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-rw-r--r--sound/pci/cs5535audio/cs5535audio_pcm.c2
-rw-r--r--sound/pci/hda/hda_codec.c6
-rw-r--r--sound/pci/hda/hda_eld.c28
-rw-r--r--sound/pci/hda/hda_intel.c5
-rw-r--r--sound/pci/hda/patch_cirrus.c32
-rw-r--r--sound/pci/hda/patch_hdmi.c16
-rw-r--r--sound/pci/hda/patch_realtek.c99
-rw-r--r--sound/pci/hda/patch_sigmatel.c75
-rw-r--r--sound/pci/hda/patch_via.c76
-rw-r--r--sound/pci/lx6464es/lx_core.c23
-rw-r--r--sound/pci/lx6464es/lx_core.h3
-rw-r--r--sound/pci/rme9652/hdspm.c2
-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/Kconfig2
-rw-r--r--sound/soc/codecs/ad1836.h2
-rw-r--r--sound/soc/codecs/adau1373.c2
-rw-r--r--sound/soc/codecs/cs4270.c10
-rw-r--r--sound/soc/codecs/cs4271.c8
-rw-r--r--sound/soc/codecs/cs42l51.c2
-rw-r--r--sound/soc/codecs/jz4740.c1
-rw-r--r--sound/soc/codecs/max9877.c10
-rw-r--r--sound/soc/codecs/rt5631.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sta32x.c63
-rw-r--r--sound/soc/codecs/sta32x.h1
-rw-r--r--sound/soc/codecs/uda1380.c4
-rw-r--r--sound/soc/codecs/wm8731.c1
-rw-r--r--sound/soc/codecs/wm8753.c3
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm8994.c19
-rw-r--r--sound/soc/codecs/wm8996.c1
-rw-r--r--sound/soc/codecs/wm9081.c10
-rw-r--r--sound/soc/codecs/wm9090.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c1
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c24
-rw-r--r--sound/soc/imx/Kconfig2
-rw-r--r--sound/soc/kirkwood/Kconfig3
-rw-r--r--sound/soc/mxs/mxs-pcm.c3
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c1
-rw-r--r--sound/soc/nuc900/nuc900-ac97.c3
-rw-r--r--sound/soc/pxa/Kconfig3
-rw-r--r--sound/soc/pxa/hx4700.c5
-rw-r--r--sound/soc/samsung/jive_wm8750.c3
-rw-r--r--sound/soc/samsung/smdk2443_wm9710.c1
-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/soc/soc-utils.c31
-rw-r--r--sound/usb/quirks-table.h31
55 files changed, 497 insertions, 711 deletions
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index e083122ca55a..dbf94b189e75 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -148,7 +148,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
struct cs5535audio_dma_desc *desc =
&((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
desc->addr = cpu_to_le32(addr);
- desc->size = cpu_to_le32(period_bytes);
+ desc->size = cpu_to_le16(period_bytes);
desc->ctlreserved = cpu_to_le16(PRD_EOP);
desc_addr += sizeof(struct cs5535audio_dma_desc);
addr += period_bytes;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e44b107fdc75..4562e9de6a1a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4046,9 +4046,9 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
/* Search for codec ID */
for (q = tbl; q->subvendor; q++) {
- unsigned long vendorid = (q->subdevice) | (q->subvendor << 16);
-
- if (vendorid == codec->subsystem_id)
+ unsigned int mask = 0xffff0000 | q->subdevice_mask;
+ unsigned int id = (q->subdevice | (q->subvendor << 16)) & mask;
+ if ((codec->subsystem_id & mask) == id)
break;
}
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 7ae7578bdcc0..c1da422e085a 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -347,18 +347,28 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
for (i = 0; i < size; i++) {
unsigned int val = hdmi_get_eld_data(codec, nid, i);
+ /*
+ * Graphics driver might be writing to ELD buffer right now.
+ * Just abort. The caller will repoll after a while.
+ */
if (!(val & AC_ELDD_ELD_VALID)) {
- if (!i) {
- snd_printd(KERN_INFO
- "HDMI: invalid ELD data\n");
- ret = -EINVAL;
- goto error;
- }
snd_printd(KERN_INFO
"HDMI: invalid ELD data byte %d\n", i);
- val = 0;
- } else
- val &= AC_ELDD_ELD_DATA;
+ ret = -EINVAL;
+ goto error;
+ }
+ val &= AC_ELDD_ELD_DATA;
+ /*
+ * The first byte cannot be zero. This can happen on some DVI
+ * connections. Some Intel chips may also need some 250ms delay
+ * to return non-zero ELD data, even when the graphics driver
+ * correctly writes ELD content before setting ELD_valid bit.
