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-rw-r--r--sound/atmel/abdac.c2
-rw-r--r--sound/atmel/ac97c.c2
-rw-r--r--sound/pci/asihpi/asihpi.c1
-rw-r--r--sound/pci/cs5535audio/cs5535audio_pcm.c4
-rw-r--r--sound/pci/hda/hda_eld.c2
-rw-r--r--sound/pci/hda/patch_conexant.c4
-rw-r--r--sound/pci/hda/patch_realtek.c46
-rw-r--r--sound/pci/hda/patch_via.c35
-rw-r--r--sound/pci/rme9652/hdspm.c8
-rw-r--r--sound/soc/blackfin/bf5xx-i2s-pcm.c13
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/tlv320aic26.c14
-rw-r--r--sound/soc/codecs/tlv320aic3x.c9
-rw-r--r--sound/soc/codecs/wm8731.c29
-rw-r--r--sound/soc/codecs/wm8991.c1
-rw-r--r--sound/soc/codecs/wm8994.c2
-rw-r--r--sound/soc/imx/Kconfig7
-rw-r--r--sound/soc/imx/imx-pcm-dma-mx2.c2
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c4
-rw-r--r--sound/soc/soc-cache.c3
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/tegra/tegra_i2s.c6
-rw-r--r--sound/spi/at73c213.c2
24 files changed, 125 insertions, 80 deletions
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 6e240918189..bfee60c4d4c 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -599,4 +599,4 @@ module_exit(atmel_abdac_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index b310702c646..ac35222ad0d 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1199,4 +1199,4 @@ module_exit(atmel_ac97c_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Driver for Atmel AC97 controller");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 2ca6f4f85b4..e3569bdd3b6 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -27,7 +27,6 @@
#include "hpioctl.h"
#include <linux/pci.h>
-#include <linux/version.h>
#include <linux/init.h>
#include <linux/jiffies.h>
#include <linux/slab.h>
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index f16bc8aad6e..e083122ca55 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -149,7 +149,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
&((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
desc->addr = cpu_to_le32(addr);
desc->size = cpu_to_le32(period_bytes);
- desc->ctlreserved = cpu_to_le32(PRD_EOP);
+ desc->ctlreserved = cpu_to_le16(PRD_EOP);
desc_addr += sizeof(struct cs5535audio_dma_desc);
addr += period_bytes;
}
@@ -157,7 +157,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods];
lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr);
lastdesc->size = 0;
- lastdesc->ctlreserved = cpu_to_le32(PRD_JMP);
+ lastdesc->ctlreserved = cpu_to_le16(PRD_JMP);
jmpprd_addr = cpu_to_le32(lastdesc->addr +
(sizeof(struct cs5535audio_dma_desc)*periods));
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b05f7be9dc1..e3e853153d1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_eld *e,
snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
goto out_fail;
} else
- strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl);
+ strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1);
for (i = 0; i < e->sad_count; i++) {
if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 694b9daf691..7bbc5f237a5 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3074,6 +3074,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
};
static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
@@ -4389,6 +4390,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5066 },
{ .id = 0x14f15069, .name = "CX20585",
.patch = patch_cxt5066 },
+ { .id = 0x14f1506c, .name = "CX20588",
+ .patch = patch_cxt5066 },
{ .id = 0x14f1506e, .name = "CX20590",
.patch = patch_cxt5066 },
{ .id = 0x14f15097, .name = "CX20631",
@@ -4417,6 +4420,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
MODULE_ALIAS("snd-hda-codec-id:14f15067");
MODULE_ALIAS("snd-hda-codec-id:14f15068");
MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506c");
MODULE_ALIAS("snd-hda-codec-id:14f1506e");
MODULE_ALIAS("snd-hda-codec-id:14f15097");
MODULE_ALIAS("snd-hda-codec-id:14f15098");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 61a774b3d3c..b48fb43b544 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
- getput_call_t func)
+ getput_call_t func, bool check_adc_switch)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- int err;
+ int i, err = 0;
mutex_lock(&codec->control_mutex);
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
- 3, 0, HDA_INPUT);
- err = func(kcontrol, ucontrol);
+ if (check_adc_switch && spec->dual_adc_switch) {
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
+ err = func(kcontrol, ucontrol);
+ if (err < 0)
+ goto error;
+ }
+ } else {
+ i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
+ err = func(kcontrol, ucontrol);
+ }
+ error:
mutex_unlock(&codec->control_mutex);
return err;
}
@@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_volume_get);
