diff options
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r-- | sound/soc/codecs/alc5623.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/jz4740.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/lm4857.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/sn95031.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/ssm2602.c | 10 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic26.h | 4 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320dac33.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/twl4030.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/uda134x.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8580.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8753.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8903.c | 40 | ||||
-rw-r--r-- | sound/soc/codecs/wm8904.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8955.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8962.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8991.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8993.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.c | 22 | ||||
-rw-r--r-- | sound/soc/codecs/wm9081.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm_hubs.c | 8 |
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" }, |