+ */
+ if (!val && !i) {
+ snd_printdd(KERN_INFO "HDMI: 0 ELD data\n");
+ ret = -EINVAL;
+ goto error;
+ }
buf[i] = val;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 096507d2ca9a..c2f79e63124d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2507,8 +2507,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", 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),
@@ -2971,7 +2971,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
- AZX_DCAPS_BUFSIZE},
+ AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */
+ /* ICH */
{ PCI_DEVICE(0x8086, 0x2668),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
AZX_DCAPS_BUFSIZE }, /* ICH6 */
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2fbab8e29576..70a7abda7e22 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -58,6 +58,8 @@ struct cs_spec {
unsigned int gpio_mask;
unsigned int gpio_dir;
unsigned int gpio_data;
+ unsigned int gpio_eapd_hp; /* EAPD GPIO bit for headphones */
+ unsigned int gpio_eapd_speaker; /* EAPD GPIO bit for speakers */
struct hda_pcm pcm_rec[2]; /* PCM information */
@@ -76,6 +78,7 @@ enum {
CS420X_MBP53,
CS420X_MBP55,
CS420X_IMAC27,
+ CS420X_APPLE,
CS420X_AUTO,
CS420X_MODELS
};
@@ -928,10 +931,9 @@ static void cs_automute(struct hda_codec *codec)
spdif_present ? 0 : PIN_OUT);
}
}
- if (spec->board_config == CS420X_MBP53 ||
- spec->board_config == CS420X_MBP55 ||
- spec->board_config == CS420X_IMAC27) {
- unsigned int gpio = hp_present ? 0x02 : 0x08;
+ if (spec->gpio_eapd_hp) {
+ unsigned int gpio = hp_present ?
+ spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, gpio);
}
@@ -1276,6 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = {
[CS420X_MBP53] = "mbp53",
[CS420X_MBP55] = "mbp55",
[CS420X_IMAC27] = "imac27",
+ [CS420X_APPLE] = "apple",
[CS420X_AUTO] = "auto",
};
@@ -1285,7 +1288,13 @@ static const struct snd_pci_quirk cs420x_cfg_tbl[] = {
SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55),
SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55),
- SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),
+ /* this conflicts with too many other models */
+ /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/
+ {} /* terminator */
+};
+
+static const struct snd_pci_quirk cs420x_codec_cfg_tbl[] = {
+ SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
{} /* terminator */
};
@@ -1367,6 +1376,10 @@ static int patch_cs420x(struct hda_codec *codec)
spec->board_config =
snd_hda_check_board_config(codec, CS420X_MODELS,
cs420x_models, cs420x_cfg_tbl);
+ if (spec->board_config < 0)
+ spec->board_config =
+ snd_hda_check_board_codec_sid_config(codec,
+ CS420X_MODELS, NULL, cs420x_codec_cfg_tbl);
if (spec->board_config >= 0)
fix_pincfg(codec, spec->board_config, cs_pincfgs);
@@ -1374,10 +1387,11 @@ static int patch_cs420x(struct hda_codec *codec)
case CS420X_IMAC27:
case CS420X_MBP53:
case CS420X_MBP55:
- /* GPIO1 = headphones */
- /* GPIO3 = speakers */
- spec->gpio_mask = 0x0a;
- spec->gpio_dir = 0x0a;
+ case CS420X_APPLE:
+ spec->gpio_eapd_hp = 2; /* GPIO1 = headphones */
+ spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */
+ spec->gpio_mask = spec->gpio_dir =
+ spec->gpio_eapd_hp | spec->gpio_eapd_speaker;
break;
}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 9850c5b481ea..c505fd5d338c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -69,6 +69,7 @@ struct hdmi_spec_per_pin {
struct hda_codec *codec;
struct hdmi_eld sink_eld;
struct delayed_work work;
+ int repoll_count;
};
struct hdmi_spec {
@@ -748,7 +749,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx,
* Unsolicited events
*/
-static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry);
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
{
@@ -766,7 +767,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
if (pin_idx < 0)
return;
- hdmi_present_sense(&spec->pins[pin_idx], true);
+ hdmi_present_sense(&spec->pins[pin_idx], 1);
}
static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -960,7 +961,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx)
return 0;
}
-static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry)
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
{
struct hda_codec *codec = per_pin->codec;
struct hdmi_eld *eld = &per_pin->sink_eld;
@@ -989,7 +990,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry)
if (eld_valid) {
if (!snd_hdmi_get_eld(eld, codec, pin_nid))
snd_hdmi_show_eld(eld);
- else if (retry) {
+ else if (repoll) {
queue_delayed_work(codec->bus->workq,
&per_pin->work,
msecs_to_jiffies(300));
@@ -1004,7 +1005,10 @@ static void hdmi_repoll_eld(struct work_struct *work)
struct hdmi_spec_per_pin *per_pin =
container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work);
