[PATCH v2 00/15] Report jack and button detection + Capture Support

[PATCH v2 00/15] Report jack and button detection + Capture Support

[Lucas Tanure]

Hi All,

Here is a patch series for reporting to user space jack and button events and
add the support for Capture. With some cleanups and fixes along the way.

Regards,

Lucas Tanure

Lucas Tanure (12):
  ASoC: cs42l42: Fix Bitclock polarity inversion
  ASoC: cs42l42: Fix channel width support
  ASoC: cs42l42: Fix mixer volume control
  ASoC: cs42l42: Don't enable/disable regulator at Bias Level
  ASoC: cs42l42: Always wait at least 3ms after reset
  ASoC: cs42l42: Remove power if the driver is being removed
  ASoC: cs42l42: Disable regulators if probe fails
  ASoC: cs42l42: Provide finer control on playback path
  ASoC: cs42l42: Set clock source for both ways of stream
  ASoC: cs42l42: Add Capture Support
  ASoC: cs42l42: Report jack and button detection
  ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called

Richard Fitzgerald (3):
  ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT
  ASoC: cs42l42: Only start PLL if it is needed
  ASoC: cs42l42: Wait for PLL to lock before switching to it

 sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++----------------
 sound/soc/codecs/cs42l42.h |  41 +++-
 2 files changed, 282 insertions(+), 196 deletions(-)

-- 
2.30.1

[PATCH v2 01/15] ASoC: cs42l42: Fix Bitclock polarity inversion

[Lucas Tanure]

The driver was setting bit clock polarity opposite to intended polarity.
Also simplify the code by grouping ADC and DAC clock configurations into
a single field.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 20 ++++++++------------
 sound/soc/codecs/cs42l42.h | 11 ++++++-----
 2 files changed, 14 insertions(+), 17 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 210fcbedf2413..df0d5fec0287a 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -797,27 +797,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* Bitclock/frame inversion */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
+		asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_LCPOL_IN_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_IB_NF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_LCPOL_IN_SHIFT;
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
 		break;
 	}
 
-	snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG,
-				CS42L42_ASP_MODE_MASK |
-				CS42L42_ASP_SCPOL_IN_DAC_MASK |
-				CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val);
+	snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK |
+								      CS42L42_ASP_SCPOL_MASK |
+								      CS42L42_ASP_LCPOL_MASK,
+								      asp_cfg_val);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9e3cc528dcff0..1f0d67c95a9ad 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -258,11 +258,12 @@
 #define CS42L42_ASP_SLAVE_MODE		0x00
 #define CS42L42_ASP_MODE_SHIFT		4
 #define CS42L42_ASP_MODE_MASK		(1 << CS42L42_ASP_MODE_SHIFT)
-#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT	2
-#define CS42L42_ASP_SCPOL_IN_DAC_MASK	(1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT)
-#define CS42L42_ASP_LCPOL_IN_SHIFT	0
-#define CS42L42_ASP_LCPOL_IN_MASK	(1 << CS42L42_ASP_LCPOL_IN_SHIFT)
-#define CS42L42_ASP_POL_INV		1
+#define CS42L42_ASP_SCPOL_SHIFT		2
+#define CS42L42_ASP_SCPOL_MASK		(3 << CS42L42_ASP_SCPOL_SHIFT)
+#define CS42L42_ASP_SCPOL_NOR		3
+#define CS42L42_ASP_LCPOL_SHIFT		0
+#define CS42L42_ASP_LCPOL_MASK		(3 << CS42L42_ASP_LCPOL_SHIFT)
+#define CS42L42_ASP_LCPOL_INV		3
 
 #define CS42L42_ASP_FRM_CFG		(CS42L42_PAGE_12 + 0x08)
 #define CS42L42_ASP_STP_SHIFT		4
-- 
2.30.1

[PATCH v2 02/15] ASoC: cs42l42: Fix channel width support

[Lucas Tanure]

Remove the hard coded 32 bits width and replace with the correct width
calculated by params_width.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++--------------------
 sound/soc/codecs/cs42l42.h |  1 -
 2 files changed, 22 insertions(+), 26 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index df0d5fec0287a..4f9ad95479292 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -691,24 +691,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					CS42L42_CLK_OASRC_SEL_MASK,
 					CS42L42_CLK_OASRC_SEL_12 <<
 					CS42L42_CLK_OASRC_SEL_SHIFT);
-			/* channel 1 on low LRCLK, 32 bit */
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_RX_DAI0_CH1_AP_RES,
-					CS42L42_ASP_RX_CH_AP_MASK |
-					CS42L42_ASP_RX_CH_RES_MASK,
-					(CS42L42_ASP_RX_CH_AP_LOW <<
-					CS42L42_ASP_RX_CH_AP_SHIFT) |
-					(CS42L42_ASP_RX_CH_RES_32 <<
-					CS42L42_ASP_RX_CH_RES_SHIFT));
-			/* Channel 2 on high LRCLK, 32 bit */
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_RX_DAI0_CH2_AP_RES,
-					CS42L42_ASP_RX_CH_AP_MASK |
-					CS42L42_ASP_RX_CH_RES_MASK,
-					(CS42L42_ASP_RX_CH_AP_HI <<
-					CS42L42_ASP_RX_CH_AP_SHIFT) |
-					(CS42L42_ASP_RX_CH_RES_32 <<
-					CS42L42_ASP_RX_CH_RES_SHIFT));
 			if (pll_ratio_table[i].mclk_src_sel == 0) {
 				/* Pass the clock straight through */
 				snd_soc_component_update_bits(component,
@@ -824,14 +806,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
-	int retval;
+	unsigned int width = (params_width(params) / 8) - 1;
+	unsigned int val = 0;
 
 	cs42l42->srate = params_rate(params);
-	cs42l42->swidth = params_width(params);
 
