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-rw-r--r--sound/soc/codecs/alc5623.c2
-rw-r--r--sound/soc/codecs/jz4740.c2
-rw-r--r--sound/soc/codecs/lm4857.c2
-rw-r--r--sound/soc/codecs/sn95031.c4
-rw-r--r--sound/soc/codecs/ssm2602.c10
-rw-r--r--sound/soc/codecs/tlv320aic26.h4
-rw-r--r--sound/soc/codecs/tlv320aic3x.c2
-rw-r--r--sound/soc/codecs/tlv320dac33.c2
-rw-r--r--sound/soc/codecs/twl4030.c6
-rw-r--r--sound/soc/codecs/uda134x.c2
-rw-r--r--sound/soc/codecs/wm8580.c2
-rw-r--r--sound/soc/codecs/wm8753.c2
-rw-r--r--sound/soc/codecs/wm8903.c40
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8955.c2
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/codecs/wm8991.c2
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm8994.c22
-rw-r--r--sound/soc/codecs/wm9081.c4
-rw-r--r--sound/soc/codecs/wm_hubs.c8
21 files changed, 73 insertions, 51 deletions
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 4f377c9e868d..eecffb548947 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -481,7 +481,7 @@ struct _pll_div {
};
/* Note : pll code from original alc5623 driver. Not sure of how good it is */
-/* usefull only for master mode */
+/* useful only for master mode */
static const struct _pll_div codec_master_pll_div[] = {
{ 2048000, 8192000, 0x0ea0},
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index f7cd346fd727..f5ccdbf7ebc6 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
ARRAY_SIZE(jz4740_codec_dapm_routes));
- snd_soc_dapm_new_widgets(codec);
-
jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 72de47e5d040..2c2a681da0d7 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
lm4857_get_mode, lm4857_set_mode),
};
-/* There is a demux inbetween the the input signal and the output signals.
+/* There is a demux between the input signal and the output signals.
* Currently there is no easy way to model it in ASoC and since it does not make
* much of a difference in practice simply connect the input direclty to the
* outputs. */
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 2a30eae1881c..4d9fb279e146 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -26,7 +26,9 @@
#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
#include <linux/platform_device.h>
+#include <linux/delay.h>
#include <linux/slab.h>
+
#include <asm/intel_scu_ipc.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -925,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = {
.owner = THIS_MODULE,
},
.probe = sn95031_device_probe,
- .remove = sn95031_device_remove,
+ .remove = __devexit_p(sn95031_device_remove),
};
static int __init sn95031_init(void)
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 2727befd158e..b04d28039c16 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -139,7 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
-SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0),
+SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
@@ -602,7 +602,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.read = ssm2602_read_reg_cache,
.write = ssm2602_write,
.set_bias_level = ssm2602_set_bias_level,
- .reg_cache_size = sizeof(ssm2602_reg),
+ .reg_cache_size = ARRAY_SIZE(ssm2602_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = ssm2602_reg,
};
@@ -614,7 +614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
* low = 0x1a
* high = 0x1b
*/
-static int ssm2602_i2c_probe(struct i2c_client *i2c,
+static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ssm2602_priv *ssm2602;
@@ -635,7 +635,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
return ret;
}
-static int ssm2602_i2c_remove(struct i2c_client *client)
+static int __devexit ssm2602_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
kfree(i2c_get_clientdata(client));
@@ -655,7 +655,7 @@ static struct i2c_driver ssm2602_i2c_driver = {
.owner = THIS_MODULE,
},
.probe = ssm2602_i2c_probe,
- .remove = ssm2602_i2c_remove,
+ .remove = __devexit_p(ssm2602_i2c_remove),
.id_table = ssm2602_i2c_id,
};
#endif
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 62b1f2261429..67f19c3bebe6 100644
--- a/sound/soc/codecs/tlv320aic26.h
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -14,14 +14,14 @@
#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset)
#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0)
-/* Page 0: Auxillary data registers */
+/* Page 0: Auxiliary data registers */
#define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05)
#define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06)
#define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07)
#define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09)
#define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A)
-/* Page 1: Auxillary control registers */
+/* Page 1: Auxiliary control registers */
#define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00)
#define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01)
#define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 3bedab26892f..6c43c13f0430 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
if (bypass_pll)
return 0;
- /* Use PLL, compute apropriate setup for j, d, r and p, the closest
+ /* Use PLL, compute appropriate setup for j, d, r and p, the closest
* one wins the game. Try with d==0 first, next with d!=0.