+ snd_hda_mixer_amp_volume_get, false);
}
static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_volume_put);
+ snd_hda_mixer_amp_volume_put, true);
}
/* capture mixer elements */
@@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_switch_get);
+ snd_hda_mixer_amp_switch_get, false);
}
static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_switch_put);
+ snd_hda_mixer_amp_switch_put, true);
}
#define _DEFINE_CAPMIX(num) \
@@ -4883,7 +4896,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
@@ -12600,6 +12612,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
*/
enum {
PINFIX_FSC_H270,
+ PINFIX_HP_Z200,
};
static const struct alc_fixup alc262_fixups[] = {
@@ -12612,9 +12625,17 @@ static const struct alc_fixup alc262_fixups[] = {
{ }
}
},
+ [PINFIX_HP_Z200] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x16, 0x99130120 }, /* internal speaker */
+ { }
+ }
+ },
};
static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
{}
};
@@ -12731,6 +12752,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = {
ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+ ALC262_AUTO),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -13872,7 +13895,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
- SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
{}
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index c952582fb21..f43bb0eaed8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
struct via_spec *spec = codec->spec;
hda_nid_t nid = kcontrol->private_value;
unsigned int pinsel = ucontrol->value.enumerated.item[0];
+ unsigned int parm0, parm1;
/* Get Independent Mode index of headphone pin widget */
spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
? 1 : 0;
- if (spec->codec_type == VT1718S)
+ if (spec->codec_type == VT1718S) {
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+ /* Set correct mute switch for MW3 */
+ parm0 = spec->hp_independent_mode ?
+ AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
+ parm1 = spec->hp_independent_mode ?
+ AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+ snd_hda_codec_write(codec, 0x1b, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, parm0);
+ snd_hda_codec_write(codec, 0x1b, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, parm1);
+ }
else
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, pinsel);
@@ -4283,9 +4294,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-
- /* Setup default input of Front HP to MW9 */
- {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
/* PW9 PW10 Output enable */
{0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
{0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
@@ -4294,10 +4302,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
/* Enable Boost Volume backdoor */
{0x1, 0xf88, 0x8},
/* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4307,8 +4315,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
/* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
{0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
{0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Unmute MW4's index 0 */
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ }
};
@@ -4456,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
if (err < 0)
return err;
} else if (i == AUTO_SEQ_FRONT) {
+ /* add control to mixer index 0 */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
/* Front */
sprintf(name, "%s Playback Volume", chname[i]);
err = via_add_control(
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 3f08afc0f0d..c8e402fc378 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -896,11 +896,11 @@ struct hdspm {
unsigned char max_channels_in;
unsigned char max_channels_out;
- char *channel_map_in;
- char *channel_map_out;
+ signed char *channel_map_in;
+ signed char *channel_map_out;
- char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
- char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
+ signed char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
+ signed char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
char **port_names_in;
char **port_names_out;
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101efd1c8..f1fd95bb641 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
pr_debug("%s enter\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff = sport_curr_offset_tx(sport);
- frames = bytes_to_frames(substream->runtime, diff);
} else {
diff = sport_curr_offset_rx(sport);
- frames = bytes_to_frames(substream->runtime, diff);
}
+
+ /*
+ * TX at least can report one frame beyond the end of the
+ * buffer if we hit the wraparound case - clamp to within the
+ * buffer as the ALSA APIs require.
