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-rw-r--r--sound/soc/codecs/da7210.c24
-rw-r--r--sound/soc/codecs/da7213.c24
-rw-r--r--sound/soc/codecs/da7218.c34
-rw-r--r--sound/soc/codecs/da7219-aad.c16
-rw-r--r--sound/soc/codecs/da7219.c20
-rw-r--r--sound/soc/codecs/da732x.c18
-rw-r--r--sound/soc/codecs/da9055.c14
7 files changed, 75 insertions, 75 deletions
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index e172913d04a4..0c99dcf242e4 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -330,7 +330,7 @@ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol,
if (ucontrol->value.integer.value[0]) {
/* Check if noise suppression is enabled */
- if (snd_soc_component_read32(component, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) {
+ if (snd_soc_component_read(component, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) {
dev_dbg(component->dev,
"Disable noise suppression to enable ALC\n");
return -EINVAL;
@@ -354,27 +354,27 @@ static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol,
if (ucontrol->value.integer.value[0]) {
/* Check if ALC is enabled */
- if (snd_soc_component_read32(component, DA7210_ADC) & DA7210_ADC_ALC_EN)
+ if (snd_soc_component_read(component, DA7210_ADC) & DA7210_ADC_ALC_EN)
goto err;
/* Check ZC for HP and AUX1 PGA */
- if ((snd_soc_component_read32(component, DA7210_ZERO_CROSS) &
+ if ((snd_soc_component_read(component, DA7210_ZERO_CROSS) &
(DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC |
DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC |
DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC))
goto err;
/* Check INPGA_L_VOL and INPGA_R_VOL */
- val = snd_soc_component_read32(component, DA7210_IN_GAIN);
+ val = snd_soc_component_read(component, DA7210_IN_GAIN);
if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) ||
(((val & DA7210_INPGA_R_VOL) >> 4) <
DA7210_INPGA_MIN_VOL_NS))
goto err;
/* Check AUX1_L_VOL and AUX1_R_VOL */
- if (((snd_soc_component_read32(component, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) <
+ if (((snd_soc_component_read(component, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) <
DA7210_AUX1_MIN_VOL_NS) ||
- ((snd_soc_component_read32(component, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) <
+ ((snd_soc_component_read(component, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) <
DA7210_AUX1_MIN_VOL_NS))
goto err;
}
@@ -767,7 +767,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
/* Enable DAI */
snd_soc_component_write(component, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN);
- dai_cfg1 = 0xFC & snd_soc_component_read32(component, DA7210_DAI_CFG1);
+ dai_cfg1 = 0xFC & snd_soc_component_read(component, DA7210_DAI_CFG1);
switch (params_width(params)) {
case 16:
@@ -874,11 +874,11 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
u32 dai_cfg1;
u32 dai_cfg3;
- dai_cfg1 = 0x7f & snd_soc_component_read32(component, DA7210_DAI_CFG1);
- dai_cfg3 = 0xfc & snd_soc_component_read32(component, DA7210_DAI_CFG3);
+ dai_cfg1 = 0x7f & snd_soc_component_read(component, DA7210_DAI_CFG1);
+ dai_cfg3 = 0xfc & snd_soc_component_read(component, DA7210_DAI_CFG3);
- if ((snd_soc_component_read32(component, DA7210_PLL) & DA7210_PLL_EN) &&
- (!(snd_soc_component_read32(component, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
