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-rw-r--r--sound/soc/codecs/88pm860x-codec.c3
-rw-r--r--sound/soc/codecs/ab8500-codec.c7
-rw-r--r--sound/soc/codecs/adau1701.c2
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/arizona.c4
-rw-r--r--sound/soc/codecs/cs42l52.c2
-rw-r--r--sound/soc/codecs/cs42l52.h2
-rw-r--r--sound/soc/codecs/da732x.c12
-rw-r--r--sound/soc/codecs/max98088.c2
-rw-r--r--sound/soc/codecs/max98090.c20
-rw-r--r--sound/soc/codecs/max98095.c4
-rw-r--r--sound/soc/codecs/mc13783.c4
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sgtl5000.h2
-rw-r--r--sound/soc/codecs/sta32x.c76
-rw-r--r--sound/soc/codecs/wm5110.c93
-rw-r--r--sound/soc/codecs/wm8731.c4
-rw-r--r--sound/soc/codecs/wm8770.c4
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8960.c6
-rw-r--r--sound/soc/codecs/wm8962.c26
-rw-r--r--sound/soc/codecs/wm8990.c2
-rw-r--r--sound/soc/codecs/wm_adsp.c10
-rw-r--r--sound/soc/codecs/wm_hubs.c1
25 files changed, 211 insertions, 83 deletions
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 60159c07448d..6fd174be3bdf 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -351,6 +351,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
val = ucontrol->value.integer.value[0];
val2 = ucontrol->value.integer.value[1];
+ if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table))
+ return -EINVAL;
+
err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
if (err < 0)
return err;
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index a153b168129b..bce45c197e1d 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
struct device *dev = codec->dev;
bool apply_fir, apply_iir;
- int req, status;
+ unsigned int req;
+ int status;
dev_dbg(dev, "%s: Enter.\n", __func__);
mutex_lock(&drvdata->anc_lock);
req = ucontrol->value.integer.value[0];
+ if (req >= ARRAY_SIZE(enum_anc_state)) {
+ status = -EINVAL;
+ goto cleanup;
+ }
if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
req != ANC_APPLY_IIR) {
dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index dafdbe87edeb..0c499c638692 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -64,7 +64,7 @@
#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000
#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001
-#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010
+#define ADAU1701_SEROCTL_WORD_LEN_16 0x0002
#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003
#define ADAU1701_AUXNPOW_VBPD 0x40
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 2d0378709702..687565d08d9c 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p94.
*/
- snd_soc_write(codec, SG_SL1, PMMP | MGAIN0);
+ snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 389f23253831..663a2a748626 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1455,6 +1455,8 @@ static void arizona_enable_fll(struct arizona_fll *fll,
try_wait_for_completion(&fll->ok);
regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, 0);
+ regmap_update_bits(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
if (fll->ref_src >= 0 && fll->sync_src >= 0 &&
fll->ref_src != fll->sync_src)
@@ -1473,6 +1475,8 @@ static void arizona_disable_fll(struct arizona_fll *fll)
struct arizona *arizona = fll->arizona;
bool change;
+ regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN);
regmap_update_bits_check(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, 0, &change);
regmap_update_bits(arizona->regmap, fll->base + 0x11,
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 987f728718c5..ee25f325d65c 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("Beep Pitch", beep_pitch_enum),
SOC_ENUM("Beep on Time", beep_ontime_enum),
SOC_ENUM("Beep off Time", beep_offtime_enum),
- SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+ SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv),
SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 4277012c4719..a935d7381af6 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -179,7 +179,7 @@
#define CS42L52_MICB_CTL 0x11
#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF
#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6
-#define CS42L52_MIC_CTL_TYPE_MASK 0xDF
+#define CS42L52_MIC_CTL_TYPE_MASK 0x20
#define CS42L52_MIC_CTL_TYPE_SHIFT 5
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index dc0284dc9e6f..76fdf0a598bc 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = {
},
};
+static bool da732x_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case DA732X_REG_HPL_DAC_OFF_CNTL:
+ case DA732X_REG_HPR_DAC_OFF_CNTL:
+ return true;
+ default:
+ return false;
+ }
+}
+
static const struct regmap_config da732x_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = DA732X_MAX_REG,
+ .volatile_reg = da732x_volatile,
.reg_defaults = da732x_reg_cache,
.num_reg_defaults = ARRAY_SIZE(da732x_reg_cache),
.cache_type = REGCACHE_RBTREE,
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 3eeada57e87d..566a367c94fa 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
static void max98088_sync_cache(struct snd_soc_codec *codec)
{
- u16 *reg_cache = codec->reg_cache;
+ u8 *reg_cache = codec->reg_cache;
int i;
if (!codec->cache_sync)
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 8d14a76c7249..819c90fe021f 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1755,16 +1755,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
- if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
- ret = regcache_sync(max98090->regmap);
-
- if (ret != 0) {
- dev_err(codec->dev,
- "Failed to sync cache: %d\n", ret);
- return ret;
- }
- }
-
if (max98090->jack_state == M98090_JACK_STATE_HEADSET) {
/*
* Set to normal bias level.
