[PATCH v3 00/15] Report jack and button detection + Capture Support

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Lucas Tanure

6 Mar 2021 6 Mar '21

7:55 p.m.

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 | 435 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 41 +++- 2 files changed, 282 insertions(+), 194 deletions(-)

-- 2.30.1

Show replies by date

Lucas Tanure

6 Mar 6 Mar

7:55 p.m.

New subject: [PATCH v3 01/15] ASoC: cs42l42: Fix Bitclock polarity inversion

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 v3: - No changes

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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 02/15] ASoC: cs42l42: Fix channel width support

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 v3: - No changes

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; }

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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 03/15] ASoC: cs42l42: Fix mixer volume control

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

Signed-off-by: Lucas Tanure tanureal@opensource.cirrus.com --- Changes in v3: - No changes

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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 04/15] ASoC: cs42l42: Don't enable/disable regulator at Bias Level

dev_pm_ops already enable/disable the codec if not in use

Signed-off-by: Lucas Tanure tanureal@opensource.cirrus.com --- Changes in v3: - No changes

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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 05/15] ASoC: cs42l42: Always wait at least 3ms after reset

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 v3: - No changes

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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 06/15] ASoC: cs42l42: Remove power if the driver is being removed

Ensure the power supplies are turned off when removing the driver

Signed-off-by: Lucas Tanure tanureal@opensource.cirrus.com --- Changes in v3: - Free IRQ before pm_suspend

Changes in v2: - Use PM functions to shutdown the codec - Disable IRQ before pm_suspend

sound/soc/codecs/cs42l42.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 811b7b1c9732e..f61404de139b0 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> @@ -1842,8 +1843,9 @@ 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); + devm_free_irq(&i2c_client->dev, i2c_client->irq, cs42l42); + pm_runtime_suspend(&i2c_client->dev); + pm_runtime_disable(&i2c_client->dev);

return 0; }

-- 2.30.1

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 07/15] ASoC: cs42l42: Disable regulators if probe fails

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

Signed-off-by: Lucas Tanure tanureal@opensource.cirrus.com --- Changes in v3: - No changes

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 f61404de139b0..81cdd09d80bb3 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 08/15] ASoC: cs42l42: Provide finer control on playback path

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 v3: - No changes

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 81cdd09d80bb3..0b5c8e4afff07 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 09/15] ASoC: cs42l42: Set clock source for both ways of stream

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 v3: - No changes

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 0b5c8e4afff07..2dca55dfa46d7 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 10/15] ASoC: cs42l42: Add Capture Support

Add support for capture path on headseat pins

Signed-off-by: Lucas Tanure tanureal@opensource.cirrus.com --- Changes in v3: - No changes

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 2dca55dfa46d7..b5084681aaab2 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 11/15] ASoC: cs42l42: Report jack and button detection

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 v3: - No changes

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 b5084681aaab2..594bf22521037 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 12/15] ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called

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 --- Changes in v3: - No changes

Changes in v2: - No changes

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 594bf22521037..68b7ed71ad542 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 13/15] ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT

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 v3: - No changes

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 68b7ed71ad542..08718fd10fb9b 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 14/15] ASoC: cs42l42: Only start PLL if it is needed

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 v3: - No changes

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 08718fd10fb9b..d7a314aa59b73 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

Lucas Tanure

7:55 p.m.

New subject: [PATCH v3 15/15] ASoC: cs42l42: Wait for PLL to lock before switching to it

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 v3: - No changes

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 d7a314aa59b73..bf982e145e945 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

Mark Brown

9 Mar 9 Mar

8:06 p.m.

New subject: (subset) [PATCH v3 00/15] Report jack and button detection + Capture Support

On Sat, 6 Mar 2021 18:55:38 +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!

[06/15] ASoC: cs42l42: Remove power if the driver is being removed commit: 0e992635233c909652b87f1a84775746e2780306 [07/15] ASoC: cs42l42: Disable regulators if probe fails commit: 384c0c11be3f4c4bd28196f97507788cc84dd5e1 [08/15] ASoC: cs42l42: Provide finer control on playback path commit: 48a679742612308b320f1ca89366ee7fde04f547 [09/15] ASoC: cs42l42: Set clock source for both ways of stream commit: f1fe73ce62864cb48e603d61a3936b475ba5bbef [10/15] ASoC: cs42l42: Add Capture Support commit: a6ea36692a4846d9470bdeeb90081e1dc5144b95 [11/15] ASoC: cs42l42: Report jack and button detection commit: f3f6f77beaee1b955ea835358e4c3ab1bbb56927 [12/15] ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called commit: 8ecb782355f30cfb207bffd5d04c4c5f2ac98ae3 [13/15] ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT commit: 43cb98d6ba84d8917f377b5720fb1451ce86de13 [14/15] ASoC: cs42l42: Only start PLL if it is needed commit: b5019672a0526a2e0de6843734d1b7687f138b02 [15/15] ASoC: cs42l42: Wait for PLL to lock before switching to it commit: 5974fb2911b92b1921ab4aa35aede7454d72052a

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

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