-	retval = cs42l42_pll_config(component);
+	switch(substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		val |= width << CS42L42_ASP_RX_CH_RES_SHIFT;
+		/* channel 1 on low LRCLK */
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES,
+							 CS42L42_ASP_RX_CH_AP_MASK |
+							 CS42L42_ASP_RX_CH_RES_MASK, val);
+		/* Channel 2 on high LRCLK */
+		val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT;
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
+							 CS42L42_ASP_RX_CH_AP_MASK |
+							 CS42L42_ASP_RX_CH_RES_MASK, val);
+		break;
+	default:
+		break;
+	}
 
-	return retval;
+	return cs42l42_pll_config(component);
 }
 
 static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
@@ -896,9 +893,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction)
 	return 0;
 }
 
-#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
-			SNDRV_PCM_FMTBIT_S32_LE)
+#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE |\
+			 SNDRV_PCM_FMTBIT_S32_LE )
 
 
 static const struct snd_soc_dai_ops cs42l42_ops = {
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 1f0d67c95a9ad..9b017b76828a4 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -757,7 +757,6 @@ struct  cs42l42_private {
 	struct completion pdn_done;
 	u32 sclk;
 	u32 srate;
-	u32 swidth;
 	u8 plug_state;
 	u8 hs_type;
 	u8 ts_inv;
-- 
2.30.1

[PATCH v2 03/15] ASoC: cs42l42: Fix mixer volume control

[Lucas Tanure]

The minimum value is 0x3f (-63dB), which also is mute

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 4f9ad95479292..d5078ce79fadd 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = {
 };
 
 static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
-static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false);
+static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
 
 static const char * const cs42l42_hpf_freq_text[] = {
 	"1.86Hz", "120Hz", "235Hz", "466Hz"
@@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 				CS42L42_DAC_HPF_EN_SHIFT, true, false),
 	SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL,
 			 CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT,
-				0x3e, 1, mixer_tlv)
+				0x3f, 1, mixer_tlv)
 };
 
 static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w,
-- 
2.30.1

[PATCH v2 04/15] ASoC: cs42l42: Don't enable/disable regulator at Bias Level

[Lucas Tanure]

dev_pm_ops already enable/disable the codec if not in use

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 38 --------------------------------------
 1 file changed, 38 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index d5078ce79fadd..eee3fc3200308 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -511,43 +511,6 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = {
 	{"HP", NULL, "HPDRV"}
 };
 
-static int cs42l42_set_bias_level(struct snd_soc_component *component,
-					enum snd_soc_bias_level level)
-{
-	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
-	int ret;
-
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
-			regcache_cache_only(cs42l42->regmap, false);
-			regcache_sync(cs42l42->regmap);
-			ret = regulator_bulk_enable(
-						ARRAY_SIZE(cs42l42->supplies),
-						cs42l42->supplies);
-			if (ret != 0) {
-				dev_err(component->dev,
-					"Failed to enable regulators: %d\n",
-					ret);
-				return ret;
-			}
-		}
-		break;
-	case SND_SOC_BIAS_OFF:
-
-		regcache_cache_only(cs42l42->regmap, true);
-		regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies),
-						    cs42l42->supplies);
-		break;
-	}
-
-	return 0;
-}
-
 static int cs42l42_component_probe(struct snd_soc_component *component)
 {
 	struct cs42l42_private *cs42l42 =
@@ -560,7 +523,6 @@ static int cs42l42_component_probe(struct snd_soc_component *component)
 
 static const struct snd_soc_component_driver soc_component_dev_cs42l42 = {
 	.probe			= cs42l42_component_probe,
-	.set_bias_level		= cs42l42_set_bias_level,
 	.dapm_widgets		= cs42l42_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(cs42l42_dapm_widgets),
 	.dapm_routes		= cs42l42_audio_map,
-- 
2.30.1

[PATCH v2 05/15] ASoC: cs42l42: Always wait at least 3ms after reset

[Lucas Tanure]

This delay is part of the power-up sequence defined in the datasheet.
A runtime_resume is a power-up so must also include the delay.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 3 ++-
 sound/soc/codecs/cs42l42.h | 1 +
 2 files changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index eee3fc3200308..811b7b1c9732e 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1756,7 +1756,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
 		dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
 		gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
 	}
-	mdelay(3);
+	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
 	/* Request IRQ */
 	ret = devm_request_threaded_irq(&i2c_client->dev,
@@ -1881,6 +1881,7 @@ static int cs42l42_runtime_resume(struct device *dev)
 	}
 
 	gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
 	regcache_cache_only(cs42l42->regmap, false);
 	regcache_sync(cs42l42->regmap);
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9b017b76828a4..866d7c873e3c9 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -740,6 +740,7 @@
 #define CS42L42_FRAC2_VAL(val)	(((val) & 0xff0000) >> 16)
 
 #define CS42L42_NUM_SUPPLIES	5
+#define CS42L42_BOOT_TIME_US	3000
 
 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
 	"VA",
-- 
2.30.1

[PATCH v2 06/15] ASoC: cs42l42: Remove power if the driver is being removed

[Lucas Tanure]

Ensure the power supplies are turned off when removing the driver

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- Use PM functions to shutdown the codec
- Disable IRQ before pm_suspend

 sound/soc/codecs/cs42l42.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 811b7b1c9732e..78274537a22af 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -25,6 +25,7 @@
 #include <linux/of.h>
 #include <linux/of_gpio.h>
 #include <linux/of_device.h>
+#include <linux/pm_runtime.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -1840,10 +1841,9 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,

 static int cs42l42_i2c_remove(struct i2c_client *i2c_client)
 {
-	struct cs42l42_private *cs42l42 = i2c_get_clientdata(i2c_client);
-
-	/* Hold down reset */
-	gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
+	disable_irq(i2c_client->irq);
+	pm_runtime_suspend(&i2c_client->dev);
+	pm_runtime_disable(&i2c_client->dev);

 	return 0;
 }
--
2.30.1

Re: [PATCH v2 06/15] ASoC: cs42l42: Remove power if the driver is being removed

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 417 bytes --]

On Fri, Mar 05, 2021 at 05:34:33PM +0000, Lucas Tanure wrote:

> -	/* Hold down reset */
> -	gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
> +	disable_irq(i2c_client->irq);
> +	pm_runtime_suspend(&i2c_client->dev);
> +	pm_runtime_disable(&i2c_client->dev);

You should be unregistering the interrupt, not disabling it - this will
better support any shared IRQ usage that appears and is much, much more
idiomatic.