@@ -1778,6 +1768,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regcache_sync(max98090->regmap);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
case SND_SOC_BIAS_OFF:
/* Set internal pull-up to lowest power mode */
snd_soc_update_bits(codec, M98090_REG_JACK_DETECT,
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 41cdd1642970..8dbcacd44e6a 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
struct max98095_pdata *pdata = max98095->pdata;
int channel = max98095_get_eq_channel(kcontrol->id.name);
struct max98095_cdata *cdata;
- int sel = ucontrol->value.integer.value[0];
+ unsigned int sel = ucontrol->value.integer.value[0];
struct max98095_eq_cfg *coef_set;
int fs, best, best_val, i;
int regmask, regsave;
@@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
struct max98095_pdata *pdata = max98095->pdata;
int channel = max98095_get_bq_channel(codec, kcontrol->id.name);
struct max98095_cdata *cdata;
- int sel = ucontrol->value.integer.value[0];
+ unsigned int sel = ucontrol->value.integer.value[0];
struct max98095_biquad_cfg *coef_set;
int fs, best, best_val, i;
int regmask, regsave;
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 5402dfbbb716..8a8d9364e87f 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -126,6 +126,10 @@ static int mc13783_write(struct snd_soc_codec *codec,
ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+ /* include errata fix for spi audio problems */
+ if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC)
+ ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+
mc13xxx_unlock(priv->mc13xxx);
return ret;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 92bbfec9b107..ea479388fb5c 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -37,7 +37,7 @@
static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = {
[SGTL5000_CHIP_CLK_CTRL] = 0x0008,
[SGTL5000_CHIP_I2S_CTRL] = 0x0010,
- [SGTL5000_CHIP_SSS_CTRL] = 0x0008,
+ [SGTL5000_CHIP_SSS_CTRL] = 0x0010,
[SGTL5000_CHIP_DAC_VOL] = 0x3c3c,
[SGTL5000_CHIP_PAD_STRENGTH] = 0x015f,
[SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818,
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 8a9f43534b79..d3a68bbfea00 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -347,7 +347,7 @@
#define SGTL5000_PLL_INT_DIV_MASK 0xf800
#define SGTL5000_PLL_INT_DIV_SHIFT 11
#define SGTL5000_PLL_INT_DIV_WIDTH 5
-#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700
+#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff
#define SGTL5000_PLL_FRAC_DIV_SHIFT 0
#define SGTL5000_PLL_FRAC_DIV_WIDTH 11
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index cfb55fe35e98..8517e70bc24b 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = {
13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
};
-static const struct soc_enum sta32x_drc_ac_enum =
- SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
- 2, sta32x_drc_ac);
-static const struct soc_enum sta32x_auto_eq_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
- 3, sta32x_auto_eq_mode);
-static const struct soc_enum sta32x_auto_gc_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
- 4, sta32x_auto_gc_mode);
-static const struct soc_enum sta32x_auto_xo_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
- 16, sta32x_auto_xo_mode);
-static const struct soc_enum sta32x_preset_eq_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
- 32, sta32x_preset_eq_mode);
-static const struct soc_enum sta32x_limiter_ch1_enum =
- SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch2_enum =
- SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch3_enum =
- SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter1_attack_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
- 16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter2_attack_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
- 16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter1_release_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
- 16, sta32x_limiter_release_rate);
-static const struct soc_enum sta32x_limiter2_release_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
- 16, sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum,
+ STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+ sta32x_drc_ac);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum,
+ STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+ sta32x_auto_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum,
+ STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+ sta32x_auto_gc_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum,
+ STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+ sta32x_auto_xo_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum,
+ STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+ sta32x_preset_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum,
+ STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum,
+ STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum,
+ STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum,
+ STA32X_L1AR, STA32X_LxA_SHIFT,
+ sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum,
+ STA32X_L2AR, STA32X_LxA_SHIFT,
+ sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum,
+ STA32X_L1AR, STA32X_LxR_SHIFT,
+ sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum,
+ STA32X_L2AR, STA32X_LxR_SHIFT,
+ sta32x_limiter_release_rate);
/* byte array controls for setting biquad, mixer, scaling coefficients;
* for biquads all five coefficients need to be set in one go,
@@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
static int sta32x_cache_sync(struct snd_soc_codec *codec)
{
- struct sta32x_priv *sta32x = codec->control_data;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
unsigned int mute;
int rc;
@@ -432,7 +432,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0,
SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum),
SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum),
SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
-SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum),
/* depending on mode, the attack/release thresholds have
* two different enum definitions; provide both
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 88ad7db52dde..3775394c9c8b 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -37,6 +37,95 @@ struct wm5110_priv {
struct arizona_fll fll[2];
};
+static const struct reg_default wm5110_sysclk_revd_patch[] = {
+ { 0x3093, 0x1001 },
+ { 0x30E3, 0x1301 },
+ { 0x3133, 0x1201 },
+ { 0x3183, 0x1501 },
+ { 0x31D3, 0x1401 },
+ { 0x0049, 0x01ea },
+ { 0x004a, 0x01f2 },
+ { 0x0057, 0x01e7 },
+ { 0x0058, 0x01fb },
+ { 0x33ce, 0xc4f5 },
+ { 0x33cf, 0x1361 },
+ { 0x33d0, 0x0402 },
+ { 0x33d1, 0x4700 },
+ { 0x33d2, 0x026d },
+ { 0x33d3, 0xff00 },
+ { 0x33d4, 0x026d },
+ { 0x33d5, 0x0101 },
+ { 0x33d6, 0xc4f5 },
+ { 0x33d7, 0x0361 },
+ { 0x33d8, 0x0402 },
+ { 0x33d9, 0x6701 },
+ { 0x33da, 0xc4f5 },
+ { 0x33db, 0x136f },
+ { 0x33dc, 0xc4f5 },
+ { 0x33dd, 0x134f },
+ { 0x33de, 0xc4f5 },
+ { 0x33df, 0x131f },
+ { 0x33e0, 0x026d },
+ { 0x33e1, 0x4f01 },
+ { 0x33e2, 0x026d },
+ { 0x33e3, 0xf100 },
+ { 0x33e4, 0x026d },
+ { 0x33e5, 0x0001 },
+ { 0x33e6, 0xc4f5 },
+ { 0x33e7, 0x0361 },
+ { 0x33e8, 0x0402 },
+ { 0x33e9, 0x6601 },
+ { 0x33ea, 0xc4f5 },
+ { 0x33eb, 0x136f },
+ { 0x33ec, 0xc4f5 },
+ { 0x33ed, 0x134f },
+ { 0x33ee, 0xc4f5 },
+ { 0x33ef, 0x131f },
+ { 0x33f0, 0x026d },
+ { 0x33f1, 0x4e01 },
+ { 0x33f2, 0x026d },
+ { 0x33f3, 0xf000 },
+ { 0x33f6, 0xc4f5 },
+ { 0x33f7, 0x1361 },
+ { 0x33f8, 0x0402 },
+ { 0x33f9, 0x4600 },
+ { 0x33fa, 0x026d },
+ { 0x33fb, 0xfe00 },
+};
+
+static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct regmap *regmap = codec->control_data;
+ const struct reg_default *patch = NULL;
+ int i, patch_size;
+
+ switch (arizona->rev) {
+ case 3:
+ patch = wm5110_sysclk_revd_patch;
+ patch_size = ARRAY_SIZE(wm5110_sysclk_revd_patch);
+ break;
+ default:
+ return 0;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (patch)
+ for (i = 0; i < patch_size; i++)
+ regmap_write(regmap, patch[i].reg,
+ patch[i].def);
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
@@ -386,7 +475,7 @@ static const struct snd_kcontrol_new wm5110_aec_loopback_mux =
static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
- 0, NULL, 0),
+ 0, wm5110_sysclk_ev, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
@@ -856,7 +945,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
{ "HPOUT2R", NULL, "OUT2R" },
{ "HPOUT3L", NULL, "OUT3L" },
- { "HPOUT3R", NULL, "OUT3L" },
+ { "HPOUT3R", NULL, "OUT3R" },
{ "SPKOUTLN", NULL, "OUT4L" },
{ "SPKOUTLP", NULL, "OUT4L" },
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5276062d6c79..10d492b6a5b4 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -407,10 +407,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
+ iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0013;
+ iface |= 0x0003;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 89a18d82f303..5bce21013485 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -196,8 +196,8 @@ static const char *ain_text[] = {
"AIN5", "AIN6", "AIN7", "AIN8"
};