* Constraints for j are according to the datasheet.
* The sysclk is divided by 1000 to prevent integer overflows.
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index eb1a0b4e09b6..082e9d51963f 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1027,7 +1027,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
/*
* For FIFO bypass mode:
* Enable the FIFO bypass (Disable the FIFO use)
- * Set the BCLK as continous
+ * Set the BCLK as continuous
*/
fifoctrl_a |= DAC33_FBYPAS;
aictrl_b |= DAC33_BCLKON;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8512800f6326..575238d68e5e 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -281,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
i, val, twl4030_reg[i]);
}
}
- dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+ dev_dbg(codec->dev, "Found %d non-matching registers. %s\n",
difference, difference ? "Not OK" : "OK");
}
@@ -2018,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
u8 mode;
/* If the system master clock is not 26MHz, the voice PCM interface is
- * not avilable.
+ * not available.
*/
if (twl4030->sysclk != 26000) {
dev_err(codec->dev, "The board is configured for %u Hz, while"
@@ -2028,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
}
/* If the codec mode is not option2, the voice PCM interface is not
- * avilable.
+ * available.
*/
mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
& TWL4030_OPT_MODE;
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 48ffd406a71d..a7b8f301bad3 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -601,9 +601,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
.reg_cache_step = 1,
.read = uda134x_read_reg_cache,
.write = uda134x_write,
-#ifdef POWER_OFF_ON_STANDBY
.set_bias_level = uda134x_set_bias_level,
-#endif
};
static int __devinit uda134x_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 8f6b5ee6645b..4bbc0a79f01e 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
snd_soc_write(codec, WM8580_PWRDN1, reg);
- /* Make VMID high impedence */
+ /* Make VMID high impedance */
reg = snd_soc_read(codec, WM8580_ADC_CONTROL1);
reg &= ~0x100;
snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 3f09deea8d9d..ffa2ffe5ec11 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_FMTBIT_S24_LE)
/*
- * The WM8753 supports upto 4 different and mutually exclusive DAI
+ * The WM8753 supports up to 4 different and mutually exclusive DAI
* configurations. This gives 2 PCM's available for use, hifi and voice.
* NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
* is connected between the wm8753 and a BT codec or GSM modem.
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ae1cadfae84c..824d1c8c8a35 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re
case WM8903_REVISION_NUMBER:
case WM8903_INTERRUPT_STATUS_1:
case WM8903_WRITE_SEQUENCER_4:
- case WM8903_POWER_MANAGEMENT_3:
- case WM8903_POWER_MANAGEMENT_2:
case WM8903_DC_SERVO_READBACK_1:
case WM8903_DC_SERVO_READBACK_2:
case WM8903_DC_SERVO_READBACK_3:
@@ -694,7 +692,7 @@ SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
- WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv),
+ WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
SOC_ENUM("ADC Companding Mode", adc_companding),
SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
@@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
- 4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
+ 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
0, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0,
NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
@@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
- { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
- { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
- { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
- { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+ { "HPL_ENA", NULL, "Left Headphone Output PGA" },
+ { "HPR_ENA", NULL, "Right Headphone Output PGA" },
+ { "HPL_ENA_DLY", NULL, "HPL_ENA" },
+ { "HPR_ENA_DLY", NULL, "HPR_ENA" },
+ { "LINEOUTL_ENA", NULL, "Left Line Output PGA" },
+ { "LINEOUTR_ENA", NULL, "Right Line Output PGA" },
+ { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" },
+ { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" },
{ "HPL_DCS", NULL, "DCS Master" },
{ "HPR_DCS", NULL, "DCS Master" },
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 443ae580445c..9b3bba4df5b3 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 5e0214d6293e..3c7198779c31 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev,
return 0;
}
-/* Lookup table specifiying SRATE (table 25 in datasheet); some of the
+/* Lookup table specifying SRATE (table 25 in datasheet); some of the