[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]

[PATCH v2 07/15] ASoC: cs42l42: Disable regulators if probe fails

[Lucas Tanure]

In case of cs42l42_i2c_probe() fail, the regulators were left enabled.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 78274537a22af..33c072f71867b 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1750,8 +1750,10 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
 	/* Reset the Device */
 	cs42l42->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
 		"reset", GPIOD_OUT_LOW);
-	if (IS_ERR(cs42l42->reset_gpio))
-		return PTR_ERR(cs42l42->reset_gpio);
+	if (IS_ERR(cs42l42->reset_gpio)) {
+		ret = PTR_ERR(cs42l42->reset_gpio);
+		goto err_disable;
+	}
 
 	if (cs42l42->reset_gpio) {
 		dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
@@ -1785,13 +1787,13 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
 		dev_err(&i2c_client->dev,
 			"CS42L42 Device ID (%X). Expected %X\n",
 			devid, CS42L42_CHIP_ID);
-		return ret;
+		goto err_disable;
 	}
 
 	ret = regmap_read(cs42l42->regmap, CS42L42_REVID, &reg);
 	if (ret < 0) {
 		dev_err(&i2c_client->dev, "Get Revision ID failed\n");
-		return ret;
+		goto err_disable;
 	}
 
 	dev_info(&i2c_client->dev,
@@ -1817,7 +1819,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
 	if (i2c_client->dev.of_node) {
 		ret = cs42l42_handle_device_data(i2c_client, cs42l42);
 		if (ret != 0)
-			return ret;
+			goto err_disable;
 	}
 
 	/* Setup headset detection */
-- 
2.30.1

[PATCH v2 08/15] ASoC: cs42l42: Provide finer control on playback path

[Lucas Tanure]

Removing cs42l42_hpdrv_evt that enables the entire chain and replace by
a set of widgets that can better define the codec

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 65 +++++++++++++-------------------------
 sound/soc/codecs/cs42l42.h |  8 ++---
 2 files changed, 26 insertions(+), 47 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 33c072f71867b..cad623e73d0fd 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -462,54 +462,33 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 				0x3f, 1, mixer_tlv)
 };
 
-static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w,
-				struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-
-	if (event & SND_SOC_DAPM_POST_PMU) {
-		/* Enable the channels */
-		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN,
-				CS42L42_ASP_RX0_CH_EN_MASK,
-				(CS42L42_ASP_RX0_CH1_EN |
-				CS42L42_ASP_RX0_CH2_EN) <<
-				CS42L42_ASP_RX0_CH_EN_SHIFT);
-
-		/* Power up */
-		snd_soc_component_update_bits(component, CS42L42_PWR_CTL1,
-			CS42L42_ASP_DAI_PDN_MASK | CS42L42_MIXER_PDN_MASK |
-				CS42L42_HP_PDN_MASK, 0);
-	} else if (event & SND_SOC_DAPM_PRE_PMD) {
-		/* Disable the channels */
-		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN,
-				CS42L42_ASP_RX0_CH_EN_MASK, 0);
-
-		/* Power down */
-		snd_soc_component_update_bits(component, CS42L42_PWR_CTL1,
-			CS42L42_ASP_DAI_PDN_MASK | CS42L42_MIXER_PDN_MASK |
-				CS42L42_HP_PDN_MASK,
-			CS42L42_ASP_DAI_PDN_MASK | CS42L42_MIXER_PDN_MASK |
-				CS42L42_HP_PDN_MASK);
-	} else {
-		dev_err(component->dev, "Invalid event 0x%x\n", event);
-	}
-	return 0;
-}
-
 static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = {
+	/* Playback Path */
 	SND_SOC_DAPM_OUTPUT("HP"),
-	SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS42L42_ASP_CLK_CFG,
-					CS42L42_ASP_SCLK_EN_SHIFT, false),
-	SND_SOC_DAPM_OUT_DRV_E("HPDRV", SND_SOC_NOPM, 0,
-					0, NULL, 0, cs42l42_hpdrv_evt,
-					SND_SOC_DAPM_POST_PMU |
-					SND_SOC_DAPM_PRE_PMD)
+	SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1),
+	SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0),
+	SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH1_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH2_SHIFT, 0),
+
+	/* Playback Requirements */
+	SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("SCLK", CS42L42_ASP_CLK_CFG, CS42L42_ASP_SCLK_EN_SHIFT, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route cs42l42_audio_map[] = {
-	{"SDIN", NULL, "Playback"},
-	{"HPDRV", NULL, "SDIN"},
-	{"HP", NULL, "HPDRV"}
+	/* Playback Path */
+	{"HP", NULL, "DAC"},
+	{"DAC", NULL, "MIXER"},
+	{"MIXER", NULL, "SDIN1"},
+	{"MIXER", NULL, "SDIN2"},
+	{"SDIN1", NULL, "Playback"},
+	{"SDIN2", NULL, "Playback"},
+
+	/* Playback Requirements */
+	{"SDIN1", NULL, "ASP DAI0"},
+	{"SDIN2", NULL, "ASP DAI0"},
+	{"SDIN1", NULL, "SCLK"},
+	{"SDIN2", NULL, "SCLK"},
 };
 