+ if ((snd_soc_component_read(component, DA7210_PLL) & DA7210_PLL_EN) &&
+ (!(snd_soc_component_read(component, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
return -EINVAL;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -927,7 +927,7 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
static int da7210_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
- u8 mute_reg = snd_soc_component_read32(component, DA7210_DAC_HPF) & 0xFB;
+ u8 mute_reg = snd_soc_component_read(component, DA7210_DAC_HPF) & 0xFB;
if (mute)
snd_soc_component_write(component, DA7210_DAC_HPF, mute_reg | 0x4);
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 3e6ad996741b..cc4ae7b311b4 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -205,12 +205,12 @@ static int da7213_get_alc_data(struct snd_soc_component *component, u8 reg_val)
/* Select middle 8 bits for read back from data register */
snd_soc_component_write(component, DA7213_ALC_CIC_OP_LVL_CTRL,
reg_val | DA7213_ALC_DATA_MIDDLE);
- mid_data = snd_soc_component_read32(component, DA7213_ALC_CIC_OP_LVL_DATA);
+ mid_data = snd_soc_component_read(component, DA7213_ALC_CIC_OP_LVL_DATA);
/* Select top 8 bits for read back from data register */
snd_soc_component_write(component, DA7213_ALC_CIC_OP_LVL_CTRL,
reg_val | DA7213_ALC_DATA_TOP);
- top_data = snd_soc_component_read32(component, DA7213_ALC_CIC_OP_LVL_DATA);
+ top_data = snd_soc_component_read(component, DA7213_ALC_CIC_OP_LVL_DATA);
sum += ((mid_data << 8) | (top_data << 16));
}
@@ -259,7 +259,7 @@ static void da7213_alc_calib_auto(struct snd_soc_component *component)
snd_soc_component_update_bits(component, DA7213_ALC_CTRL1, DA7213_ALC_AUTO_CALIB_EN,
DA7213_ALC_AUTO_CALIB_EN);
do {
- alc_ctrl1 = snd_soc_component_read32(component, DA7213_ALC_CTRL1);
+ alc_ctrl1 = snd_soc_component_read(component, DA7213_ALC_CTRL1);
} while (alc_ctrl1 & DA7213_ALC_AUTO_CALIB_EN);
/* If auto calibration fails, fall back to digital gain only mode */
@@ -286,16 +286,16 @@ static void da7213_alc_calib(struct snd_soc_component *component)
u8 mic_1_ctrl, mic_2_ctrl;
/* Save current values from ADC control registers */
- adc_l_ctrl = snd_soc_component_read32(component, DA7213_ADC_L_CTRL);
- adc_r_ctrl = snd_soc_component_read32(component, DA7213_ADC_R_CTRL);
+ adc_l_ctrl = snd_soc_component_read(component, DA7213_ADC_L_CTRL);
+ adc_r_ctrl = snd_soc_component_read(component, DA7213_ADC_R_CTRL);
/* Save current values from MIXIN_L/R_SELECT registers */
- mixin_l_sel = snd_soc_component_read32(component, DA7213_MIXIN_L_SELECT);
- mixin_r_sel = snd_soc_component_read32(component, DA7213_MIXIN_R_SELECT);
+ mixin_l_sel = snd_soc_component_read(component, DA7213_MIXIN_L_SELECT);
+ mixin_r_sel = snd_soc_component_read(component, DA7213_MIXIN_R_SELECT);
/* Save current values from MIC control registers */
- mic_1_ctrl = snd_soc_component_read32(component, DA7213_MIC_1_CTRL);
- mic_2_ctrl = snd_soc_component_read32(component, DA7213_MIC_2_CTRL);
+ mic_1_ctrl = snd_soc_component_read(component, DA7213_MIC_1_CTRL);
+ mic_2_ctrl = snd_soc_component_read(component, DA7213_MIC_2_CTRL);
/* Enable ADC Left and Right */
snd_soc_component_update_bits(component, DA7213_ADC_L_CTRL, DA7213_ADC_EN,
@@ -751,7 +751,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w,
DA7213_PC_FREERUN_MASK, 0);
/* If SRM not enabled then nothing more to do */
