From 9e0bdaa9fcb8c64efc1487a7fba07722e7bc515e Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 22 Feb 2021 17:00:56 +0800 Subject: 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: -- cgit v1.2.3 From 0d2b6e398975bcc6a29f1d466229a312dde71b53 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 22 Feb 2021 17:00:57 +0800 Subject: 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); -- cgit v1.2.3 From 0c0a5883783540a56e6a5dbf5868f045dbeaa888 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 15 Feb 2021 16:33:13 +0000 Subject: 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); -- cgit v1.2.3 From 87263968516fb9507d6215d53f44052627fae8d8 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Tue, 16 Feb 2021 14:42:21 +0300 Subject: 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; -- cgit v1.2.3 From d917b5dde660b11abd757bf99a29353c36880b2c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:15 +0100 Subject: 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) -- cgit v1.2.3 From caba8d764770b6824391c5bf3c3eba6e51b69330 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:16 +0100 Subject: 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), -- cgit v1.2.3 From 02aa946ef3762aa456d87cc55606667942b3f354 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:17 +0100 Subject: 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)), -- cgit v1.2.3 From 42121c2645d229d348399ad278b6c3fd224bd6a2 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 15 Feb 2021 15:21:18 +0100 Subject: 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); -- cgit v1.2.3 From 4ec5b96775a88dd9b1c3ba1d23c43c478cab95a2 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 24 Feb 2021 14:57:51 +0800 Subject: 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 = { -- cgit v1.2.3 From 80cffd2468ddb850e678f17841fc356930b2304a Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 24 Feb 2021 14:57:52 +0800 Subject: 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 = { -- cgit v1.2.3 From ca08ddfd961d2a17208d9182e0ee5791b39bd8bf Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 24 Feb 2021 11:50:52 +0100 Subject: 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), }, -- cgit v1.2.3 From cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 26 Feb 2021 15:38:13 +0100 Subject: 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 */ -- cgit v1.2.3 From eee51df776bd6cac10a76b2779a9fdee3f622b2b Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 26 Feb 2021 15:38:14 +0100 Subject: 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 */ -- cgit v1.2.3 From f86f58e3594fb0ab1993d833d3b9a2496f3c928c Mon Sep 17 00:00:00 2001 From: Benjamin Rood Date: Fri, 19 Feb 2021 13:33:08 -0500 Subject: 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 }, -- cgit v1.2.3 From bb18c678754ce1514100fb4c0bf6113b5af36c48 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 28 Feb 2021 17:04:41 +0100 Subject: 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, -- cgit v1.2.3 From 5bb0ecddb2a7f638d65e457f3da9fa334c967b14 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 1 Mar 2021 18:34:10 -0600 Subject: 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 1d29b1fd6a94..0c096db07322 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -897,6 +897,7 @@ 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: -- cgit v1.2.3 From 7de14d581dbed57c2b3c6afffa2c3fdc6955a3cd Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Wed, 3 Mar 2021 11:55:26 +0000 Subject: 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)) { -- cgit v1.2.3 From e793c965519b8b7f2fea51a48398405e2a501729 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:28 +0000 Subject: 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 -- cgit v1.2.3 From 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:29 +0000 Subject: 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; -- cgit v1.2.3 From 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:30 +0000 Subject: 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, -- cgit v1.2.3 From 15013240fcf791691f99c884802099db34c099b9 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:31 +0000 Subject: 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, -- cgit v1.2.3 From 19325cfea04446bc79b36bffd4978af15f46a00e Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 5 Mar 2021 17:34:32 +0000 Subject: 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", -- cgit v1.2.3 From 5346f0e80b7160c91fb599d4545fd12560c286ed Mon Sep 17 00:00:00 2001 From: Jonathan Marek Date: Thu, 4 Mar 2021 16:56:46 -0500 Subject: ASoC: codecs: lpass-va-macro: mute/unmute all active decimators An interface can have multiple decimators enabled, so loop over all active decimators. Otherwise only one channel will be unmuted, and other channels will be zero. This fixes recording from dual DMIC as a single two channel stream. Also remove the now unused "active_decimator" field. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Jonathan Marek Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210304215646.17956-1-jonathan@marek.ca Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-va-macro.