 static int cs42l42_component_probe(struct snd_soc_component *component)
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 866d7c873e3c9..4b448c102f538 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -695,10 +695,10 @@
 #define CS42L42_ASP_RX_DAI0_EN		(CS42L42_PAGE_2A + 0x01)
 #define CS42L42_ASP_RX0_CH_EN_SHIFT	2
 #define CS42L42_ASP_RX0_CH_EN_MASK	(0xf << CS42L42_ASP_RX0_CH_EN_SHIFT)
-#define CS42L42_ASP_RX0_CH1_EN		1
-#define CS42L42_ASP_RX0_CH2_EN		2
-#define CS42L42_ASP_RX0_CH3_EN		4
-#define CS42L42_ASP_RX0_CH4_EN		8
+#define CS42L42_ASP_RX0_CH1_SHIFT	2
+#define CS42L42_ASP_RX0_CH2_SHIFT	3
+#define CS42L42_ASP_RX0_CH3_SHIFT	4
+#define CS42L42_ASP_RX0_CH4_SHIFT	5
 
 #define CS42L42_ASP_RX_DAI0_CH1_AP_RES	(CS42L42_PAGE_2A + 0x02)
 #define CS42L42_ASP_RX_DAI0_CH1_BIT_MSB	(CS42L42_PAGE_2A + 0x03)
-- 
2.30.1

[PATCH v2 09/15] ASoC: cs42l42: Set clock source for both ways of stream

[Lucas Tanure]

Move the enable/disable of clocks to cs42l42_mute_stream so the record
path also get clocks.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- Add missing cs42l42_private struct pointer

 sound/soc/codecs/cs42l42.c | 86 +++++++++++++++++++++-----------------
 sound/soc/codecs/cs42l42.h |  1 +
 2 files changed, 49 insertions(+), 38 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index cad623e73d0fd..774d81a1de1d8 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -784,52 +784,63 @@ static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction)
+static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 {
 	struct snd_soc_component *component = dai->component;
+	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
 	unsigned int regval;
 	u8 fullScaleVol;
 
 	if (mute) {
-		/* Mark SCLK as not present to turn on the internal
-		 * oscillator.
-		 */
-		snd_soc_component_update_bits(component, CS42L42_OSC_SWITCH,
-						CS42L42_SCLK_PRESENT_MASK, 0);
-
-		snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
-				CS42L42_PLL_START_MASK,
-				0 << CS42L42_PLL_START_SHIFT);
-
 		/* Mute the headphone */
-		snd_soc_component_update_bits(component, CS42L42_HP_CTL,
-				CS42L42_HP_ANA_AMUTE_MASK |
-				CS42L42_HP_ANA_BMUTE_MASK,
-				CS42L42_HP_ANA_AMUTE_MASK |
-				CS42L42_HP_ANA_BMUTE_MASK);
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			snd_soc_component_update_bits(component, CS42L42_HP_CTL,
+						      CS42L42_HP_ANA_AMUTE_MASK |
+						      CS42L42_HP_ANA_BMUTE_MASK,
+						      CS42L42_HP_ANA_AMUTE_MASK |
+						      CS42L42_HP_ANA_BMUTE_MASK);
+
+		cs42l42->stream_use &= ~(1 << stream);
+		if(!cs42l42->stream_use) {
+			/*
+			 * Switch to the internal oscillator.
+			 * SCLK must remain running until after this clock switch.
+			 * Without a source of clock the I2C bus doesn't work.
+			 */
+			snd_soc_component_update_bits(component, CS42L42_OSC_SWITCH,
+						      CS42L42_SCLK_PRESENT_MASK, 0);
+			snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
+						      CS42L42_PLL_START_MASK, 0);
+		}
 	} else {
-		snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
-				CS42L42_PLL_START_MASK,
-				1 << CS42L42_PLL_START_SHIFT);
-		/* Read the headphone load */
-		regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT);
-		if (((regval & CS42L42_RLA_STAT_MASK) >>
-			CS42L42_RLA_STAT_SHIFT) == CS42L42_RLA_STAT_15_OHM) {
-			fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK;
-		} else {
-			fullScaleVol = 0;
+		if (!cs42l42->stream_use) {
+			/* SCLK must be running before codec unmute */
+			snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
+						      CS42L42_PLL_START_MASK, 1);
+
+			/* Mark SCLK as present, turn off internal oscillator */
+			snd_soc_component_update_bits(component, CS42L42_OSC_SWITCH,
+						      CS42L42_SCLK_PRESENT_MASK,
+						      CS42L42_SCLK_PRESENT_MASK);
 		}
+		cs42l42->stream_use |= 1 << stream;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			/* Read the headphone load */
+			regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT);
+			if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) ==
+			    CS42L42_RLA_STAT_15_OHM) {
+				fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK;
+			} else {
+				fullScaleVol = 0;
+			}
 
-		/* Un-mute the headphone, set the full scale volume flag */
-		snd_soc_component_update_bits(component, CS42L42_HP_CTL,
-				CS42L42_HP_ANA_AMUTE_MASK |
-				CS42L42_HP_ANA_BMUTE_MASK |
-				CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol);
-
-		/* Mark SCLK as present, turn off internal oscillator */
-		snd_soc_component_update_bits(component, CS42L42_OSC_SWITCH,
-				CS42L42_SCLK_PRESENT_MASK,
-				CS42L42_SCLK_PRESENT_MASK);
+			/* Un-mute the headphone, set the full scale volume flag */
+			snd_soc_component_update_bits(component, CS42L42_HP_CTL,
+						      CS42L42_HP_ANA_AMUTE_MASK |
+						      CS42L42_HP_ANA_BMUTE_MASK |
+						      CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol);
+		}
 	}
 
 	return 0;
@@ -844,8 +855,7 @@ static const struct snd_soc_dai_ops cs42l42_ops = {
 	.hw_params	= cs42l42_pcm_hw_params,
 	.set_fmt	= cs42l42_set_dai_fmt,
 	.set_sysclk	= cs42l42_set_sysclk,
-	.mute_stream	= cs42l42_mute,
-	.no_capture_mute = 1,
+	.mute_stream	= cs42l42_mute_stream,
 };
 
 static struct snd_soc_dai_driver cs42l42_dai = {
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 4b448c102f538..3dcbfebc53b0f 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -768,6 +768,7 @@ struct  cs42l42_private {
 	u8 bias_thresholds[CS42L42_NUM_BIASES];
 	u8 hs_bias_ramp_rate;
 	u8 hs_bias_ramp_time;
+	u8 stream_use;
 };
 
