From 7fb08871c38ba9e871d20d64f3a27409baf7b754 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 22 Feb 2021 17:00:56 +0800 Subject: [PATCH 01/22] ASoC: rt1015: fix i2c communication error Remove 0x100 cache re-sync to solve i2c communication error. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 37b5795b00d1..90767490af82 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_VENDOR_ID: case RT1015_DEVICE_ID: case RT1015_PRO_ALT: + case RT1015_MAN_I2C: case RT1015_DAC3: case RT1015_VBAT_TEST_OUT1: case RT1015_VBAT_TEST_OUT2: From 2979ef760e73e2a1a34cd4da5d2c78371dfe1028 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 22 Feb 2021 17:00:57 +0800 Subject: [PATCH 02/22] ASoC: rt1015: enable BCLK detection after calibration Enable BCLK detection after calibration. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/20210222090057.29532-2-jack.yu@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 90767490af82..844e4079d176 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -514,6 +514,7 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015) msleep(300); regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0008); regmap_write(regmap, RT1015_SYS_RST1, 0x05F5); + regmap_write(regmap, RT1015_CLK_DET, 0x8000); regcache_cache_bypass(regmap, false); regcache_mark_dirty(regmap); From 2d003ec15396cc8ffa2a887605c98a967de3078d Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 15 Feb 2021 16:33:13 +0000 Subject: [PATCH 03/22] ASoC: codecs: lpass-rx-macro: Fix uninitialized variable ec_tx There is potential read of the uninitialized variable ec_tx if the call to snd_soc_component_read fails or returns an unrecognized w->name. To avoid this corner case, initialize ec_tx to -1 so that it is caught later when ec_tx is bounds checked. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: 4f692926f562 ("ASoC: codecs: lpass-rx-macro: add dapm widgets and route") Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20210215163313.84026-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index c9c21d22c2c4..8c04b3b2c907 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2895,7 +2895,7 @@ static int rx_macro_enable_echo(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); u16 val, ec_hq_reg; - int ec_tx; + int ec_tx = -1; val = snd_soc_component_read(component, CDC_RX_INP_MUX_RX_MIX_CFG4); From 9fd914d917da05641b42cab7d40bdf8ab06dac3b Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Tue, 16 Feb 2021 14:42:21 +0300 Subject: [PATCH 04/22] ASoC: fsl_ssi: Fix TDM slot setup for I2S mode When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt() function rewrites the number of slots previously set by the fsl_ssi_set_dai_tdm_slot() function to 2 by default. To fix this, let's use the saved slot count value or, if TDM is not used and the slot count is not set, proceed as before. Fixes: 4f14f5c11db1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode") Signed-off-by: Alexander Shiyan Acked-by: Nicolin Chen Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ru Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 57811743c294..ad8af3f450e2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -878,6 +878,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) { u32 strcr = 0, scr = 0, stcr, srcr, mask; + unsigned int slots; ssi->dai_fmt = fmt; @@ -909,10 +910,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) return -EINVAL; } + slots = ssi->slots ? : 2; regmap_update_bits(ssi->regs, REG_SSI_STCCR, - SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2)); + SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots)); regmap_update_bits(ssi->regs, REG_SSI_SRCCR, - SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2)); + SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots)); /* Data on rising edge of bclk, frame low, 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS; From 30be2641848b2450f0f1b62e3a8aea42e14db640 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:15 +0100 Subject: [PATCH 05/22] ASoC: rt5670: Remove 'OUT Channel Switch' control The "OUT Channel Switch" control is a left over from code copied from thr rt5640 codec driver. With the rt5640 codec driver the output volume controls have 2 pairs of mute bits: bit 7, 15: Mute Control for Spk/Headphone/Line Output Port bit 6, 14: Mute Control for Spk/Headphone/Line Volume Channel Bits 7 and 15 are normal mute bits on the rt5670/5672 which are controlled by 2 dapm widgets: SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0, &lout_l_enable_control), SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0, &lout_r_enable_control), But on the 5670/5672 bit 6 is always reserved, where as bit 14 is "LOUT Differential Mode" on the 5670 and also reserved on the 5672. So the "OUT Channel Switch" control which is controlling bits 6+14 of the "LINE Output Control" register is bogus -> remove it. This should not cause any issues for userspace. AFAICT the rt567x codecs are only used on x86/ACPI devices and the UCM profiles used there do not use the "OUT Channel Switch" control. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210215142118.308516-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 2 -- sound/soc/codecs/rt5670.h | 4 ---- 2 files changed, 6 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c29317ea5df2..2e799e21dbda 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -637,8 +637,6 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ - SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, - RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), /* DAC Digital Volume */ diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 56b13fe6bd3c..f9c4db156c80 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -212,12 +212,8 @@ /* global definition */ #define RT5670_L_MUTE (0x1 << 15) #define RT5670_L_MUTE_SFT 15 -#define RT5670_VOL_L_MUTE (0x1 << 14) -#define RT5670_VOL_L_SFT 14 #define RT5670_R_MUTE (0x1 << 7) #define RT5670_R_MUTE_SFT 7 -#define RT5670_VOL_R_MUTE (0x1 << 6) -#define RT5670_VOL_R_SFT 6 #define RT5670_L_VOL_MASK (0x3f << 8) #define RT5670_L_VOL_SFT 8 #define RT5670_R_VOL_MASK (0x3f) From 8022f09883e827855d86173756caa07b891100f0 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:16 +0100 Subject: [PATCH 06/22] ASoC: rt5670: Remove 'HP Playback Switch' control The RT5670_L_MUTE_SFT and RT5670_R_MUTE_SFT bits (bits 15 and 7) of the RT5670_HP_VOL register are set / unset by the headphones deplop code run by rt5670_hp_event() on SND_SOC_DAPM_POST_PMU / SND_SOC_DAPM_PRE_PMD. So we should not also export a control to userspace which toggles these same bits. This should not cause any issues for userspace. AFAICT the rt567x codecs are only used on x86/ACPI devices and the UCM profiles used there do not use the "HP Playback Switch" control. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210215142118.308516-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 2e799e21dbda..932e4cd1e9a6 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -631,8 +631,6 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, static const struct snd_kcontrol_new rt5670_snd_controls[] = { /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, - RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), From 674e4ff4c2326c6e3f8ddc73c61910bf32228720 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:17 +0100 Subject: [PATCH 07/22] ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer settings The SND_SOC_DAPM_MIXER declaration for "Sto1 ADC MIXL" and "Sto1 ADC MIXR" was using the mute bits from the RT5670_STO1_ADC_DIG_VOL control as mixer master mute bits. But these bits are already exposed to userspace as controls as part of the "ADC Capture Volume" / "ADC Capture Switch" control pair: SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 127, 0, adc_vol_tlv), Both the fact that the mute bits belong to the same reg as the vol-ctrl and the "Digital Mixer Path" diagram in the datasheet clearly shows that these mute bits are not part of the mixer and having 2 separate controls poking at the same bits is a bad idea. Remove the master-mute bits settings from the "Sto1 ADC MIXL" and "Sto1 ADC MIXR" DAPM widget declarations, avoiding these bits getting poked from 2 different places. This should not cause any issues for userspace. AFAICT the rt567x codecs are only used on x86/ACPI devices and the UCM profiles used there already set the "ADC Capture Switch" as needed. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210215142118.308516-4-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 932e4cd1e9a6..2f015c24c637 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1652,12 +1652,10 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2, RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL, - RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix, - ARRAY_SIZE(rt5670_sto1_adc_l_mix)), - SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL, - RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix, - ARRAY_SIZE(rt5670_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto1_adc_l_mix, ARRAY_SIZE(rt5670_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto1_adc_r_mix, ARRAY_SIZE(rt5670_sto1_adc_r_mix)), SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, rt5670_sto2_adc_l_mix, ARRAY_SIZE(rt5670_sto2_adc_l_mix)), From 982042931c255e2e7f196c24f1e5d6de780e04f9 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:18 +0100 Subject: [PATCH 08/22] ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control For reliable output-mute LED control we need a "DAC1 Playback Switch" control. The "DAC Playback volume" control is the only control in the path from the DAC1 data input to the speaker output, so the UCM profile for the speaker output will have its PlaybackMixerElem set to "DAC1". But userspace (pulseaudio) will set the "DAC1 Playback Volume" control to its softest setting (which is not fully muted) while still showing the speaker as being enabled at a low volume in the UI. If we were to set the SNDRV_CTL_ELEM_ACCESS_SPK_LED on the "DAC1 Playback Volume" control, this would mean then what pressing KEY_VOLUMEDOWN the speaker-mute LED (embedded in the volume-mute toggle key) would light while the UI is still showing the speaker as being enabled at a low volume, meaning that the UI and the LED are out of sync. Only after an _extra_ KEY_VOLUMEDOWN press would the UI show the speaker as being muted. The path from DAC1 data input to the speaker output does have a digital mixer with DAC1's data as one of its inputs direclty after the "DAC1 Playback Volume" control. This commit adds an emulated "DAC1 Playback Switch" control by: 1. Declaring the enable flag for that mixers DAC1 input as well as the "DAC1 Playback Switch" control both as SND_SOC_NOPM controls. 2. Storing the settings of both controls as driver-private data 3. Only clearing the mute flag for the DAC1 input of that mixer if the stored values indicate both controls are enabled. This is a preparation patch for adding "audio-mute" LED trigger support. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210215142118.308516-5-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 96 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt5670.h | 5 ++ 2 files changed, 97 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 2f015c24c637..4063aac2a443 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -629,6 +629,56 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); +/* + * For reliable output-mute LED control we need a "DAC1 Playback Switch" control. + * We emulate this by only clearing the RT5670_M_DAC1_L/_R AD_DA_MIXER register + * bits when both our emulated DAC1 Playback Switch control and the DAC1 MIXL/R + * DAPM-mixer DAC1 input are enabled. + */ +static void rt5670_update_ad_da_mixer_dac1_m_bits(struct rt5670_priv *rt5670) +{ + int val = RT5670_M_DAC1_L | RT5670_M_DAC1_R; + + if (rt5670->dac1_mixl_dac1_switch && rt5670->dac1_playback_switch_l) + val &= ~RT5670_M_DAC1_L; + + if (rt5670->dac1_mixr_dac1_switch && rt5670->dac1_playback_switch_r) + val &= ~RT5670_M_DAC1_R; + + regmap_update_bits(rt5670->regmap, RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L | RT5670_M_DAC1_R, val); +} + +static int rt5670_dac1_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5670->dac1_playback_switch_l; + ucontrol->value.integer.value[1] = rt5670->dac1_playback_switch_r; + + return 0; +} + +static int rt5670_dac1_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component); + + if (rt5670->dac1_playback_switch_l == ucontrol->value.integer.value[0] && + rt5670->dac1_playback_switch_r == ucontrol->value.integer.value[1]) + return 0; + + rt5670->dac1_playback_switch_l = ucontrol->value.integer.value[0]; + rt5670->dac1_playback_switch_r = ucontrol->value.integer.value[1]; + + rt5670_update_ad_da_mixer_dac1_m_bits(rt5670); + + return 1; +} + static const struct snd_kcontrol_new rt5670_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, @@ -640,6 +690,8 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_EXT("DAC1 Playback Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt5670_dac1_playback_switch_get, rt5670_dac1_playback_switch_put), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 175, 0, dac_vol_tlv), @@ -909,18 +961,44 @@ static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { RT5670_M_MONO_ADC_R2_SFT, 1, 1), }; +/* See comment above rt5670_update_ad_da_mixer_dac1_m_bits() */ +static int rt5670_put_dac1_mix_dac1_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component); + int ret; + + if (mc->shift == 0) + rt5670->dac1_mixl_dac1_switch = ucontrol->value.integer.value[0]; + else + rt5670->dac1_mixr_dac1_switch = ucontrol->value.integer.value[0]; + + /* Apply the update (if any) */ + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + if (ret == 0) + return 0; + + rt5670_update_ad_da_mixer_dac1_m_bits(rt5670); + + return 1; +} + +#define SOC_DAPM_SINGLE_RT5670_DAC1_SW(name, shift) \ + SOC_SINGLE_EXT(name, SND_SOC_NOPM, shift, 1, 0, \ + snd_soc_dapm_get_volsw, rt5670_put_dac1_mix_dac1_switch) + static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, RT5670_M_ADCMIX_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, - RT5670_M_DAC1_L_SFT, 