- pll_ctrl = snd_soc_component_read32(component, DA7213_PLL_CTRL);
+ pll_ctrl = snd_soc_component_read(component, DA7213_PLL_CTRL);
if (!(pll_ctrl & DA7213_PLL_SRM_EN))
return 0;
@@ -764,7 +764,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w,
/* Check SRM has locked */
do {
- pll_status = snd_soc_component_read32(component, DA7213_PLL_STATUS);
+ pll_status = snd_soc_component_read(component, DA7213_PLL_STATUS);
if (pll_status & DA7219_PLL_SRM_LOCK) {
srm_lock = true;
} else {
@@ -779,7 +779,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w,
return 0;
case SND_SOC_DAPM_POST_PMD:
/* Revert 32KHz PLL lock udpates if applied previously */
- pll_ctrl = snd_soc_component_read32(component, DA7213_PLL_CTRL);
+ pll_ctrl = snd_soc_component_read(component, DA7213_PLL_CTRL);
if (pll_ctrl & DA7213_PLL_32K_MODE) {
snd_soc_component_write(component, 0xF0, 0x8B);
snd_soc_component_write(component, 0xF2, 0x01);
diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c
index a3003f299868..6d78bccb55c3 100644
--- a/sound/soc/codecs/da7218.c
+++ b/sound/soc/codecs/da7218.c
@@ -298,22 +298,22 @@ static void da7218_alc_calib(struct snd_soc_component *component)
bool calibrated = false;
/* Save current state of MIC control registers */
- mic_1_ctrl = snd_soc_component_read32(component, DA7218_MIC_1_CTRL);
- mic_2_ctrl = snd_soc_component_read32(component, DA7218_MIC_2_CTRL);
+ mic_1_ctrl = snd_soc_component_read(component, DA7218_MIC_1_CTRL);
+ mic_2_ctrl = snd_soc_component_read(component, DA7218_MIC_2_CTRL);
/* Save current state of input mixer control registers */
- mixin_1_ctrl = snd_soc_component_read32(component, DA7218_MIXIN_1_CTRL);
- mixin_2_ctrl = snd_soc_component_read32(component, DA7218_MIXIN_2_CTRL);
+ mixin_1_ctrl = snd_soc_component_read(component, DA7218_MIXIN_1_CTRL);
+ mixin_2_ctrl = snd_soc_component_read(component, DA7218_MIXIN_2_CTRL);
/* Save current state of input filter control registers */
- in_1l_filt_ctrl = snd_soc_component_read32(component, DA7218_IN_1L_FILTER_CTRL);
- in_1r_filt_ctrl = snd_soc_component_read32(component, DA7218_IN_1R_FILTER_CTRL);
- in_2l_filt_ctrl = snd_soc_component_read32(component, DA7218_IN_2L_FILTER_CTRL);
- in_2r_filt_ctrl = snd_soc_component_read32(component, DA7218_IN_2R_FILTER_CTRL);
+ in_1l_filt_ctrl = snd_soc_component_read(component, DA7218_IN_1L_FILTER_CTRL);
+ in_1r_filt_ctrl = snd_soc_component_read(component, DA7218_IN_1R_FILTER_CTRL);
+ in_2l_filt_ctrl = snd_soc_component_read(component, DA7218_IN_2L_FILTER_CTRL);
+ in_2r_filt_ctrl = snd_soc_component_read(component, DA7218_IN_2R_FILTER_CTRL);
/* Save current state of input HPF control registers */
- in_1_hpf_ctrl = snd_soc_component_read32(component, DA7218_IN_1_HPF_FILTER_CTRL);
- in_2_hpf_ctrl = snd_soc_component_read32(component, DA7218_IN_2_HPF_FILTER_CTRL);
+ in_1_hpf_ctrl = snd_soc_component_read(component, DA7218_IN_1_HPF_FILTER_CTRL);
+ in_2_hpf_ctrl = snd_soc_component_read(component, DA7218_IN_2_HPF_FILTER_CTRL);
/* Enable then Mute MIC PGAs */
snd_soc_component_update_bits(component, DA7218_MIC_1_CTRL, DA7218_MIC_1_AMP_EN_MASK,
@@ -369,7 +369,7 @@ static void da7218_alc_calib(struct snd_soc_component *component)
snd_soc_component_update_bits(component, DA7218_CALIB_CTRL, DA7218_CALIB_AUTO_EN_MASK,
DA7218_CALIB_AUTO_EN_MASK);
do {
- calib_ctrl = snd_soc_component_read32(component, DA7218_CALIB_CTRL);