 #endif /* __CS42L42_H__ */
-- 
2.30.1

[PATCH v2 10/15] ASoC: cs42l42: Add Capture Support

[Lucas Tanure]

Add support for capture path on headseat pins

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- No changes

 sound/soc/codecs/cs42l42.c | 39 ++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/cs42l42.h | 12 ++++++++++++
 2 files changed, 51 insertions(+)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 774d81a1de1d8..54dd942f3731e 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -472,6 +472,18 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = {
 
 	/* Playback Requirements */
 	SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0),
+
+	/* Capture Path */
+	SND_SOC_DAPM_INPUT("HS"),
+	SND_SOC_DAPM_ADC("ADC", NULL, CS42L42_PWR_CTL1, CS42L42_ADC_PDN_SHIFT, 1),
+	SND_SOC_DAPM_AIF_OUT("SDOUT1", NULL, 0, CS42L42_ASP_TX_CH_EN, CS42L42_ASP_TX0_CH1_SHIFT, 0),
+	SND_SOC_DAPM_AIF_OUT("SDOUT2", NULL, 1, CS42L42_ASP_TX_CH_EN, CS42L42_ASP_TX0_CH2_SHIFT, 0),
+
+	/* Capture Requirements */
+	SND_SOC_DAPM_SUPPLY("ASP DAO0", CS42L42_PWR_CTL1, CS42L42_ASP_DAO_PDN_SHIFT, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ASP TX EN", CS42L42_ASP_TX_SZ_EN, CS42L42_ASP_TX_EN_SHIFT, 0, NULL, 0),
+
+	/* Playback/Capture Requirements */
 	SND_SOC_DAPM_SUPPLY("SCLK", CS42L42_ASP_CLK_CFG, CS42L42_ASP_SCLK_EN_SHIFT, 0, NULL, 0),
 };
 
@@ -489,6 +501,21 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = {
 	{"SDIN2", NULL, "ASP DAI0"},
 	{"SDIN1", NULL, "SCLK"},
 	{"SDIN2", NULL, "SCLK"},
+
+	/* Capture Path */
+	{"ADC", NULL, "HS"},
+	{ "SDOUT1", NULL, "ADC" },
+	{ "SDOUT2", NULL, "ADC" },
+	{ "Capture", NULL, "SDOUT1" },
+	{ "Capture", NULL, "SDOUT2" },
+
+	/* Capture Requirements */
+	{ "SDOUT1", NULL, "ASP DAO0" },
+	{ "SDOUT2", NULL, "ASP DAO0" },
+	{ "SDOUT1", NULL, "SCLK" },
+	{ "SDOUT2", NULL, "SCLK" },
+	{ "SDOUT1", NULL, "ASP TX EN" },
+	{ "SDOUT2", NULL, "ASP TX EN" },
 };
 
 static int cs42l42_component_probe(struct snd_soc_component *component)
@@ -748,12 +775,24 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
+	unsigned int channels = params_channels(params);
 	unsigned int width = (params_width(params) / 8) - 1;
 	unsigned int val = 0;
 
 	cs42l42->srate = params_rate(params);
 
 	switch(substream->stream) {
+	case SNDRV_PCM_STREAM_CAPTURE:
+		if (channels == 2) {
+			val |= CS42L42_ASP_TX_CH2_AP_MASK;
+			val |= width << CS42L42_ASP_TX_CH2_RES_SHIFT;
+		}
+		val |= width << CS42L42_ASP_TX_CH1_RES_SHIFT;
+
+		snd_soc_component_update_bits(component, CS42L42_ASP_TX_CH_AP_RES,
+				CS42L42_ASP_TX_CH1_AP_MASK | CS42L42_ASP_TX_CH2_AP_MASK |
+				CS42L42_ASP_TX_CH2_RES_MASK | CS42L42_ASP_TX_CH1_RES_MASK, val);
+		break;
 	case SNDRV_PCM_STREAM_PLAYBACK:
 		val |= width << CS42L42_ASP_RX_CH_RES_SHIFT;
 		/* channel 1 on low LRCLK */
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 3dcbfebc53b0f..c373259ed46f7 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -683,8 +683,20 @@
 
 /* Page 0x29 Serial Port TX Registers */
 #define CS42L42_ASP_TX_SZ_EN		(CS42L42_PAGE_29 + 0x01)
+#define CS42L42_ASP_TX_EN_SHIFT		0
 #define CS42L42_ASP_TX_CH_EN		(CS42L42_PAGE_29 + 0x02)
+#define CS42L42_ASP_TX0_CH2_SHIFT	1
+#define CS42L42_ASP_TX0_CH1_SHIFT	0
+
 #define CS42L42_ASP_TX_CH_AP_RES	(CS42L42_PAGE_29 + 0x03)
+#define CS42L42_ASP_TX_CH1_AP_SHIFT	7
+#define CS42L42_ASP_TX_CH1_AP_MASK	(1 << CS42L42_ASP_TX_CH1_AP_SHIFT)
+#define CS42L42_ASP_TX_CH2_AP_SHIFT	6
+#define CS42L42_ASP_TX_CH2_AP_MASK	(1 << CS42L42_ASP_TX_CH2_AP_SHIFT)
+#define CS42L42_ASP_TX_CH2_RES_SHIFT	2
+#define CS42L42_ASP_TX_CH2_RES_MASK	(3 << CS42L42_ASP_TX_CH2_RES_SHIFT)
+#define CS42L42_ASP_TX_CH1_RES_SHIFT	0
+#define CS42L42_ASP_TX_CH1_RES_MASK	(3 << CS42L42_ASP_TX_CH1_RES_SHIFT)
 #define CS42L42_ASP_TX_CH1_BIT_MSB	(CS42L42_PAGE_29 + 0x04)
 #define CS42L42_ASP_TX_CH1_BIT_LSB	(CS42L42_PAGE_29 + 0x05)
 #define CS42L42_ASP_TX_HIZ_DLY_CFG	(CS42L42_PAGE_29 + 0x06)
-- 
2.30.1