-static const struct soc_enum ain_enum =
- SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text);
+static SOC_ENUM_DOUBLE_DECL(ain_enum,
+ WM8770_ADCMUX, 0, 4, ain_text);
static const struct snd_kcontrol_new ain_mux =
SOC_DAPM_ENUM("Capture Mux", ain_enum);
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3ff195c541db..af62f843a691 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1449,7 +1449,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
- aif1 |= WM8904_AIF_LRCLK_INV;
+ aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
case SND_SOC_DAIFMT_DSP_A:
aif1 |= 0x3;
break;
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index b0710d817a65..754f88e1fdab 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
data32 &= 0xffffff;
- wm8994_bulk_write(codec->control_data,
+ wm8994_bulk_write(wm8994->wm8994,
data32 & 0xffffff,
block_len / 2,
(void *)(data + 8));
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0a4ffdd1d2a7..5e5af898f7f8 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
if (pll_div.k) {
reg |= 0x20;
- snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
- snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
- snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff);
+ snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff);
+ snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff);
}
snd_soc_write(codec, WM8960_PLL1, reg);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index e9710280e5e1..e3cd86514cea 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1600,7 +1600,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- u16 *reg_cache = codec->reg_cache;
int ret;
/* Apply the update (if any) */
@@ -1609,16 +1608,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
return 0;
/* If the left PGA is enabled hit that VU bit... */
- if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
- return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
- reg_cache[WM8962_HPOUTL_VOLUME]);
+ ret = snd_soc_read(codec, WM8962_PWR_MGMT_2);
+ if (ret & WM8962_HPOUTL_PGA_ENA) {
+ snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
+ snd_soc_read(codec, WM8962_HPOUTL_VOLUME));
+ return 1;
+ }
/* ...otherwise the right. The VU is stereo. */
- if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
- return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
- reg_cache[WM8962_HPOUTR_VOLUME]);
+ if (ret & WM8962_HPOUTR_PGA_ENA)
+ snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
+ snd_soc_read(codec, WM8962_HPOUTR_VOLUME));
- return 0;
+ return 1;
}
/* The VU bits for the speakers are in a different register to the mute
@@ -3374,7 +3376,6 @@ static int wm8962_probe(struct snd_soc_codec *codec)
int ret;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
- u16 *reg_cache = codec->reg_cache;
int i, trigger, irq_pol;
bool dmicclk, dmicdat;
@@ -3432,8 +3433,9 @@ static int wm8962_probe(struct snd_soc_codec *codec)
/* Put the speakers into mono mode? */
if (pdata->spk_mono)
- reg_cache[WM8962_CLASS_D_CONTROL_2]
- |= WM8962_SPK_MONO;
+ snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2,
+ WM8962_SPK_MONO_MASK, WM8962_SPK_MONO);
+
/* Micbias setup, detection enable and detection
* threasholds. */
@@ -3684,6 +3686,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
if (ret < 0)
goto err_enable;
+ regcache_cache_only(wm8962->regmap, true);
+
/* The drivers should power up as needed */
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 837978e16e9d..ded9ed854a1f 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1264,6 +1264,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+ codec->cache_sync = 1;
break;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 3470b649c0b2..6dbb17d050c9 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1073,13 +1073,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
return ret;
/* Wait for the RAM to start, should be near instantaneous */
- count = 0;
- do {
+ for (count = 0; count < 10; ++count) {
ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
&val);
if (ret != 0)
return ret;
- } while (!(val & ADSP2_RAM_RDY) && ++count < 10);
+
+ if (val & ADSP2_RAM_RDY)
+ break;
+
+ msleep(1);
+ }
if (!(val & ADSP2_RAM_RDY)) {
adsp_err(dsp, "Failed to start DSP RAM\n");
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index f5d81b948759..7a0466eb7ede 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -530,6 +530,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w,
hubs->hp_startup_mode);
break;
}
+ break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-28 10:52:40 +0000