+ calib_ctrl = snd_soc_component_read(component, DA7218_CALIB_CTRL);
if (calib_ctrl & DA7218_CALIB_AUTO_EN_MASK) {
++i;
usleep_range(DA7218_ALC_CALIB_DELAY_MIN,
@@ -613,7 +613,7 @@ static int da7218_biquad_coeff_put(struct snd_kcontrol *kcontrol,
}
/* Make sure at least out filter1 enabled to allow programming */
- out_filt1l = snd_soc_component_read32(component, DA7218_OUT_1L_FILTER_CTRL);
+ out_filt1l = snd_soc_component_read(component, DA7218_OUT_1L_FILTER_CTRL);
snd_soc_component_write(component, DA7218_OUT_1L_FILTER_CTRL,
out_filt1l | DA7218_OUT_1L_FILTER_EN_MASK);
@@ -1419,7 +1419,7 @@ static int da7218_dai_event(struct snd_soc_dapm_widget *w,
i = 0;
success = false;
do {
- refosc_cal = snd_soc_component_read32(component, DA7218_PLL_REFOSC_CAL);
+ refosc_cal = snd_soc_component_read(component, DA7218_PLL_REFOSC_CAL);
if (!(refosc_cal & DA7218_PLL_REFOSC_CAL_START_MASK)) {
success = true;
} else {
@@ -1438,7 +1438,7 @@ static int da7218_dai_event(struct snd_soc_dapm_widget *w,
DA7218_PC_RESYNC_AUTO_MASK);
/* If SRM not enabled, we don't need to check status */
- pll_ctrl = snd_soc_component_read32(component, DA7218_PLL_CTRL);
+ pll_ctrl = snd_soc_component_read(component, DA7218_PLL_CTRL);
if ((pll_ctrl & DA7218_PLL_MODE_MASK) != DA7218_PLL_MODE_SRM)
return 0;
@@ -1446,7 +1446,7 @@ static int da7218_dai_event(struct snd_soc_dapm_widget *w,
i = 0;
success = false;
do {
- pll_status = snd_soc_component_read32(component, DA7218_PLL_STATUS);
+ pll_status = snd_soc_component_read(component, DA7218_PLL_STATUS);
if (pll_status & DA7218_PLL_SRM_STATUS_SRM_LOCK) {
success = true;
} else {
@@ -2236,7 +2236,7 @@ static void da7218_hpldet_irq(struct snd_soc_component *component)
u8 jack_status;
int report;
- jack_status = snd_soc_component_read32(component, DA7218_EVENT_STATUS);
+ jack_status = snd_soc_component_read(component, DA7218_EVENT_STATUS);
if (jack_status & DA7218_HPLDET_JACK_STS_MASK)
report = SND_JACK_HEADPHONE;
@@ -2256,7 +2256,7 @@ static irqreturn_t da7218_irq_thread(int irq, void *data)
u8 status;
/* Read IRQ status reg */
- status = snd_soc_component_read32(component, DA7218_EVENT);
+ status = snd_soc_component_read(component, DA7218_EVENT);
if (!status)
return IRQ_NONE;
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 4f2a96e9fd45..b1dfd91609f7 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -73,7 +73,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
snd_soc_dapm_sync(dapm);
do {
- statusa = snd_soc_component_read32(component, DA7219_ACCDET_STATUS_A);
+ statusa = snd_soc_component_read(component, DA7219_ACCDET_STATUS_A);
if (statusa & DA7219_MICBIAS_UP_STS_MASK)
micbias_up = true;
else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES)
@@ -91,7 +91,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
*/
if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) {
/* Pulse higher level voltage */
- micbias_ctrl = snd_soc_component_read32(component, DA7219_MICBIAS_CTRL);
+ micbias_ctrl = snd_soc_component_read(component, DA7219_MICBIAS_CTRL);
snd_soc_component_update_bits(component, DA7219_MICBIAS_CTRL,
DA7219_MICBIAS1_LEVEL_MASK,
da7219_aad->micbias_pulse_lvl);
@@ -141,11 +141,11 @@ static void da7219_aad_hptest_work(struct work_struct *work)
* If MCLK is present, but PLL is not enabled then we enable it here to
* ensure a consistent detection procedure.