1, 1), + SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 0), }; static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, RT5670_M_ADCMIX_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, - RT5670_M_DAC1_R_SFT, 1, 1), + SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 1), }; static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { @@ -2993,6 +3071,16 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "quirk JD mode 3\n"); } + /* + * Enable the emulated "DAC1 Playback Switch" by default to avoid + * muting the output with older UCM profiles. + */ + rt5670->dac1_playback_switch_l = true; + rt5670->dac1_playback_switch_r = true; + /* The Power-On-Reset values for the DAC1 mixer have the DAC1 input enabled. */ + rt5670->dac1_mixl_dac1_switch = true; + rt5670->dac1_mixr_dac1_switch = true; + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index f9c4db156c80..6fb3c369ee98 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -2013,6 +2013,11 @@ struct rt5670_priv { int dsp_rate; int jack_type; int jack_type_saved; + + bool dac1_mixl_dac1_switch; + bool dac1_mixr_dac1_switch; + bool dac1_playback_switch_l; + bool dac1_playback_switch_r; }; void rt5670_jack_suspend(struct snd_soc_component *component); From f84b4524005238fc9fd5cf615bb426fa40a99494 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 24 Feb 2021 14:57:51 +0800 Subject: [PATCH 09/22] ASoC: ak4458: Add MODULE_DEVICE_TABLE Add missed MODULE_DEVICE_TABLE for the driver can be loaded automatically at boot. Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver") Cc: Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/ak4458.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 472caad17012..85a1d00894a9 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -812,6 +812,7 @@ static const struct of_device_id ak4458_of_match[] = { { .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata}, { }, }; +MODULE_DEVICE_TABLE(of, ak4458_of_match); static struct i2c_driver ak4458_i2c_driver = { .driver = { From 741c8397e5d0b339fb3e614a9ff5cb4bf7ae1a65 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 24 Feb 2021 14:57:52 +0800 Subject: [PATCH 10/22] ASoC: ak5558: Add MODULE_DEVICE_TABLE Add missed MODULE_DEVICE_TABLE for the driver can be loaded automatically at boot. Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver") Cc: Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/ak5558.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 8a32b0139cb0..85bdd0534180 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -419,6 +419,7 @@ static const struct of_device_id ak5558_i2c_dt_ids[] __maybe_unused = { { .compatible = "asahi-kasei,ak5558"}, { } }; +MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids); static struct i2c_driver ak5558_i2c_driver = { .driver = { From 1045a5c04e16716870cc953872e703258e7896de Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 24 Feb 2021 11:50:52 +0100 Subject: [PATCH 11/22] ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX models since these use almost the same settings. While doing this I accidentally also copied and kept the non-standard OVCD_TH_1500UA setting used on those models. This too low threshold is causing headsets to often be seen as headphones (without a headset-mic) and when correctly identified it is causing ghost play/pause button-presses to get detected. Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA setting, fixing these problems. Fixes: fbdae7d6d04d ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks") Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210224105052.42116-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 782f2b4d72ad..5d48cc359c3d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -581,7 +581,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, From 24a7b77daed8f973bf8a5ed2f83344f44f9f6396 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 26 Feb 2021 15:38:13 +0100 Subject: [PATCH 12/22] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1414ad15d01c..a5674c227b3a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ From e4ffab875d32bf4ffa37b5cd725ace9e15d1707d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 26 Feb 2021 15:38:14 +0100 Subject: [PATCH 13/22] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index d198e191fb0c..e59fdc81dbd4 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ From d74fcdc51afd431ca9d956e032e14d12f0ee4153 Mon Sep 17 00:00:00 2001 From: Benjamin Rood Date: Fri, 19 Feb 2021 13:33:08 -0500 Subject: [PATCH 14/22] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood Reviewed-by: Fabio Estevam Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 73551e36695e..6d9bb256a2cf 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, From 