- hdmi_present_sense(per_pin, false);
+ if (per_pin->repoll_count++ > 6)
+ per_pin->repoll_count = 0;
+
+ hdmi_present_sense(per_pin, per_pin->repoll_count);
}
static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
@@ -1235,7 +1239,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx)
if (err < 0)
return err;
- hdmi_present_sense(per_pin, false);
+ hdmi_present_sense(per_pin, 0);
return 0;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 336d14eb72af..1d07e8fa2433 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);
@@ -1054,8 +1061,19 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec)
spec->imux_pins[2] = spec->dock_mic_pin;
for (i = 0; i < 3; i++) {
strcpy(imux->items[i].label, texts[i]);
- if (spec->imux_pins[i])
+ if (spec->imux_pins[i]) {
+ hda_nid_t pin = spec->imux_pins[i];
+ int c;
+ for (c = 0; c < spec->num_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;
+ break;
+ }
+ }
imux->num_items = i + 1;
+ }
}
spec->num_mux_defs = 1;
spec->input_mux = imux;
@@ -1957,10 +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;
}
@@ -2615,6 +2631,8 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
case AUTO_PIN_SPEAKER_OUT:
if (cfg->line_outs == 1)
return "Speaker";
+ if (cfg->line_outs == 2)
+ return ch ? "Bass Speaker" : "Speaker";
break;
case AUTO_PIN_HP_OUT:
/* for multi-io case, only the primary out */
@@ -2747,8 +2765,7 @@ 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;
@@ -2889,7 +2906,7 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
if (!nid)
continue;
if (found_in_nid_list(nid, spec->multiout.dac_nids,
- spec->multiout.num_dacs))
+ ARRAY_SIZE(spec->private_dac_nids)))
continue;
if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
ARRAY_SIZE(spec->multiout.hp_out_nid)))
@@ -2910,6 +2927,7 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
return 0;
}
+/* return 0 if no possible DAC is found, 1 if one or more found */
static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
const hda_nid_t *pins, hda_nid_t *dacs)
{
@@ -2927,7 +2945,7 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
if (!dacs[i])
dacs[i] = alc_auto_look_for_dac(codec, pins[i]);
}
- return 0;
+ return 1;
}
static int alc_auto_fill_multi_ios(struct hda_codec *codec,
@@ -2937,7 +2955,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
static int alc_auto_fill_dac_nids(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- const struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
bool redone = false;
int i;
@@ -2948,6 +2966,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
spec->multiout.extra_out_nid[0] = 0;
memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
spec->multiout.dac_nids = spec->private_dac_nids;
+ spec->multi_ios = 0;
/* fill hard-wired DACs first */
if (!redone) {
@@ -2981,10 +3000,12 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
for (i = 0; i < cfg->line_outs; i++) {
if (spec->private_dac_nids[i])
spec->multiout.num_dacs++;
- else
+ else {
memmove(spec->private_dac_nids + i,
spec->private_dac_nids + i + 1,
sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
+ spec->private_dac_nids[cfg->line_outs - 1] = 0;
+ }
}
if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
@@ -3006,9 +3027,28 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
if (cfg->line_out_type != AUTO_PIN_HP_OUT)
alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins,
spec->multiout.hp_out_nid);
- if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT)
- alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins,
- spec->multiout.extra_out_nid);
+ if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs,
+ cfg->speaker_pins,
+ spec->multiout.extra_out_nid);
+ /* if no speaker volume is assigned, try again as the primary
+ * output
+ */
+ if (!err && cfg->speaker_outs > 0 &&
+ cfg->line_out_type == AUTO_PIN_HP_OUT) {
+ cfg->hp_outs = cfg->line_outs;
+ memcpy(cfg->hp_pins, cfg->line_out_pins,
+ sizeof(cfg->hp_pins));
+ cfg->line_outs = cfg->speaker_outs;
+ memcpy(cfg->line_out_pins, cfg->speaker_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->speaker_outs = 0;
+ memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+ cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+ redone = false;
+ goto again;
+ }
+ }
return 0;
}
@@ -3158,7 +3198,8 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
}
static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
- hda_nid_t dac, const char *pfx)
+ hda_nid_t dac, const char *pfx,
+ int cidx)
{
struct alc_spec *spec = codec->spec;
hda_nid_t sw, vol;
@@ -3174,15 +3215,15 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
if (is_ctl_used(spec->sw_ctls, val))
return 0; /* already created */
mark_ctl_usage(spec->sw_ctls, val);