[PATCH v2 11/15] ASoC: cs42l42: Report jack and button detection

[Lucas Tanure]

Report the Jack events to the user space through ALSA.
Also moves request_threaded_irq() to component_probe so it don't get
interrupts before the initialization the struct snd_soc_jack.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- Don't move the code around
- Removed rename component
- Request IRQ moved back to main probe
- cs42l42_component_remove removed as is not needed anymore

 sound/soc/codecs/cs42l42.c | 70 ++++++++++++++++++++++++++++++--------
 sound/soc/codecs/cs42l42.h |  3 ++
 2 files changed, 59 insertions(+), 14 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 54dd942f3731e..db8c8f0121d12 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -522,10 +522,18 @@ static int cs42l42_component_probe(struct snd_soc_component *component)
 {
 	struct cs42l42_private *cs42l42 =
 		(struct cs42l42_private *)snd_soc_component_get_drvdata(component);
+	struct snd_soc_card *crd = component->card;
+	int ret = 0;
 
 	cs42l42->component = component;
 
-	return 0;
+	ret = snd_soc_card_jack_new(crd, "CS42L42 Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 |
+				    SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3,
+				    &cs42l42->jack, NULL, 0);
+	if (ret < 0)
+		dev_err(component->dev, "Cannot create CS42L42 Headset: %d\n", ret);
+
+	return ret;
 }
 
 static const struct snd_soc_component_driver soc_component_dev_cs42l42 = {
@@ -1198,7 +1206,7 @@ static void cs42l42_cancel_hs_type_detect(struct cs42l42_private *cs42l42)
 				(3 << CS42L42_HSDET_AUTO_TIME_SHIFT));
 }
 
-static void cs42l42_handle_button_press(struct cs42l42_private *cs42l42)
+static int cs42l42_handle_button_press(struct cs42l42_private *cs42l42)
 {
 	int bias_level;
 	unsigned int detect_status;
@@ -1241,17 +1249,24 @@ static void cs42l42_handle_button_press(struct cs42l42_private *cs42l42)
 
 	switch (bias_level) {
 	case 1: /* Function C button press */
+		bias_level = SND_JACK_BTN_2;
 		dev_dbg(cs42l42->component->dev, "Function C button press\n");
 		break;
 	case 2: /* Function B button press */
+		bias_level = SND_JACK_BTN_1;
 		dev_dbg(cs42l42->component->dev, "Function B button press\n");
 		break;
 	case 3: /* Function D button press */
+		bias_level = SND_JACK_BTN_3;
 		dev_dbg(cs42l42->component->dev, "Function D button press\n");
 		break;
 	case 4: /* Function A button press */
+		bias_level = SND_JACK_BTN_0;
 		dev_dbg(cs42l42->component->dev, "Function A button press\n");
 		break;
+	default:
+		bias_level = 0;
+		break;
 	}
 
 	/* Set button detect level sensitivity back to default */
@@ -1281,6 +1296,8 @@ static void cs42l42_handle_button_press(struct cs42l42_private *cs42l42)
 		(0 << CS42L42_M_HSBIAS_HIZ_SHIFT) |
 		(1 << CS42L42_M_SHORT_RLS_SHIFT) |
 		(1 << CS42L42_M_SHORT_DET_SHIFT));
+
+	return bias_level;
 }
 
 struct cs42l42_irq_params {
@@ -1325,6 +1342,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data)
 	unsigned int current_plug_status;
 	unsigned int current_button_status;
 	unsigned int i;
+	int report = 0;
+
 
 	/* Read sticky registers to clear interurpt */
 	for (i = 0; i < ARRAY_SIZE(stickies); i++) {
@@ -1351,9 +1370,20 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data)
 	if ((~masks[5]) & irq_params_table[5].mask) {
 		if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) {
 			cs42l42_process_hs_type_detect(cs42l42);
-			dev_dbg(component->dev,
-				"Auto detect done (%d)\n",
-				cs42l42->hs_type);
+			switch(cs42l42->hs_type){
+			case CS42L42_PLUG_CTIA:
+			case CS42L42_PLUG_OMTP:
+				snd_soc_jack_report(&cs42l42->jack, SND_JACK_HEADSET,
+						    SND_JACK_HEADSET);
+				break;
+			case CS42L42_PLUG_HEADPHONE:
+				snd_soc_jack_report(&cs42l42->jack, SND_JACK_HEADPHONE,
+						    SND_JACK_HEADPHONE);
+				break;
+			default:
+				break;
+			}
+			dev_dbg(component->dev, "Auto detect done (%d)\n", cs42l42->hs_type);
 		}
 	}
 