*/
- pll_srm_sts = snd_soc_component_read32(component, DA7219_PLL_SRM_STS);
+ pll_srm_sts = snd_soc_component_read(component, DA7219_PLL_SRM_STS);
if (pll_srm_sts & DA7219_PLL_SRM_STS_MCLK) {
tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ);
- pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL);
+ pll_ctrl = snd_soc_component_read(component, DA7219_PLL_CTRL);
if ((pll_ctrl & DA7219_PLL_MODE_MASK) == DA7219_PLL_MODE_BYPASS)
da7219_set_pll(component, DA7219_SYSCLK_PLL,
DA7219_PLL_FREQ_OUT_98304);
@@ -154,7 +154,7 @@ static void da7219_aad_hptest_work(struct work_struct *work)
}
/* Ensure gain ramping at fastest rate */
- gain_ramp_ctrl = snd_soc_component_read32(component, DA7219_GAIN_RAMP_CTRL);
+ gain_ramp_ctrl = snd_soc_component_read(component, DA7219_GAIN_RAMP_CTRL);
snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_X8);
/* Bypass cache so it saves current settings */
@@ -248,7 +248,7 @@ static void da7219_aad_hptest_work(struct work_struct *work)
msleep(DA7219_AAD_HPTEST_PERIOD);
/* Grab comparator reading */
- accdet_cfg8 = snd_soc_component_read32(component, DA7219_ACCDET_CONFIG_8);
+ accdet_cfg8 = snd_soc_component_read(component, DA7219_ACCDET_CONFIG_8);
if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK)
report |= SND_JACK_HEADPHONE;
else
@@ -357,7 +357,7 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
return IRQ_NONE;
/* Read status register for jack insertion & type status */
- statusa = snd_soc_component_read32(component, DA7219_ACCDET_STATUS_A);
+ statusa = snd_soc_component_read(component, DA7219_ACCDET_STATUS_A);
/* Clear events */
regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
@@ -847,7 +847,7 @@ void da7219_aad_suspend(struct snd_soc_component *component)
* suspend then this will be dealt with through the IRQ handler.
*/
if (da7219_aad->jack_inserted) {
- micbias_ctrl = snd_soc_component_read32(component, DA7219_MICBIAS_CTRL);
+ micbias_ctrl = snd_soc_component_read(component, DA7219_MICBIAS_CTRL);
if (micbias_ctrl & DA7219_MICBIAS1_EN_MASK) {
snd_soc_dapm_disable_pin(dapm, "Mic Bias");
snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index f83a6eaba12c..f2520a6c7875 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -313,13 +313,13 @@ static void da7219_alc_calib(struct snd_soc_component *component)
u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl;
/* Save current state of mic control register */
- mic_ctrl = snd_soc_component_read32(component, DA7219_MIC_1_CTRL);
+ mic_ctrl = snd_soc_component_read(component, DA7219_MIC_1_CTRL);
/* Save current state of input mixer control register */
- mixin_ctrl = snd_soc_component_read32(component, DA7219_MIXIN_L_CTRL);
+ mixin_ctrl = snd_soc_component_read(component, DA7219_MIXIN_L_CTRL);
/* Save current state of input ADC control register */
- adc_ctrl = snd_soc_component_read32(component, DA7219_ADC_L_CTRL);
+ adc_ctrl = snd_soc_component_read(component, DA7219_ADC_L_CTRL);
/* Enable then Mute MIC PGAs */
snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK,
@@ -344,7 +344,7 @@ static void da7219_alc_calib(struct snd_soc_component *component)
DA7219_ALC_AUTO_CALIB_EN_MASK,
DA7219_ALC_AUTO_CALIB_EN_MASK);
do {
- calib_ctrl = snd_soc_component_read32(component, DA7219_ALC_CTRL1);
+ calib_ctrl = snd_soc_component_read(component, DA7219_ALC_CTRL1);
} while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK);
/* If auto calibration fails, disable DC offset, hybrid ALC */
@@ -822,13 +822,13 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w,
DA7219_PC_FREERUN_MASK, 0);
/* Slave mode, if SRM not enabled no need for status checks */
- pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL);
+ pll_ctrl = snd_soc_component_read(component, DA7219_PLL_CTRL);
if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM)
return 0;
/* Check SRM has locked */
do {
- pll_status = snd_soc_component_read32(component, DA7219_PLL_SRM_STS);
+ pll_status = snd_soc_component_read(component, DA7219_PLL_SRM_STS);
if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) {
srm_lock = true;
} else {
@@ -928,7 +928,7 @@ static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w,
case SND_SOC_DAPM_PRE_PMD:
/* Ensure nominal gain ramping for DAPM sequence */
da7219->gain_ramp_ctrl =
- snd_soc_component_read32(component, DA7219_GAIN_RAMP_CTRL);
+ snd_soc_component_read(component, DA7219_GAIN_RAMP_CTRL);
snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL,
DA7219_GAIN_RAMP_RATE_NOMINAL);
break;
@@ -1930,7 +1930,7 @@ static int da7219_wclk_is_prepared(struct clk_hw *hw)
if (!da7219->master)
return -EINVAL;
- clk_reg = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE);
+ clk_reg = snd_soc_component_read(component, DA7219_DAI_CLK_MODE);
return !!(clk_reg & DA7219_DAI_CLK_EN_MASK);
}
@@ -1942,7 +1942,7 @@ static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw,
container_of(hw, struct da7219_priv,
dai_clks_hw[DA7219_DAI_WCLK_IDX]);
struct snd_soc_component *component = da7219->component;
- u8 fs = snd_soc_component_read32(component, DA7219_SR);
+ u8 fs = snd_soc_component_read(component, DA7219_SR);
switch (fs & DA7219_SR_MASK) {
case DA7219_SR_8000:
@@ -2027,7 +2027,7 @@ static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw,
container_of(hw, struct da7219_priv,
dai_clks_hw[DA7219_DAI_BCLK_IDX]);
struct snd_soc_component *component = da7219->component;
- u8 bclks_per_wclk = snd_soc_component_read32(component,
+ u8 bclks_per_wclk = snd_soc_component_read(component,
DA7219_DAI_CLK_MODE);
switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) {
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 3f60c45e1e6d..d43ee7159ae0 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -361,7 +361,7 @@ static int da732x_hpf_get(struct snd_kcontrol *kcontrol,
unsigned int reg = enum_ctrl->reg;
int val;
- val = snd_soc_component_read32(component, reg) & DA732X_HPF_MASK;
+ val = snd_soc_component_read(component, reg) & DA732X_HPF_MASK;
switch (val) {
case DA732X_HPF_VOICE_EN:
@@ -1287,9 +1287,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_component *component)
msleep(DA732X_WAIT_FOR_STABILIZATION);
/* Check DAC offset sign */
- sign[DA732X_HPL_DAC] = (snd_soc_component_read32(component, DA732X_REG_HPL_DAC_OFF_CNTL) &
+ sign[DA732X_HPL_DAC] = (snd_soc_component_read(component, DA732X_REG_HPL_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO);
- sign[DA732X_HPR_DAC] = (snd_soc_component_read32(component, DA732X_REG_HPR_DAC_OFF_CNTL) &
+ sign[DA732X_HPR_DAC] = (snd_soc_component_read(component, DA732X_REG_HPR_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO);
/* Binary search DAC offset values (both channels at once) */
@@ -1306,10 +1306,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_component *component)
msleep(DA732X_WAIT_FOR_STABILIZATION);
- if ((snd_soc_component_read32(component, DA732X_REG_HPL_DAC_OFF_CNTL) &
+ if ((snd_soc_component_read(component, DA732X_REG_HPL_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC])
offset[DA732X_HPL_DAC] &= ~step;
- if ((snd_soc_component_read32(component, DA732X_REG_HPR_DAC_OFF_CNTL) &
+ if ((snd_soc_component_read(component, DA732X_REG_HPR_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC])
offset[DA732X_HPR_DAC] &= ~step;
@@ -1350,9 +1350,9 @@ static void da732x_output_offset_adjust(struct snd_soc_component *component)
msleep(DA732X_WAIT_FOR_STABILIZATION);
/* Check output offset sign */
- sign[DA732X_HPL_AMP] = snd_soc_component_read32(component, DA732X_REG_HPL) &
+ sign[DA732X_HPL_AMP] = snd_soc_component_read(component, DA732X_REG_HPL) &
DA732X_HP_OUT_COMPO;
- sign[DA732X_HPR_AMP] = snd_soc_component_read32(component, DA732X_REG_HPR) &
+ sign[DA732X_HPR_AMP] = snd_soc_component_read(component, DA732X_REG_HPR) &
DA732X_HP_OUT_COMPO;
snd_soc_component_write(component, DA732X_REG_HPL, DA732X_HP_OUT_COMP |
@@ -1373,10 +1373,10 @@ static void da732x_output_offset_adjust(struct snd_soc_component *component)
msleep(DA732X_WAIT_FOR_STABILIZATION);
- if ((snd_soc_component_read32(component, DA732X_REG_HPL) &
+ if ((snd_soc_component_read(component, DA732X_REG_HPL) &
DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP])
offset[DA732X_HPL_AMP] &= ~step;
- if ((snd_soc_component_read32(component, DA732X_REG_HPR) &
+ if ((snd_soc_component_read(component, DA732X_REG_HPR) &
DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP])
offset[DA732X_HPR_AMP] &= ~step;
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 94800f522d3e..e93436ccb674 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -461,12 +461,12 @@ static int da9055_get_alc_data(struct snd_soc_component *component, u8 reg_val)
/* Select middle 8 bits for read back from data register */
snd_soc_component_write(component, DA9055_ALC_CIC_OP_LVL_CTRL,
reg_val | DA9055_ALC_DATA_MIDDLE);
- mid_data = snd_soc_component_read32(component, DA9055_ALC_CIC_OP_LVL_DATA);
+ mid_data = snd_soc_component_read(component, DA9055_ALC_CIC_OP_LVL_DATA);
/* Select top 8 bits for read back from data register */
snd_soc_component_write(component, DA9055_ALC_CIC_OP_LVL_CTRL,
reg_val | DA9055_ALC_DATA_TOP);
- top_data = snd_soc_component_read32(component, DA9055_ALC_CIC_OP_LVL_DATA);
+ top_data = snd_soc_component_read(component, DA9055_ALC_CIC_OP_LVL_DATA);
sum += ((mid_data << 8) | (top_data << 16));
}
@@ -488,8 +488,8 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
*/
/* Save current values from Mic control registers */
- mic_left = snd_soc_component_read32(component, DA9055_MIC_L_CTRL);
- mic_right = snd_soc_component_read32(component, DA9055_MIC_R_CTRL);
+ mic_left = snd_soc_component_read(component, DA9055_MIC_L_CTRL);
+ mic_right = snd_soc_component_read(component, DA9055_MIC_R_CTRL);
/* Mute Mic PGA Left and Right */
snd_soc_component_update_bits(component, DA9055_MIC_L_CTRL,
@@ -498,8 +498,8 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN);
/* Save current values from ADC control registers */
- adc_left = snd_soc_component_read32(component, DA9055_ADC_L_CTRL);
- adc_right = snd_soc_component_read32(component, DA9055_ADC_R_CTRL);
+ adc_left = snd_soc_component_read(component, DA9055_ADC_L_CTRL);
+ adc_right = snd_soc_component_read(component, DA9055_ADC_R_CTRL);
/* Enable ADC Left and Right */
snd_soc_component_update_bits(component, DA9055_ADC_L_CTRL,
@@ -1176,7 +1176,7 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
}
/* Don't allow change of mode if PLL is enabled */
- if ((snd_soc_component_read32(component, DA9055_PLL_CTRL) & DA9055_PLL_EN) &&
+ if ((snd_soc_component_read(component, DA9055_PLL_CTRL) & DA9055_PLL_EN) &&
(da9055->master != mode))
return -EINVAL;
generated by cgit (git 2.34.1) at 2024-07-04 12:36:45 +0000