default:
return -EINVAL;
}
@@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
struct ssp_device *ssp = priv->ssp;
u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
-#if defined(CONFIG_PXA3xx)
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
pxa_ssp_write_reg(ssp, SSACDD, 0);
-#endif
switch (freq_out) {
case 5622000:
@@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
break;
default:
-#ifdef CONFIG_PXA3xx
/* PXA3xx has a clock ditherer which can be used to generate
* a wider range of frequencies - calculate a value for it.
*/
- if (cpu_is_pxa3xx()) {
+ if (ssp->type == PXA3xx_SSP) {
u32 val;
u64 tmp = 19968;
tmp *= 1000000;
@@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
val, freq_out);
break;
}
-#endif
return -EINVAL;
}
@@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
-#ifdef CONFIG_PXA3xx
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
sscr0 |= SSCR0_FPCKE;
-#endif
sscr0 |= SSCR0_DataSize(16);
break;
case SNDRV_PCM_FORMAT_S24_LE:
@@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
* trying and failing a lot; some of the registers
* needed for that mode are only available on PXA3xx.
*/
-
-#ifdef CONFIG_PXA3xx
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
return -EINVAL;
sspsp |= SSPSP_SFRMWDTH(width * 2);
@@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
sspsp |= SSPSP_EDMYSTOP(3);
sspsp |= SSPSP_DMYSTOP(3);
sspsp |= SSPSP_DMYSTRT(1);
-#else
- return -EINVAL;
-#endif
} else {
/* The frame width is the width the LRCLK is
* asserted for; the delay is expressed in
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0b0df49d9d33..3b6da91188a9 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
+static int tegra_wm8903_remove(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]);
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ wm8903_mic_detect(codec, NULL, 0, 0);
+
+ return 0;
+}
+
static struct snd_soc_dai_link tegra_wm8903_dai = {
.name = "WM8903",
.stream_name = "WM8903 PCM",
@@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
.dai_link = &tegra_wm8903_dai,
.num_links = 1,
+ .remove = tegra_wm8903_remove,
+
.controls = tegra_wm8903_controls,
.num_controls = ARRAY_SIZE(tegra_wm8903_controls),
.dapm_widgets = tegra_wm8903_dapm_widgets,
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 6f9715ab32fe..56ad923bf6b5 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload(
int ret;
u8 data;
struct usb_device *device = interface_to_usbdev(intf);
- const struct firmware *fw = 0;
+ const struct firmware *fw = NULL;
struct ihex_record *rec = kmalloc(sizeof(struct ihex_record),
GFP_KERNEL);
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 41daaa24c25f..e71fe55cebef 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -341,6 +341,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
.map = audigy2nx_map,
.selector_map = audigy2nx_selectors,
},
+ { /* Logitech, Inc. QuickCam Pro for Notebooks */
+ .id = USB_ID(0x046d, 0x0991),
+ .ignore_ctl_error = 1,
+ },
+ { /* Logitech, Inc. QuickCam E 3500 */
+ .id = USB_ID(0x046d, 0x09a4),
+ .ignore_ctl_error = 1,
+ },
{
/* Hercules DJ Console (Windows Edition) */
.id = USB_ID(0x06f8, 0xb000),
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cdf8b7601973..54607f8c4f66 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
(get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
get_endpoint(alts, 1)->bSynchAddress != 0 &&
!implicit_fb)) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
- dev->devnum, fmt->iface, fmt->altsetting);
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ get_endpoint(alts, 1)->bmAttributes,
+ get_endpoint(alts, 1)->bLength,
+ get_endpoint(alts, 1)->bSynchAddress);
return -EINVAL;
}
ep = get_endpoint(alts, 1)->bEndpointAddress;
- if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ if (!implicit_fb &&
+ get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
(( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
- (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) ||
- ( is_playback && !implicit_fb))) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
- dev->devnum, fmt->iface, fmt->altsetting);
+ (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
return -EINVAL;
}
@@ -1147,7 +1151,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
return -EINVAL;
}
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
{
int err;
struct snd_usb_substream *subs = substream->runtime->private_data;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d89ab4c7d44b..79780fa57a43 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1831,6 +1831,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
+{
+ USB_DEVICE(0x0582, 0x014d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "BOSS", */
+ /* .product_name = "GT-100", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
/* Guillemot devices */
{
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.