@@ -1371,8 +1401,19 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data)
 			if (cs42l42->plug_state != CS42L42_TS_UNPLUG) {
 				cs42l42->plug_state = CS42L42_TS_UNPLUG;
 				cs42l42_cancel_hs_type_detect(cs42l42);
-				dev_dbg(component->dev,
-					"Unplug event\n");
+
+				switch(cs42l42->hs_type){
+				case CS42L42_PLUG_CTIA:
+				case CS42L42_PLUG_OMTP:
+					snd_soc_jack_report(&cs42l42->jack, 0, SND_JACK_HEADSET);
+					break;
+				case CS42L42_PLUG_HEADPHONE:
+					snd_soc_jack_report(&cs42l42->jack, 0, SND_JACK_HEADPHONE);
+					break;
+				default:
+					break;
+				}
+				dev_dbg(component->dev, "Unplug event\n");
 			}
 			break;
 
@@ -1387,14 +1428,15 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data)
 		if (!(current_button_status &
 			CS42L42_M_HSBIAS_HIZ_MASK)) {
 
-			if (current_button_status &
-				CS42L42_M_DETECT_TF_MASK) {
-				dev_dbg(component->dev,
-					"Button released\n");
-			} else if (current_button_status &
-				CS42L42_M_DETECT_FT_MASK) {
-				cs42l42_handle_button_press(cs42l42);
+			if (current_button_status & CS42L42_M_DETECT_TF_MASK) {
+				dev_dbg(component->dev, "Button released\n");
+				report = 0;
+			} else if (current_button_status & CS42L42_M_DETECT_FT_MASK) {
+				report = cs42l42_handle_button_press(cs42l42);
+
 			}
+			snd_soc_jack_report(&cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+								   SND_JACK_BTN_2 | SND_JACK_BTN_3);
 		}
 	}
 
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index c373259ed46f7..e12828877a20d 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -12,6 +12,8 @@
 #ifndef __CS42L42_H__
 #define __CS42L42_H__
 
+#include <sound/jack.h>
+
 #define CS42L42_PAGE_REGISTER	0x00	/* Page Select Register */
 #define CS42L42_WIN_START	0x00
 #define CS42L42_WIN_LEN		0x100
@@ -768,6 +770,7 @@ struct  cs42l42_private {
 	struct regulator_bulk_data supplies[CS42L42_NUM_SUPPLIES];
 	struct gpio_desc *reset_gpio;
 	struct completion pdn_done;
+	struct snd_soc_jack jack;
 	u32 sclk;
 	u32 srate;
 	u8 plug_state;
-- 
2.30.1

[PATCH v2 12/15] ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called

[Lucas Tanure]

Add support for reading the source clock from snd_soc_params_to_bclk
so the machine driver is not required to call cs42l42_set_sysclk

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
 sound/soc/codecs/cs42l42.c | 17 +++++++++++++----
 sound/soc/codecs/cs42l42.h |  1 +
 2 files changed, 14 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index db8c8f0121d12..f0270de2ccb8a 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -588,10 +588,16 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 {
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
 	int i;
+	u32 clk;
 	u32 fsync;
 
+	if (!cs42l42->sclk)
+		clk = cs42l42->bclk;
+	else
+		clk = cs42l42->sclk;
+
 	for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
-		if (pll_ratio_table[i].sclk == cs42l42->sclk) {
+		if (pll_ratio_table[i].sclk == clk) {
 			/* Configure the internal sample rate */
 			snd_soc_component_update_bits(component, CS42L42_MCLK_CTL,
 					CS42L42_INTERNAL_FS_MASK,
@@ -611,12 +617,12 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					(pll_ratio_table[i].mclk_div <<
 					CS42L42_MCLKDIV_SHIFT));
 			/* Set up the LRCLK */
-			fsync = cs42l42->sclk / cs42l42->srate;
-			if (((fsync * cs42l42->srate) != cs42l42->sclk)
+			fsync = clk / cs42l42->srate;
+			if (((fsync * cs42l42->srate) != clk)
 				|| ((fsync % 2) != 0)) {
 				dev_err(component->dev,
 					"Unsupported sclk %d/sample rate %d\n",
-					cs42l42->sclk,
+					clk,
 					cs42l42->srate);
 				return -EINVAL;
 			}
@@ -788,6 +794,7 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 	unsigned int val = 0;
 
 	cs42l42->srate = params_rate(params);
+	cs42l42->bclk = snd_soc_params_to_bclk(params);
 
 	switch(substream->stream) {
 	case SNDRV_PCM_STREAM_CAPTURE:
@@ -921,6 +928,8 @@ static struct snd_soc_dai_driver cs42l42_dai = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = CS42L42_FORMATS,
 		},
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 		.ops = &cs42l42_ops,
 };
 
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index e12828877a20d..429c6833fc811 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -771,6 +771,7 @@ struct  cs42l42_private {
 	struct gpio_desc *reset_gpio;
 	struct completion pdn_done;
 	struct snd_soc_jack jack;
+	int bclk;
 	u32 sclk;
 	u32 srate;
 	u8 plug_state;
-- 
2.30.1

[PATCH v2 13/15] ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT

[Lucas Tanure]

From: Richard Fitzgerald <rf@opensource.cirrus.com>

There must be a delay of at least 150us after writing SCLK_PRESENT
before issuing another I2C write.

This is done using struct reg_sequence because it can specify a delay
after the write and the whole sequence is written atomically.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- struct cs42l42_private add by this patch was wrong and moved to the
correct patch
- Lucas signed-off added

 sound/soc/codecs/cs42l42.c | 27 ++++++++++++++++++++++-----
 sound/soc/codecs/cs42l42.h |  1 +
 2 files changed, 23 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index f0270de2ccb8a..4181765d99af7 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -549,6 +549,24 @@ static const struct snd_soc_component_driver soc_component_dev_cs42l42 = {
 	.non_legacy_dai_naming	= 1,
 };
 