* output frequencies have been rounded to the standard frequencies
* they are intended to match where the error is slight. */
static struct {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3b71dd65c966..500011eb8b2b 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("FLL Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 28fdfd66661d..3c2ee1bb73cd 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
reg = snd_soc_read(codec, WM8991_CLOCKING_2);
snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
- /* set up N , fractional mode and pre-divisor if neccessary */
+ /* set up N , fractional mode and pre-divisor if necessary */
snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM |
(pll_div.div2 ? WM8991_PRESCALE : 0));
snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 379fa22c5b6c..056aef904347 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3dc64c8b6a5c..84e1bd1d2822 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -82,18 +82,18 @@ struct wm8994_priv {
int mbc_ena[3];
- /* Platform dependant DRC configuration */
+ /* Platform dependent DRC configuration */
const char **drc_texts;
int drc_cfg[WM8994_NUM_DRC];
struct soc_enum drc_enum;
- /* Platform dependant ReTune mobile configuration */
+ /* Platform dependent ReTune mobile configuration */
int num_retune_mobile_texts;
const char **retune_mobile_texts;
int retune_mobile_cfg[WM8994_NUM_EQ];
struct soc_enum retune_mobile_enum;
- /* Platform dependant MBC configuration */
+ /* Platform dependent MBC configuration */
int mbc_cfg;
const char **mbc_texts;
struct soc_enum mbc_enum;
@@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Latch volume updates (right only; we always do left then right). */
+ snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME,
+ WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME,
WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
+ snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME,
+ WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME,
WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
+ snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME,
+ WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME,
WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
+ snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME,
+ WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME,
WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
+ snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME,
+ WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME,
WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
+ snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME,
+ WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME,
WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
+ snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME,
+ WM8994_DAC1_VU, WM8994_DAC1_VU);
snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME,
WM8994_DAC1_VU, WM8994_DAC1_VU);
+ snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME,
+ WM8994_DAC2_VU, WM8994_DAC2_VU);
snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME,
WM8994_DAC2_VU, WM8994_DAC2_VU);
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 55cdf2982020..91c6b39de50c 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol,
/*
* Stop any attempts to change speaker mode while the speaker is enabled.
*
- * We also have some special anti-pop controls dependant on speaker
+ * We also have some special anti-pop controls dependent on speaker
* mode which must be changed along with the mode.
*/
static int speaker_mode_put(struct snd_kcontrol *kcontrol,
@@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 7b6b3c18e299..4005e9af5d61 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
{ "SPKL", "Input Switch", "MIXINL" },
{ "SPKL", "IN1LP Switch", "IN1LP" },
- { "SPKL", "Output Switch", "Left Output Mixer" },
+ { "SPKL", "Output Switch", "Left Output PGA" },
{ "SPKL", NULL, "TOCLK" },
{ "SPKR", "Input Switch", "MIXINR" },
{ "SPKR", "IN1RP Switch", "IN1RP" },
- { "SPKR", "Output Switch", "Right Output Mixer" },
+ { "SPKR", "Output Switch", "Right Output PGA" },
{ "SPKR", NULL, "TOCLK" },
{ "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
@@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
{ "SPKOUTRP", NULL, "SPKR Driver" },
{ "SPKOUTRN", NULL, "SPKR Driver" },
- { "Left Headphone Mux", "Mixer", "Left Output Mixer" },
- { "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+ { "Left Headphone Mux", "Mixer", "Left Output PGA" },
+ { "Right Headphone Mux", "Mixer", "Right Output PGA" },
{ "Headphone PGA", NULL, "Left Headphone Mux" },
{ "Headphone PGA", NULL, "Right Headphone Mux" },