+ */
+ if (diff == snd_pcm_lib_buffer_bytes(substream))
+ diff = 0;
+
+ frames = bytes_to_frames(substream->runtime, diff);
+
return frames;
}
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1..65f46047b1c 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
default:
return -EINVAL;
}
- snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+ snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
/* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608d394..7859bdcc93d 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
}
- /* Configure PLL */
+ /**
+ * Configure PLL
+ * fsref = (mclk * PLLM) / 2048
+ * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+ */
pval = 1;
- jval = (fsref == 44100) ? 7 : 8;
- dval = (fsref == 44100) ? 5264 : 1920;
+ /* compute J portion of multiplier */
+ jval = fsref / (aic26->mclk / 2048);
+ /* compute fractional DDDD component of multiplier */
+ dval = fsref - (jval * (aic26->mclk / 2048));
+ dval = (10000 * dval) / (aic26->mclk / 2048);
+ dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
qval = 0;
reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc8c26..789453d44ec 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
/* Sync reg_cache with the hardware */
codec->cache_only = 0;
- for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+ for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
snd_soc_write(codec, i, cache[i]);
if (aic3x->model == AIC3X_MODEL_3007)
aic3x_init_3007(codec);
codec->cache_sync = 0;
} else {
+ /*
+ * Do soft reset to this codec instance in order to clear
+ * possible VDD leakage currents in case the supply regulators
+ * remain on
+ */
+ snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+ codec->cache_sync = 1;
aic3x->power = 0;
/* HW writes are needless when bias is off */
codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b55e4..76b4361e9b8 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
&wm8731_output_mixer_controls[0],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
static const struct snd_soc_dapm_route wm8731_intercon[] = {
{"DAC", NULL, "OSC", wm8731_check_osc},
{"ADC", NULL, "OSC", wm8731_check_osc},
+ {"DAC", NULL, "ACTIVE"},
+ {"ADC", NULL, "ACTIVE"},
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
-
- /* set active */
- snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
- return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
-
- /* deactivate */
- if (!codec->active) {
- udelay(50);
- snd_soc_write(codec, WM8731_ACTIVE, 0x0);
- }
-}
-
static int wm8731_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
break;
case SND_SOC_BIAS_OFF:
- snd_soc_write(codec, WM8731_ACTIVE, 0x0);
snd_soc_write(codec, WM8731_PWR, 0xffff);
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops wm8731_dai_ops = {
- .prepare = wm8731_pcm_prepare,
.hw_params = wm8731_hw_params,
- .shutdown = wm8731_shutdown,
.digital_mute = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
.set_fmt = wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73c..6af23d06870 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360..c2fc0356c2a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
reg);
+
+ msleep(5);
}
wm8994->fll[id].in = freq_in;
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd..bb699bb55a5 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
if SND_IMX_SOC
-config SND_MXC_SOC_SSI
- tristate
-
config SND_MXC_SOC_FIQ
tristate
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
select SND_SOC_WM8350
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
help
Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
tristate "SoC audio support for Visstrim M10 boards"
depends on MACH_IMX27_VISSTRIM_M10
select SND_SOC_TVL320AIC32X4
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_MX2
help
Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
depends on MACH_PCM043 || MACH_PCA100
select SND_SOC_WM9712
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
help
Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
|| MACH_EUKREA_MBIMXSD51_BASEBOARD
select SND_SOC_TLV320AIC23
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
help
Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index b2ed764fd89..43fdc24f7e8 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
platform_driver_unregister(&imx_pcm_driver);
}
module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca753..61fceb09cdb 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d..fab20a54e86 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- if (dai->driver->playback.channels_min) {
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
goto out;
}
- if (dai->driver->capture.channels_min) {
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index c005ceb70c9..039b9532b27 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
switch (control) {
- case SND_SOC_CUSTOM:
- break;
-
case SND_SOC_I2C:
#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
codec->hw_write = (hw_write_t)i2c_master_send;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043ed7fc..b194be09e74 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->long_name ? card->long_name : card->name);
- snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
- "%s", card->driver_name ? card->driver_name : card->name);
+ if (card->driver_name)
+ strlcpy(card->snd_card->driver, card->driver_name,
+ sizeof(card->snd_card->driver));
if (card->late_probe) {
ret = card->late_probe(card);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548..95f03c10b4f 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
if (i2sclock % (2 * srate))
reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
+ if (!i2s->clk_refs)
+ clk_enable(i2s->clk_i2s);
+
tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+ if (!i2s->clk_refs)
+ clk_disable(i2s->clk_i2s);
+
return 0;
}
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 337a00241a1..4dd051bdf4f 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -1124,6 +1124,6 @@ static void __exit at73c213_exit(void)
}
module_exit(at73c213_exit);
-MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC");
MODULE_LICENSE("GPL");
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-18 18:07:17 +0000