- return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val);
+ return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, cidx, val);
}
sw = alc_look_for_out_mute_nid(codec, pin, dac);
vol = alc_look_for_out_vol_nid(codec, pin, dac);
- err = alc_auto_add_stereo_vol(codec, pfx, 0, vol);
+ err = alc_auto_add_stereo_vol(codec, pfx, cidx, vol);
if (err < 0)
return err;
- err = alc_auto_add_stereo_sw(codec, pfx, 0, sw);
+ err = alc_auto_add_stereo_sw(codec, pfx, cidx, sw);
if (err < 0)
return err;
return 0;
@@ -3223,16 +3264,21 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins,
hda_nid_t dac = *dacs;
if (!dac)
dac = spec->multiout.dac_nids[0];
- return alc_auto_create_extra_out(codec, *pins, dac, pfx);
+ return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0);
}
if (dacs[num_pins - 1]) {
/* OK, we have a multi-output system with individual volumes */
for (i = 0; i < num_pins; i++) {
- snprintf(name, sizeof(name), "%s %s",
- pfx, channel_name[i]);
- err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
- name);
+ if (num_pins >= 3) {
+ snprintf(name, sizeof(name), "%s %s",
+ pfx, channel_name[i]);
+ err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+ name, 0);
+ } else {
+ err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+ pfx, i);
+ }
if (err < 0)
return err;
}
@@ -3694,8 +3740,7 @@ 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 470f6f286e81..616678fde486 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -215,6 +215,7 @@ struct sigmatel_spec {
unsigned int gpio_mute;
unsigned int gpio_led;
unsigned int gpio_led_polarity;
+ unsigned int vref_mute_led_nid; /* pin NID for mute-LED vref control */
unsigned int vref_led;
/* stream */
@@ -1641,6 +1642,8 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
"Alienware M17x", STAC_ALIENWARE_M17X),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
"Alienware M17x", STAC_ALIENWARE_M17X),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
+ "Alienware M17x", STAC_ALIENWARE_M17X),
{} /* terminator */
};
@@ -4316,12 +4319,10 @@ static void stac_store_hints(struct hda_codec *codec)
spec->eapd_switch = val;
get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
- if (spec->gpio_led <= 8) {
- spec->gpio_mask |= spec->gpio_led;
- spec->gpio_dir |= spec->gpio_led;
- if (spec->gpio_led_polarity)
- spec->gpio_data |= spec->gpio_led;
- }
+ spec->gpio_mask |= spec->gpio_led;
+ spec->gpio_dir |= spec->gpio_led;
+ if (spec->gpio_led_polarity)
+ spec->gpio_data |= spec->gpio_led;
}
}
@@ -4439,7 +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->vref_mute_led_nid == nid) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -4911,8 +4914,14 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity)
if (sscanf(dev->name, "HP_Mute_LED_%d_%x",
&spec->gpio_led_polarity,
&spec->gpio_led) == 2) {
- if (spec->gpio_led < 4)
+ unsigned int max_gpio;
+ max_gpio = snd_hda_param_read(codec, codec->afg,
+ AC_PAR_GPIO_CAP);
+ max_gpio &= AC_GPIO_IO_COUNT;
+ if (spec->gpio_led < max_gpio)
spec->gpio_led = 1 << spec->gpio_led;
+ else
+ spec->vref_mute_led_nid = spec->gpio_led;
return 1;
}
if (sscanf(dev->name, "HP_Mute_LED_%d",
@@ -4920,6 +4929,12 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity)
set_hp_led_gpio(codec);
return 1;
}
+ /* BIOS bug: unfilled OEM string */
+ if (strstr(dev->name, "HP_Mute_LED_P_G")) {
+ set_hp_led_gpio(codec);
+ spec->gpio_led_polarity = 1;
+ return 1;
+ }
}
/*
@@ -5041,29 +5056,12 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
struct sigmatel_spec *spec = codec->spec;
/* sync mute LED */
- if (spec->gpio_led) {
- if (spec->gpio_led <= 8) {
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data);
- } else {
- stac_vrefout_set(codec,
- spec->gpio_led, spec->vref_led);
- }
- }
- 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);
- }
+ if (spec->vref_mute_led_nid)
+ stac_vrefout_set(codec, spec->vref_mute_led_nid,
+ spec->vref_led);
+ else if (spec->gpio_led)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data);
return 0;
}
@@ -5074,7 +5072,7 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
struct sigmatel_spec *spec = codec->spec;
if (power_state == AC_PWRST_D3) {
- if (spec->gpio_led > 8) {
+ if (spec->vref_mute_led_nid) {
/* with vref-out pin used for mute led control
* codec AFG is prevented from D3 state
*/
@@ -5127,7 +5125,7 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
}
}
/*polarity defines *not* muted state level*/
- if (spec->gpio_led <= 8) {
+ if (!spec->vref_mute_led_nid) {
if (muted)
spec->gpio_data &= ~spec->gpio_led; /* orange */
else
@@ -5145,7 +5143,8 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
muted_lvl = spec->gpio_led_polarity ?
AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
- stac_vrefout_set(codec, spec->gpio_led, spec->vref_led);
+ stac_vrefout_set(codec, spec->vref_mute_led_nid,
+ spec->vref_led);
}
return 0;
}
@@ -5659,15 +5658,13 @@ again:
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
- if (spec->gpio_led <= 8) {
+ if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
spec->gpio_dir |= spec->gpio_led;
spec->gpio_data |= spec->gpio_led;
} 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 =
@@ -5974,15 +5971,13 @@ again:
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
- if (spec->gpio_led <= 8) {
+ if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
spec->gpio_dir |= spec->gpio_led;
spec->gpio_data |= spec->gpio_led;
} 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/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index 5c8717e29eeb..8c3e7fcefd99 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -78,10 +78,15 @@ unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port)
return ioread32(address);
}
-void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len)
+static void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data,
+ u32 len)
{
- void __iomem *address = lx_dsp_register(chip, port);
- memcpy_fromio(data, address, len*sizeof(u32));
+ u32 __iomem *address = lx_dsp_register(chip, port);
+ int i;
+
+ /* we cannot use memcpy_fromio */
+ for (i = 0; i != len; ++i)
+ data[i] = ioread32(address + i);
}
@@ -91,11 +96,15 @@ void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data)
iowrite32(data, address);
}
-void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data,
- u32 len)
+static void lx_dsp_reg_writebuf(struct lx6464es *chip, int port,
+ const u32 *data, u32 len)
{
- void __iomem *address = lx_dsp_register(chip, port);
- memcpy_toio(address, data, len*sizeof(u32));
+ u32 __iomem *address = lx_dsp_register(chip, port);
+ int i;
+
+ /* we cannot use memcpy_to */
+ for (i = 0; i != len; ++i)
+ iowrite32(data[i], address + i);
}
diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h
index 1dd562980b6c..4d7ff797a646 100644
--- a/sound/pci/lx6464es/lx_core.h
+++ b/sound/pci/lx6464es/lx_core.h
@@ -72,10 +72,7 @@ enum {
};
unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port);
-void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len);
void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data);
-void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data,
- u32 len);
/* plx register access */
enum {
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index e760adad9523..19ee2203cbb5 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6518,7 +6518,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
hdspm->io_type = AES32;
hdspm->card_name = "RME AES32";
hdspm->midiPorts = 2;
- } else if ((hdspm->firmware_rev == 0xd5) ||
+ } else if ((hdspm->firmware_rev == 0xd2) ||
((hdspm->firmware_rev >= 0xc8) &&
(hdspm->firmware_rev <= 0xcf))) {
hdspm->io_type = MADI;
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/Kconfig b/sound/soc/codecs/Kconfig
index 4584514d93d4..fa787d45d74a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -33,7 +33,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
select SND_SOC_DFBMCS320
- select SND_SOC_JZ4740_CODEC if SOC_JZ4740
+ select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
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/adau1373.c b/sound/soc/codecs/adau1373.c
index 1ccf8dd47576..45c63028b40d 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = {
};
static const unsigned int adau1373_bass_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(3),
0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1),
3, 4, TLV_DB_SCALE_ITEM(950, 250, 0),
5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
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/cs4271.c b/sound/soc/codecs/cs4271.c
index 23d1bd5dadda..69fde1506fe1 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
{
int ret;
/* Set power-down bit */
- ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN);
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN,
+ CS4271_MODE2_PDN);
if (ret < 0)
return ret;
return 0;
@@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec)
return ret;
}
- ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
- CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+ ret = snd_soc_update_bits(codec, CS4271_MODE2,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
if (ret < 0)
return ret;
ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
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/jz4740.c b/sound/soc/codecs/jz4740.c
index e373f8f06907..3e1f4e172bfb 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -15,6 +15,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