+/* Switch to SCLK. Atomic delay after the write to allow the switch to complete. */
+static const struct reg_sequence cs42l42_to_sclk_seq[] = {
+	{
+		.reg = CS42L42_OSC_SWITCH,
+		.def = CS42L42_SCLK_PRESENT_MASK,
+		.delay_us = CS42L42_CLOCK_SWITCH_DELAY_US,
+	},
+};
+
+/* Switch to OSC. Atomic delay after the write to allow the switch to complete. */
+static const struct reg_sequence cs42l42_to_osc_seq[] = {
+	{
+		.reg = CS42L42_OSC_SWITCH,
+		.def = 0,
+		.delay_us = CS42L42_CLOCK_SWITCH_DELAY_US,
+	},
+};
+
 struct cs42l42_pll_params {
 	u32 sclk;
 	u8 mclk_div;
@@ -861,8 +879,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 			 * SCLK must remain running until after this clock switch.
 			 * Without a source of clock the I2C bus doesn't work.
 			 */
-			snd_soc_component_update_bits(component, CS42L42_OSC_SWITCH,
-						      CS42L42_SCLK_PRESENT_MASK, 0);
+			regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq,
+					       ARRAY_SIZE(cs42l42_to_osc_seq));
 			snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
 						      CS42L42_PLL_START_MASK, 0);
 		}
@@ -873,9 +891,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 						      CS42L42_PLL_START_MASK, 1);
 
 			/* Mark SCLK as present, turn off internal oscillator */
-			snd_soc_component_update_bits(component, CS42L42_OSC_SWITCH,
-						      CS42L42_SCLK_PRESENT_MASK,
-						      CS42L42_SCLK_PRESENT_MASK);
+			regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_sclk_seq,
+					       ARRAY_SIZE(cs42l42_to_sclk_seq));
 		}
 		cs42l42->stream_use |= 1 << stream;
 
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 429c6833fc811..214cee762709d 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -755,6 +755,7 @@
 
 #define CS42L42_NUM_SUPPLIES	5
 #define CS42L42_BOOT_TIME_US	3000
+#define CS42L42_CLOCK_SWITCH_DELAY_US 150
 
 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
 	"VA",
-- 
2.30.1

[PATCH v2 14/15] ASoC: cs42l42: Only start PLL if it is needed

[Lucas Tanure]

From: Richard Fitzgerald <rf@opensource.cirrus.com>

The PLL is only needed for sclk < 11289600 Hz and cs42l42_pll_config()
will not configure it for higher rates. So it must only be enabled
when it is needed.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- Lucas signed-off added

 sound/soc/codecs/cs42l42.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 4181765d99af7..066068a7ffa0c 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -887,8 +887,9 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 	} else {
 		if (!cs42l42->stream_use) {
 			/* SCLK must be running before codec unmute */
-			snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
-						      CS42L42_PLL_START_MASK, 1);
+			if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600))
+				snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
+							      CS42L42_PLL_START_MASK, 1);
 
 			/* Mark SCLK as present, turn off internal oscillator */
 			regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_sclk_seq,
-- 
2.30.1

[PATCH v2 15/15] ASoC: cs42l42: Wait for PLL to lock before switching to it

[Lucas Tanure]

From: Richard Fitzgerald <rf@opensource.cirrus.com>

The PLL should have locked before using it to supply MCLK.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
---
Changes in v2:
- Lucas signed-off added

 sound/soc/codecs/cs42l42.c | 12 +++++++++++-
 sound/soc/codecs/cs42l42.h |  2 ++
 2 files changed, 13 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 066068a7ffa0c..e1a7b41826360 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -862,6 +862,7 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
 	unsigned int regval;
 	u8 fullScaleVol;
+	int ret;
 
 	if (mute) {
 		/* Mute the headphone */
@@ -887,9 +888,18 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 	} else {
 		if (!cs42l42->stream_use) {
 			/* SCLK must be running before codec unmute */
-			if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600))
+			if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) {
 				snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
 							      CS42L42_PLL_START_MASK, 1);
+				ret = regmap_read_poll_timeout(cs42l42->regmap,
+							       CS42L42_PLL_LOCK_STATUS,
+							       regval,
+							       (regval & 1),
+							       CS42L42_PLL_LOCK_POLL_US,
+							       CS42L42_PLL_LOCK_TIMEOUT_US);
+				if (ret < 0)
+					dev_warn(component->dev, "PLL failed to lock: %d\n", ret);
+			}
 
 			/* Mark SCLK as present, turn off internal oscillator */
 			regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_sclk_seq,
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 214cee762709d..36b763f0d1a06 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -756,6 +756,8 @@
 #define CS42L42_NUM_SUPPLIES	5
 #define CS42L42_BOOT_TIME_US	3000
 #define CS42L42_CLOCK_SWITCH_DELAY_US 150
+#define CS42L42_PLL_LOCK_POLL_US	250
+#define CS42L42_PLL_LOCK_TIMEOUT_US	1250
 
 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
 	"VA",
-- 
2.30.1

Re: (subset) [PATCH v2 00/15] Report jack and button detection + Capture Support

[Mark Brown]

On Fri, 5 Mar 2021 17:34:27 +0000, Lucas Tanure wrote:
> Here is a patch series for reporting to user space jack and button events and
> add the support for Capture. With some cleanups and fixes along the way.
> 
> Regards,
> 
> Lucas Tanure
> 
> [...]

Applied to

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next

Thanks!

[01/15] ASoC: cs42l42: Fix Bitclock polarity inversion
        commit: 97e2b5e5dcd543cd4d85ecb1bfa2a9721a08f411
[02/15] ASoC: cs42l42: Fix channel width support
        commit: 3656667e66858fef45017c8e7c73e9918ed23915
[03/15] ASoC: cs42l42: Fix mixer volume control
        commit: a2ddc577ee4641889bf105d4d6e05be415bd4462
[04/15] ASoC: cs42l42: Don't enable/disable regulator at Bias Level
        commit: 9ad4f9ea976e05d4eba62ea58c7c7c45705b80a1
[05/15] ASoC: cs42l42: Always wait at least 3ms after reset
        commit: ddaa9bea4ffaba50f814585f294a5d98641b41ad

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark