[05/16] ASoC: codecs: rename to snd_soc_component_read()

Message ID 87mu534me5.wl-kuninori.morimoto.gx@renesas.com
State Accepted
Commit 981abdfe99950d6eff2481fb4c19aeeac50d0ca9
Headers show
Series
  • [01/16] ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()
Related show

Commit Message

Kuninori Morimoto June 16, 2020, 5:20 a.m.
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

We need to use snd_soc_component_read()
instead of     snd_soc_component_read32()

This patch renames _read32() to _read()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
 sound/soc/codecs/88pm860x-codec.c | 14 +++++++-------
 sound/soc/codecs/ab8500-codec.c   |  8 ++++----
 sound/soc/codecs/ad1980.c         |  4 ++--
 sound/soc/codecs/arizona.c        | 18 +++++++++---------
 sound/soc/codecs/nau8822.c        |  4 ++--
 sound/soc/codecs/sgtl5000.c       | 16 ++++++++--------
 sound/soc/codecs/sta32x.c         |  4 ++--
 sound/soc/codecs/tas2552.c        |  4 ++--
 sound/soc/codecs/tscs42xx.c       |  4 ++--
 9 files changed, 38 insertions(+), 38 deletions(-)

Patch

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 00b2c43d28a1..068914d0ef3d 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -274,10 +274,10 @@  static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
 	unsigned int reg2 = mc->rreg;
 	int val[2], val2[2], i;
 
-	val[0] = snd_soc_component_read32(component, reg) & 0x3f;
-	val[1] = (snd_soc_component_read32(component, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
-	val2[0] = snd_soc_component_read32(component, reg2) & 0x3f;
-	val2[1] = (snd_soc_component_read32(component, PM860X_SIDETONE_SHIFT)) & 0xf;
+	val[0] = snd_soc_component_read(component, reg) & 0x3f;
+	val[1] = (snd_soc_component_read(component, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
+	val2[0] = snd_soc_component_read(component, reg2) & 0x3f;
+	val2[1] = (snd_soc_component_read(component, PM860X_SIDETONE_SHIFT)) & 0xf;
 
 	for (i = 0; i < ARRAY_SIZE(st_table); i++) {
 		if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
@@ -333,8 +333,8 @@  static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
 	int max = mc->max, val, val2;
 	unsigned int mask = (1 << fls(max)) - 1;
 
-	val = snd_soc_component_read32(component, reg) >> shift;
-	val2 = snd_soc_component_read32(component, reg2) >> shift;
+	val = snd_soc_component_read(component, reg) >> shift;
+	val2 = snd_soc_component_read(component, reg2) >> shift;
 	ucontrol->value.integer.value[0] = (max - val) & mask;
 	ucontrol->value.integer.value[1] = (max - val2) & mask;
 
@@ -426,7 +426,7 @@  static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
 			snd_soc_component_update_bits(component, PM860X_EAR_CTRL_2,
 					    RSYNC_CHANGE, RSYNC_CHANGE);
 			/* update dac */
-			data = snd_soc_component_read32(component, PM860X_DAC_EN_2);
+			data = snd_soc_component_read(component, PM860X_DAC_EN_2);
 			data &= ~dac;
 			if (!(data & (DAC_LEFT | DAC_RIGHT)))
 				data &= ~MODULATOR;
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 98e25d93440c..ea92007d1ef5 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1100,7 +1100,7 @@  static void anc_configure(struct snd_soc_component *component,
 	if (apply_fir)
 		for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
 			for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) {
-				val = snd_soc_component_read32(component,
+				val = snd_soc_component_read(component,
 						drvdata->anc_fir_values[par]);
 				anc_fir(component, bnk, par, val);
 			}
@@ -1108,7 +1108,7 @@  static void anc_configure(struct snd_soc_component *component,
 	if (apply_iir)
 		for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
 			for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) {
-				val = snd_soc_component_read32(component,
+				val = snd_soc_component_read(component,
 						drvdata->anc_iir_values[par]);
 				anc_iir(component, bnk, par, val);
 			}
@@ -1153,7 +1153,7 @@  static int sid_status_control_put(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&drvdata->ctrl_lock);
 