+#include <linux/io.h>
#include <linux/delay.h>
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/rt5631.c b/sound/soc/codecs/rt5631.c
index 27a078cbb6eb..4646e808b90a 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */
static unsigned int mic_bst_tlv[] = {
- TLV_DB_RANGE_HEAD(6),
+ TLV_DB_RANGE_HEAD(7),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d15695d1c273..bbcf921166f7 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
/* tlv for mic gain, 0db 20db 30db 40db */
static const unsigned int mic_gain_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
};
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index bb82408ab8e1..d2f37152f940 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -76,6 +76,8 @@ struct sta32x_priv {
unsigned int mclk;
unsigned int format;
+
+ u32 coef_shadow[STA32X_COEF_COUNT];
};
static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
@@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
int numcoef = kcontrol->private_value >> 16;
int index = kcontrol->private_value & 0xffff;
unsigned int cfud;
@@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
snd_soc_write(codec, STA32X_CFUD, cfud);
snd_soc_write(codec, STA32X_CFADDR2, index);
+ for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++)
+ sta32x->coef_shadow[index + i] =
+ (ucontrol->value.bytes.data[3 * i] << 16)
+ | (ucontrol->value.bytes.data[3 * i + 1] << 8)
+ | (ucontrol->value.bytes.data[3 * i + 2]);
for (i = 0; i < 3 * numcoef; i++)
snd_soc_write(codec, STA32X_B1CF1 + i,
ucontrol->value.bytes.data[i]);
@@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
return 0;
}
+int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ unsigned int cfud;
+ int i;
+
+ /* preserve reserved bits in STA32X_CFUD */
+ cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0;
+
+ for (i = 0; i < STA32X_COEF_COUNT; i++) {
+ snd_soc_write(codec, STA32X_CFADDR2, i);
+ snd_soc_write(codec, STA32X_B1CF1,
+ (sta32x->coef_shadow[i] >> 16) & 0xff);
+ snd_soc_write(codec, STA32X_B1CF2,
+ (sta32x->coef_shadow[i] >> 8) & 0xff);
+ snd_soc_write(codec, STA32X_B1CF3,
+ (sta32x->coef_shadow[i]) & 0xff);
+ /* chip documentation does not say if the bits are
+ * self-clearing, so do it explicitly */
+ snd_soc_write(codec, STA32X_CFUD, cfud);
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x01);
+ }
+ return 0;
+}
+
+int sta32x_cache_sync(struct snd_soc_codec *codec)
+{
+ unsigned int mute;
+ int rc;
+
+ if (!codec->cache_sync)
+ return 0;
+
+ /* mute during register sync */
+ mute = snd_soc_read(codec, STA32X_MMUTE);
+ snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE);
+ sta32x_sync_coef_shadow(codec);
+ rc = snd_soc_cache_sync(codec);
+ snd_soc_write(codec, STA32X_MMUTE, mute);
+ return rc;
+}
+
#define SINGLE_COEF(xname, index) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = sta32x_coefficient_info, \
@@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
- snd_soc_cache_sync(codec);
+ sta32x_cache_sync(codec);
}
/* Power up to mute */
@@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec)
STA32X_CxCFG_OM_MASK,
2 << STA32X_CxCFG_OM_SHIFT);
+ /* initialize coefficient shadow RAM with reset values */
+ for (i = 4; i <= 49; i += 5)
+ sta32x->coef_shadow[i] = 0x400000;
+ for (i = 50; i <= 54; i++)
+ sta32x->coef_shadow[i] = 0x7fffff;
+ sta32x->coef_shadow[55] = 0x5a9df7;
+ sta32x->coef_shadow[56] = 0x7fffff;
+ sta32x->coef_shadow[59] = 0x7fffff;
+ sta32x->coef_shadow[60] = 0x400000;
+ sta32x->coef_shadow[61] = 0x400000;
+
sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Bias level configuration will have done an extra enable */
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
index b97ee5a75667..d8e32a6262ee 100644
--- a/sound/soc/codecs/sta32x.h
+++ b/sound/soc/codecs/sta32x.h
@@ -19,6 +19,7 @@
/* STA326 register addresses */
#define STA32X_REGISTER_COUNT 0x2d
+#define STA32X_COEF_COUNT 62
#define STA32X_CONFA 0x00
#define STA32X_CONFB 0x01
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index c5ca8cfea60f..0441893e270e 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = {
static int __init uda1380_modinit(void)
{
- int ret;
+ int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&uda1380_i2c_driver);
if (ret != 0)
pr_err("Failed to register UDA1380 I2C driver: %d\n", ret);
#endif
- return 0;
+ return ret;
}
module_init(uda1380_modinit);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7e5ec03f6f8d..a7c9ae17fc7e 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8731_PWR, 0xffff);
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
+ codec->cache_sync = 1;
break;
}