4d4e677a68e770b84c87d1438d9f4e161658536a Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 28 Feb 2021 17:04:41 +0100 Subject: [PATCH 15/22] ASoC: es8316: Simplify adc_pga_gain_tlv table Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index d632055370e0..067757d1d70a 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, From 290c323008db6e3a44d981a46b56f7f166979a04 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 1 Mar 2021 18:34:10 -0600 Subject: [PATCH 16/22] ASoC: SOF: Intel: unregister DMIC device on probe error We only unregister the platform device during the .remove operation, but if the probe fails we will never reach this sequence. Suggested-by: Bard Liao Fixes: dd96daca6c83e ("ASoC: SOF: Intel: Add APL/CNL HW DSP support") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20210302003410.1178535-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 0dc3a8c0f5e3..001fc9834a19 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -895,6 +895,7 @@ free_streams: /* dsp_unmap: not currently used */ iounmap(sdev->bar[HDA_DSP_BAR]); hdac_bus_unmap: + platform_device_unregister(hdev->dmic_dev); iounmap(bus->remap_addr); hda_codec_i915_exit(sdev); err: From c014170408bcd2e8fc726802ed16794d358742ff Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Wed, 3 Mar 2021 11:55:26 +0000 Subject: [PATCH 17/22] ASoC: soc-core: Prevent warning if no DMI table is present Many systems do not use ACPI and hence do not provide a DMI table. On non-ACPI systems a warning, such as the following, is printed on boot. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The variable 'dmi_available' is not exported and so currently cannot be used by kernel modules without adding an accessor. However, it is possible to use the function is_acpi_device_node() to determine if the sound card is an ACPI device and hence indicate if we expect a DMI table to be present. Therefore, call is_acpi_device_node() to see if we are using ACPI and only parse the DMI table if we are booting with ACPI. Signed-off-by: Jon Hunter Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..0cffc9527e28 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -31,6 +31,7 @@ #include #include #include +#include #include #include #include @@ -1573,6 +1574,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (card->long_name) return 0; /* long name already set by driver or from DMI */ + if (!is_acpi_device_node(card->dev->fwnode)) + return 0; + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { From 97e2b5e5dcd543cd4d85ecb1bfa2a9721a08f411 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:28 +0000 Subject: [PATCH 18/22] 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 Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown --- 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 210fcbedf241..df0d5fec0287 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 9e3cc528dcff..1f0d67c95a9a 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 From 3656667e66858fef45017c8e7c73e9918ed23915 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:29 +0000 Subject: [PATCH 19/22] 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 Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown --- 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 df0d5fec0287..4f9ad9547929 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 1f0d67c95a9a..9b017b76828a 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; From a2ddc577ee4641889bf105d4d6e05be415bd4462 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:30 +0000 Subject: [PATCH 20/22] ASoC: cs42l42: Fix mixer volume control The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown --- 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 4f9ad9547929..d5078ce79fad 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, From 9ad4f9ea976e05d4eba62ea58c7c7c45705b80a1 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:31 +0000 Subject: [PATCH 21/22] 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 Link: https://lore.kernel.org/r/20210305173442.195740-5-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown --- 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 d5078ce79fad..eee3fc320030 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, From ddaa9bea4ffaba50f814585f294a5d98641b41ad Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:32 +0000 Subject: [PATCH 22/22] 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 Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown --- 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 eee3fc320030..811b7b1c9732 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 9b017b76828a..866d7c873e3c 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",