-	sidconf = snd_soc_component_read32(component, AB8500_SIDFIRCONF);
+	sidconf = snd_soc_component_read(component, AB8500_SIDFIRCONF);
 	if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) {
 		if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) {
 			dev_err(component->dev, "%s: Sidetone busy while off!\n",
@@ -1168,7 +1168,7 @@  static int sid_status_control_put(struct snd_kcontrol *kcontrol,
 	snd_soc_component_write(component, AB8500_SIDFIRADR, 0);
 
 	for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) {
-		val = snd_soc_component_read32(component, drvdata->sid_fir_values[param]);
+		val = snd_soc_component_read(component, drvdata->sid_fir_values[param]);
 		snd_soc_component_write(component, AB8500_SIDFIRCOEF1, val >> 8 & 0xff);
 		snd_soc_component_write(component, AB8500_SIDFIRCOEF2, val & 0xff);
 	}
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 43b1337bac37..9fd2023da218 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -256,7 +256,7 @@  static int ad1980_soc_probe(struct snd_soc_component *component)
 	if (ret < 0)
 		goto reset_err;
 
-	vendor_id2 = snd_soc_component_read32(component, AC97_VENDOR_ID2);
+	vendor_id2 = snd_soc_component_read(component, AC97_VENDOR_ID2);
 	if (vendor_id2 == 0x5374) {
 		dev_warn(component->dev,
 			"Found AD1981 - only 2/2 IN/OUT Channels supported\n");
@@ -270,7 +270,7 @@  static int ad1980_soc_probe(struct snd_soc_component *component)
 	snd_soc_component_write(component, AC97_SURROUND_MASTER, 0x0000);
 
 	/*power on LFE/CENTER/Surround DACs*/
-	ext_status = snd_soc_component_read32(component, AC97_EXTENDED_STATUS);
+	ext_status = snd_soc_component_read(component, AC97_EXTENDED_STATUS);
 	snd_soc_component_write(component, AC97_EXTENDED_STATUS, ext_status&~0x3800);
 
 	return 0;
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 9716c9624a89..1228f2de0297 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -87,7 +87,7 @@  static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		val = snd_soc_component_read32(component,
+		val = snd_soc_component_read(component,
 					       ARIZONA_INTERRUPT_RAW_STATUS_3);
 		if (val & ARIZONA_SPK_OVERHEAT_STS) {
 			dev_crit(arizona->dev,
@@ -897,7 +897,7 @@  static void arizona_in_set_vu(struct snd_soc_component *component, int ena)
 bool arizona_input_analog(struct snd_soc_component *component, int shift)
 {
 	unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
-	unsigned int val = snd_soc_component_read32(component, reg);
+	unsigned int val = snd_soc_component_read(component, reg);
 
 	return !(val & ARIZONA_IN1_MODE_MASK);
 }
@@ -937,7 +937,7 @@  int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		break;
 	case SND_SOC_DAPM_POST_PMD:
 		/* Disable volume updates if no inputs are enabled */
-		reg = snd_soc_component_read32(component, ARIZONA_INPUT_ENABLES);
+		reg = snd_soc_component_read(component, ARIZONA_INPUT_ENABLES);
 		if (reg == 0)
 			arizona_in_set_vu(component, 0);
 		break;
@@ -1755,15 +1755,15 @@  static bool arizona_aif_cfg_changed(struct snd_soc_component *component,
 {
 	int val;
 
-	val = snd_soc_component_read32(component, base + ARIZONA_AIF_BCLK_CTRL);
+	val = snd_soc_component_read(component, base + ARIZONA_AIF_BCLK_CTRL);
 	if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
 		return true;
 
-	val = snd_soc_component_read32(component, base + ARIZONA_AIF_TX_BCLK_RATE);
+	val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE);
 	if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
 		return true;
 