codec->dapm.bias_level = level;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a9504710bb69..3a629d0d690e 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
u16 ioctl;
+ if (wm8753->dai_func == ucontrol->value.integer.value[0])
+ return 0;
+
if (codec->active)
return -EBUSY;
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 0293763debe5..5a14d5c0e0e1 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -60,6 +60,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
}
if (memcmp(fw->data, "WMFW", 4) != 0) {
+ memcpy(&data32, fw->data, sizeof(data32));
+ data32 = be32_to_cpu(data32);
dev_err(codec->dev, "%s: firmware has bad file magic %08x\n",
name, data32);
goto err;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 91d3c6dbeba3..53edd9a8c758 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec)
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0);
static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0);
static const unsigned int mixinpga_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(5),
0, 1, TLV_DB_SCALE_ITEM(0, 600, 0),
2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0),
3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0),
@@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0);
static const unsigned int classd_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index eec8e1435116..d1a142f48b09 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0);
static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0);
static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
static const unsigned int drc_max_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
};
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e47eb99..d0c545b73d78 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1325,15 +1325,15 @@ SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
};
static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
-SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
- adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
- adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
};
static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
@@ -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/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 645c980d6b80..a33b04d17195 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1968,6 +1968,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
break;
case 24576000:
ratediv = WM8996_SYSCLK_DIV;
+ wm8996->sysclk /= 2;
case 12288000:
snd_soc_update_bits(codec, WM8996_AIF_RATE,
WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE);
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 3cd35a02c28c..4a398c3bfe84 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
mdelay(100);
/* Normal bias enable & soft start off */
- reg |= WM9081_BIAS_ENA;
reg &= ~WM9081_VMID_RAMP;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -818,7 +817,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
}
/* VMID 2*240k */
- reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
reg &= ~WM9081_VMID_SEL_MASK;
reg |= 0x04;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
- /* Startup bias source */
+ /* Startup bias source and disable bias */
reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg |= WM9081_BIAS_SRC;
+ reg &= ~WM9081_BIAS_ENA;
snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
- /* Disable VMID and biases with soft ramping */
+ /* Disable VMID with soft ramping */
reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
- reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+ reg &= ~WM9081_VMID_SEL_MASK;
reg |= WM9081_VMID_RAMP;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 2b5252c9e377..f94c06057c64 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
}
static const unsigned int in_tlv[] = {
- TLV_DB_RANGE_HEAD(6),
+ TLV_DB_RANGE_HEAD(3),
0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0),
4, 6, TLV_DB_SCALE_ITEM(600, 600, 0),
};
static const unsigned int mix_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0),
3, 3, TLV_DB_SCALE_ITEM(0, 0, 0),
};
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 84f33d4ea2cd..48e61e912400 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1);
static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0268cf989736..83c4bd5b2dd7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
+ sysfs_attr_init(&dev_attr->attr);
dev_attr->attr.name = "statistics";
dev_attr->attr.mode = S_IRUGO;
dev_attr->show = fsl_sysfs_ssi_show;
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/imx/Kconfig b/sound/soc/imx/Kconfig
index b133bfcc5848..738391757f2c 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1
config SND_SOC_MX27VIS_AIC32X4
tristate "SoC audio support for Visstrim M10 boards"
- depends on MACH_IMX27_VISSTRIM_M10
