Merge tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Many updates in this development cycle are found in ASoC where it got
  a wide range of changes for the continued refactoring.

  Some highlights are below.

  ASoC:

   - Continued refactoring work by Morimoto-san toward the full
     componentization; the changes are seen allover the places

   - Support for force disconnecting muxes in DAPM

   - Continued development of ASoC Intel SOF stuff

   - New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
     CX2072X, Realtek RT1011 and RT1308

  HD-audio:

   - More fixes and adjustments for ASoC SOF HD-audio

   - Fix for resume problem on some Realtek codecs

  USB-audio:

   - A few fixes for the issues reported by syzbot USB fuzzer

   - Fix for UAC2 extension unit parser

   - Quirks for Line6 Helix, Emgaic Unitor 8

  FireWire:

   - Lots of code refactoring and fixes in most of its components"

* tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (626 commits)
  ALSA: firewire-lib: code refactoring for local variables
  ALSA: firewire-lib: code refactoring for post operation to data block counter
  ALSA: firewire-lib: code refactoring for error path of parser for CIP header
  ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet
  ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT
  ALSA: firewire-lib/fireface: fix initial value of data block counter for IR context with CIP_NO_HEADER
  ALSA: firewire-lib: fix invalid length of rx packet payload for tracepoint events
  ALSA: usb-audio: fix Line6 Helix audio format rates
  firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: bebob: fix wrong reference count for stream functionality at error path of rawmidi interface
  ASoC: SOF: Intel: implement runtime idle for CNL/APL
  ASoC: SOF: add runtime idle callback
  ASoC: hdac_hdmi: report codec link up/down status to bus
  ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
  ASoC: sunxi: sun50i-codec-analog: Add earpiece
  ASoC: rt5665: remove redundant assignment to variable idx
  ...

1 files changed, 45 insertions, 23 deletions

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 05f4514048e2..80dab5df9633 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -240,10 +240,8 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	}
 
 	/* For DSP_*, LRCLK's polarity must be inverted */
-	if (fmt & SND_SOC_DAIFMT_DSP_A) {
-		change_bit(ffs(AD193X_DAC_LEFT_HIGH) - 1,
-			   (unsigned long *)&dac_fmt);
-	}
+	if (fmt & SND_SOC_DAIFMT_DSP_A)
+		dac_fmt ^= AD193X_DAC_LEFT_HIGH;
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
@@ -415,6 +413,48 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = {
 	.ops = &ad193x_dai_ops,
 };
 
+struct ad193x_reg_default {
+	unsigned int reg;
+	unsigned int val;
+};
+
+/* codec register values to set after reset */
+static void ad193x_reg_default_init(struct ad193x_priv *ad193x)
+{
+	const struct ad193x_reg_default reg_init[] = {
+		{  0, 0x99 },	/* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */
+		{  1, 0x04 },	/* PLL_CLK_CTRL1: no on-chip Vref */
+		{  2, 0x40 },	/* DAC_CTRL0: TDM mode */
+		{  3, 0x00 },	/* DAC_CTRL1: reset */
+		{  4, 0x1A },	/* DAC_CTRL2: 48kHz de-emphasis, unmute dac */
+		{  5, 0x00 },	/* DAC_CHNL_MUTE: unmute DAC channels */
+		{  6, 0x00 },	/* DAC_L1_VOL: no attenuation */
+		{  7, 0x00 },	/* DAC_R1_VOL: no attenuation */
+		{  8, 0x00 },	/* DAC_L2_VOL: no attenuation */
+		{  9, 0x00 },	/* DAC_R2_VOL: no attenuation */
+		{ 10, 0x00 },	/* DAC_L3_VOL: no attenuation */
+		{ 11, 0x00 },	/* DAC_R3_VOL: no attenuation */
+		{ 12, 0x00 },	/* DAC_L4_VOL: no attenuation */
+		{ 13, 0x00 },	/* DAC_R4_VOL: no attenuation */
+	};
+	const struct ad193x_reg_default reg_adc_init[] = {
+		{ 14, 0x03 },	/* ADC_CTRL0: high-pass filter enable */
+		{ 15, 0x43 },	/* ADC_CTRL1: sata delay=1, adc aux mode */
+		{ 16, 0x00 },	/* ADC_CTRL2: reset */
+	};
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(reg_init); i++)
+		regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val);
+
+	if (ad193x_has_adc(ad193x)) {
+		for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) {
+			regmap_write(ad193x->regmap, reg_adc_init[i].reg,
+				     reg_adc_init[i].val);
+		}
+	}
+}
+
 static int ad193x_component_probe(struct snd_soc_component *component)
 {
 	struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component);
@@ -422,25 +462,7 @@ static int ad193x_component_probe(struct snd_soc_component *component)
 	int num, ret;
 
 	/* default setting for ad193x */
-
-	/* unmute dac channels */
-	regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0);
-	/* de-emphasis: 48kHz, powedown dac */
-	regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A);
-	/* dac in tdm mode */
-	regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40);
-
-	/* adc only */
-	if (ad193x_has_adc(ad193x)) {
-		/* high-pass filter enable */
-		regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
-		/* sata delay=1, adc aux mode */
-		regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
-	}
-
-	/* pll input: mclki/xi */
-	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
-	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
+	ad193x_reg_default_init(ad193x);
 
 	/* adc only */
 	if (ad193x_has_adc(ad193x)) {