-	val = snd_soc_component_read32(component, base + ARIZONA_AIF_FRAME_CTRL_1);
+	val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1);
 	if (frame != (val & (ARIZONA_AIF1TX_WL_MASK |
 			     ARIZONA_AIF1TX_SLOT_LEN_MASK)))
 		return true;
@@ -1813,7 +1813,7 @@  static int arizona_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Force multiple of 2 channels for I2S mode */
-	val = snd_soc_component_read32(component, base + ARIZONA_AIF_FORMAT);
+	val = snd_soc_component_read(component, base + ARIZONA_AIF_FORMAT);
 	val &= ARIZONA_AIF1_FMT_MASK;
 	if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) {
 		arizona_aif_dbg(dai, "Forcing stereo mode\n");
@@ -1845,9 +1845,9 @@  static int arizona_hw_params(struct snd_pcm_substream *substream,
 
 	if (reconfig) {
 		/* Save AIF TX/RX state */
-		aif_tx_state = snd_soc_component_read32(component,
+		aif_tx_state = snd_soc_component_read(component,
 					    base + ARIZONA_AIF_TX_ENABLES);
-		aif_rx_state = snd_soc_component_read32(component,
+		aif_rx_state = snd_soc_component_read(component,
 					    base + ARIZONA_AIF_RX_ENABLES);
 		/* Disable AIF TX/RX before reconfiguring it */
 		regmap_update_bits_async(arizona->regmap,
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index a4f661335c57..79928ddeb7a1 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -188,7 +188,7 @@  static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
 	val = (u16 *)ucontrol->value.bytes.data;
 	reg = NAU8822_REG_EQ1;
 	for (i = 0; i < params->max / sizeof(u16); i++) {
-		reg_val = snd_soc_component_read32(component, reg + i);
+		reg_val = snd_soc_component_read(component, reg + i);
 		/* conversion of 16-bit integers between native CPU format
 		 * and big endian format
 		 */
@@ -445,7 +445,7 @@  static int check_mclk_select_pll(struct snd_soc_dapm_widget *source,
 		snd_soc_dapm_to_component(source->dapm);
 	unsigned int value;
 
-	value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING);
+	value = snd_soc_component_read(component, NAU8822_REG_CLOCKING);
 
 	return (value & NAU8822_CLKM_MASK);
 }
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index e8a8bf7b4ffe..eb08976a7d06 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -156,14 +156,14 @@  struct sgtl5000_priv {
 
 static inline int hp_sel_input(struct snd_soc_component *component)
 {
-	return (snd_soc_component_read32(component, SGTL5000_CHIP_ANA_CTRL) &
+	return (snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL) &
 		SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT;
 }
 
 static inline u16 mute_output(struct snd_soc_component *component,
 			      u16 mute_mask)
 {
-	u16 mute_reg = snd_soc_component_read32(component,
+	u16 mute_reg = snd_soc_component_read(component,
 					      SGTL5000_CHIP_ANA_CTRL);
 
 	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
@@ -180,7 +180,7 @@  static inline void restore_output(struct snd_soc_component *component,
 
 static void vag_power_on(struct snd_soc_component *component, u32 source)
 {
-	if (snd_soc_component_read32(component, SGTL5000_CHIP_ANA_POWER) &
+	if (snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER) &
 	    SGTL5000_VAG_POWERUP)
 		return;
 
@@ -225,7 +225,7 @@  static int vag_power_consumers(struct snd_soc_component *component,
 
 static void vag_power_off(struct snd_soc_component *component, u32 source)
 {
-	u16 ana_pwr = snd_soc_component_read32(component,
+	u16 ana_pwr = snd_soc_component_read(component,
 					     SGTL5000_CHIP_ANA_POWER);
 
 	if (!(ana_pwr & SGTL5000_VAG_POWERUP))
@@ -545,7 +545,7 @@  static int dac_get_volsw(struct snd_kcontrol *kcontrol,
 	int l;
 	int r;
 
-	reg = snd_soc_component_read32(component, SGTL5000_CHIP_DAC_VOL);
+	reg = snd_soc_component_read(component, SGTL5000_CHIP_DAC_VOL);
 
 	/* get left channel volume */
 	l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT;
@@ -633,7 +633,7 @@  static int avc_get_threshold(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	int db, i;
-	u16 reg = snd_soc_component_read32(component, SGTL5000_DAP_AVC_THRESHOLD);
+	u16 reg = snd_soc_component_read(component, SGTL5000_DAP_AVC_THRESHOLD);
 