+ depends on MACH_IMX27_VISSTRIM_M10 && I2C
select SND_SOC_TLV320AIC32X4
select SND_MXC_SOC_MX2
help
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 8f49e165f4d1..c62d715235e2 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S
config SND_KIRKWOOD_SOC_OPENRD
tristate "SoC Audio support for Kirkwood Openrd Client"
depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
+ depends on I2C
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_CS42L51
help
@@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD
config SND_KIRKWOOD_SOC_T5325
tristate "SoC Audio support for HP t5325"
- depends on SND_KIRKWOOD_SOC && MACH_T5325
+ depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_ALC5623
help
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index dea5aa4aa647..f39d7dd9fbcb 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -357,3 +357,6 @@ static void __exit snd_mxs_pcm_exit(void)
platform_driver_unregister(&mxs_pcm_driver);
}
module_exit(snd_mxs_pcm_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mxs-pcm-audio");
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 7fbeaec06eb4..1c57f6630a48 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -171,3 +171,4 @@ module_exit(mxs_sgtl5000_exit);
MODULE_AUTHOR("Freescale Semiconductor, Inc.");
MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mxs-sgtl5000");
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 9c0edad90d8b..a4e3237956e2 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -365,7 +365,8 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
if (ret)
goto out3;
- mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/
+ /* enbale ac97 multifunction pin */
+ mfp_set_groupg(nuc900_audio->dev, "nuc900-audio");
return 0;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ffd2242e305f..a0f7d3cfa470 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE
config SND_SOC_RAUMFELD
tristate "SoC Audio support Raumfeld audio adapter"
depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+ depends on I2C && SPI_MASTER
select SND_PXA_SOC_SSP
select SND_SOC_CS4270
select SND_SOC_AK4104
@@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD
config SND_PXA2XX_SOC_HX4700
tristate "SoC Audio support for HP iPAQ hx4700"
- depends on SND_PXA2XX_SOC && MACH_H4700
+ depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_AK4641
help
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 65c124831a00..c664e33fb6d7 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -209,9 +209,10 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev)
snd_soc_card_hx4700.dev = &pdev->dev;
ret = snd_soc_register_card(&snd_soc_card_hx4700);
if (ret)
- return ret;
+ gpio_free_array(hx4700_audio_gpios,
+ ARRAY_SIZE(hx4700_audio_gpios));
- return 0;
+ return ret;
}
static int __devexit hx4700_audio_remove(struct platform_device *pdev)
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 1826acf20f7c..8e523fd9189e 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -101,7 +101,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
/* These endpoints are not being used. */
snd_soc_dapm_nc_pin(dapm, "LINPUT2");
@@ -131,7 +130,7 @@ static struct snd_soc_card snd_soc_machine_jive = {
.dai_link = &jive_dai,
.num_links = 1,
- .dapm_widgtets = wm8750_dapm_widgets,
+ .dapm_widgets = wm8750_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c
index 3a0dbfc793f0..8bd1dc5706bf 100644
--- a/sound/soc/samsung/smdk2443_wm9710.c
+++ b/sound/soc/samsung/smdk2443_wm9710.c
@@ -12,6 +12,7 @@
*
*/
+#include <linux/module.h>
#include <sound/soc.h>
static struct snd_soc_card smdk2443;
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/soc/soc-utils.c b/sound/soc/soc-utils.c
index 0c12b98484bd..4220bb0f2730 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
-static struct snd_soc_platform_driver dummy_platform;
+static const struct snd_pcm_hardware dummy_dma_hardware = {
+ .formats = 0xffffffff,
+ .channels_min = 1,
+ .channels_max = UINT_MAX,
+
+ /* Random values to keep userspace happy when checking constraints */
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .buffer_bytes_max = 128*1024,
+ .period_bytes_min = PAGE_SIZE,
+ .period_bytes_max = PAGE_SIZE*2,
+ .periods_min = 2,
+ .periods_max = 128,
+};
+
+static int dummy_dma_open(struct snd_pcm_substream *substream)
+{
+ snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
+
+ return 0;
+}
+
+static struct snd_pcm_ops dummy_dma_ops = {
+ .open = dummy_dma_open,
+ .ioctl = snd_pcm_lib_ioctl,
+};
+
+static struct snd_soc_platform_driver dummy_platform = {
+ .ops = &dummy_dma_ops,
+};
static __devinit int snd_soc_dummy_probe(struct platform_device *pdev)
{
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", */