 	/* register value 0 => -96dB */
 	if (!reg) {
@@ -1325,11 +1325,11 @@  static int sgtl5000_set_power_regs(struct snd_soc_component *component)
 	}
 
 	/* reset value */
-	ana_pwr = snd_soc_component_read32(component, SGTL5000_CHIP_ANA_POWER);
+	ana_pwr = snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER);
 	ana_pwr |= SGTL5000_DAC_STEREO |
 			SGTL5000_ADC_STEREO |
 			SGTL5000_REFTOP_POWERUP;
-	lreg_ctrl = snd_soc_component_read32(component, SGTL5000_CHIP_LINREG_CTRL);
+	lreg_ctrl = snd_soc_component_read(component, SGTL5000_CHIP_LINREG_CTRL);
 
 	if (vddio < 3100 && vdda < 3100) {
 		/* enable internal oscillator used for charge pump */
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index e9ccebbc31e4..e8d2ca4b4603 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -397,9 +397,9 @@  static void sta32x_watchdog(struct work_struct *work)
 	unsigned int confa, confa_cached;
 
 	/* check if sta32x has reset itself */
-	confa_cached = snd_soc_component_read32(component, STA32X_CONFA);
+	confa_cached = snd_soc_component_read(component, STA32X_CONFA);
 	regcache_cache_bypass(sta32x->regmap, true);
-	confa = snd_soc_component_read32(component, STA32X_CONFA);
+	confa = snd_soc_component_read(component, STA32X_CONFA);
 	regcache_cache_bypass(sta32x->regmap, false);
 	if (confa != confa_cached) {
 		regcache_mark_dirty(sta32x->regmap);
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index d90e5f2b6f27..529c0fb93f9b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -169,7 +169,7 @@  static int tas2552_setup_pll(struct snd_soc_component *component,
 		pll_clkin += tas2552->tdm_delay;
 	}
 
-	pll_enable = snd_soc_component_read32(component, TAS2552_CFG_2) & TAS2552_PLL_ENABLE;
+	pll_enable = snd_soc_component_read(component, TAS2552_CFG_2) & TAS2552_PLL_ENABLE;
 	snd_soc_component_update_bits(component, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
 
 	if (pll_clkin == pll_clk)
@@ -187,7 +187,7 @@  static int tas2552_setup_pll(struct snd_soc_component *component,
 		unsigned int d, q, t;
 		u8 j;
 		u8 pll_sel = (tas2552->pll_clk_id << 3) & TAS2552_PLL_SRC_MASK;
-		u8 p = snd_soc_component_read32(component, TAS2552_PLL_CTRL_1);
+		u8 p = snd_soc_component_read(component, TAS2552_PLL_CTRL_1);
 
 		p = (p >> 7);
 
diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c
index 27b8c6ba72fa..3265d3e8cb28 100644
--- a/sound/soc/codecs/tscs42xx.c
+++ b/sound/soc/codecs/tscs42xx.c
@@ -103,7 +103,7 @@  static bool plls_locked(struct snd_soc_component *component)
 	int count = MAX_PLL_LOCK_20MS_WAITS;
 
 	do {
-		ret = snd_soc_component_read32(component, R_PLLCTL0);
+		ret = snd_soc_component_read(component, R_PLLCTL0);
 		if (ret < 0) {
 			dev_err(component->dev,
 				"Failed to read PLL lock status (%d)\n", ret);
@@ -148,7 +148,7 @@  static int write_coeff_ram(struct snd_soc_component *component, u8 *coeff_ram,
 	for (cnt = 0; cnt < coeff_cnt; cnt++, addr++) {
 
 		for (trys = 0; trys < DACCRSTAT_MAX_TRYS; trys++) {
-			ret = snd_soc_component_read32(component, R_DACCRSTAT);
+			ret = snd_soc_component_read(component, R_DACCRSTAT);
 			if (ret < 0) {
 				dev_err(component->dev,
 					"Failed to read stat (%d)\n", ret);