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 91e6890d6efc..3d6976a3d9e4 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -189,7 +189,6 @@ struct va_macro { struct device *dev; unsigned long active_ch_mask[VA_MACRO_MAX_DAIS]; unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS]; - unsigned long active_decimator[VA_MACRO_MAX_DAIS]; u16 dmic_clk_div; int dec_mode[VA_MACRO_NUM_DECIMATORS]; @@ -549,11 +548,9 @@ static int va_macro_tx_mixer_put(struct snd_kcontrol *kcontrol, if (enable) { set_bit(dec_id, &va->active_ch_mask[dai_id]); va->active_ch_cnt[dai_id]++; - va->active_decimator[dai_id] = dec_id; } else { clear_bit(dec_id, &va->active_ch_mask[dai_id]); va->active_ch_cnt[dai_id]--; - va->active_decimator[dai_id] = -1; } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, update); @@ -880,18 +877,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) struct va_macro *va = snd_soc_component_get_drvdata(component); u16 tx_vol_ctl_reg, decimator; - decimator = va->active_decimator[dai->id]; - - tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL + - VA_MACRO_TX_PATH_OFFSET * decimator; - if (mute) - snd_soc_component_update_bits(component, tx_vol_ctl_reg, - CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, - CDC_VA_TX_PATH_PGA_MUTE_EN); - else - snd_soc_component_update_bits(component, tx_vol_ctl_reg, - CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, - CDC_VA_TX_PATH_PGA_MUTE_DISABLE); + for_each_set_bit(decimator, &va->active_ch_mask[dai->id], + VA_MACRO_DEC_MAX) { + tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL + + VA_MACRO_TX_PATH_OFFSET * decimator; + if (mute) + snd_soc_component_update_bits(component, tx_vol_ctl_reg, + CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, + CDC_VA_TX_PATH_PGA_MUTE_EN); + else + snd_soc_component_update_bits(component, tx_vol_ctl_reg, + CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, + CDC_VA_TX_PATH_PGA_MUTE_DISABLE); + } return 0; } -- cgit v1.2.3 From e4b8b7c916038c1ffcba2c4ce92d5523c4cc2f46 Mon Sep 17 00:00:00 2001 From: Jonathan Marek Date: Thu, 4 Mar 2021 19:50:48 -0500 Subject: ASoC: codecs: lpass-wsa-macro: fix RX MIX input controls Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these controls are incorrectly toggling the first bit of the register, which is part of the FS_RATE field. Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX" control, which is to use SND_SOC_NOPM as the register and use an enum in the shift field instead. Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route") Signed-off-by: Jonathan Marek Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210305005049.24726-1-jonathan@marek.ca Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-wsa-macro.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 5ebcd935ba89..9ca49a165f69 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1211,14 +1211,16 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - u16 gain_reg; + u16 path_reg, gain_reg; int val; - switch (w->reg) { - case CDC_WSA_RX0_RX_PATH_MIX_CTL: + switch (w->shift) { + case WSA_MACRO_RX_MIX0: + path_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL; gain_reg = CDC_WSA_RX0_RX_VOL_MIX_CTL; break; - case CDC_WSA_RX1_RX_PATH_MIX_CTL: + case WSA_MACRO_RX_MIX1: + path_reg = CDC_WSA_RX1_RX_PATH_MIX_CTL; gain_reg = CDC_WSA_RX1_RX_VOL_MIX_CTL; break; default: @@ -1231,7 +1233,7 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, gain_reg, val); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_component_update_bits(component, w->reg, + snd_soc_component_update_bits(component, path_reg, CDC_WSA_RX_PATH_MIX_CLK_EN_MASK, CDC_WSA_RX_PATH_MIX_CLK_DISABLE); break; @@ -2068,14 +2070,14 @@ static const struct snd_soc_dapm_widget wsa_macro_dapm_widgets[] = { SND_SOC_DAPM_MUX("WSA_RX0 INP0", SND_SOC_NOPM, 0, 0, &rx0_prim_inp0_mux), SND_SOC_DAPM_MUX("WSA_RX0 INP1", SND_SOC_NOPM, 0, 0, &rx0_prim_inp1_mux), SND_SOC_DAPM_MUX("WSA_RX0 INP2", SND_SOC_NOPM, 0, 0, &rx0_prim_inp2_mux), - SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", CDC_WSA_RX0_RX_PATH_MIX_CTL, - 0, 0, &rx0_mix_mux, wsa_macro_enable_mix_path, + SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX0, + 0, &rx0_mix_mux, wsa_macro_enable_mix_path, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("WSA_RX1 INP0", SND_SOC_NOPM, 0, 0, &rx1_prim_inp0_mux), SND_SOC_DAPM_MUX("WSA_RX1 INP1", SND_SOC_NOPM, 0, 0, &rx1_prim_inp1_mux), SND_SOC_DAPM_MUX("WSA_RX1 INP2", SND_SOC_NOPM, 0, 0, &rx1_prim_inp2_mux), - SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", CDC_WSA_RX1_RX_PATH_MIX_CTL, - 0, 0, &rx1_mix_mux, wsa_macro_enable_mix_path, + SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX1, + 0, &rx1_mix_mux, wsa_macro_enable_mix_path, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER_E("WSA_RX INT0 MIX", SND_SOC_NOPM, 0, 0, NULL, 0, -- cgit v1.2.3 From ac101985cad3912e484295bd0ec22d117fee9f17 Mon Sep 17 00:00:00 2001 From: Peter Robinson Date: Sun, 7 Mar 2021 16:23:37 +0000 Subject: ASoC: remove remnants of sirf prima/atlas audio codec In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans up a stray header and some Kconfig entries for the codec that were missed in the process. Fixes: 61fbeb5dcb3d (ASoC: remove sirf prima/atlas drivers) Signed-off-by: Peter Robinson Cc: Arnd Bergmann Cc: Mark Brown Acked-by: Arnd Bergmann Link: https://lore.kernel.org/r/20210307162338.1160604-1-pbrobinson@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 -- sound/soc/codecs/sirf-audio-codec.h | 124 ------------------------------------ 2 files changed, 129 deletions(-) delete mode 100644 sound/soc/codecs/sirf-audio-codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..1c87b42606c9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -186,7 +186,6 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX - imply SND_SOC_SIRF_AUDIO_CODEC imply SND_SOC_SPDIF imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 @@ -1279,10 +1278,6 @@ config SND_SOC_SIMPLE_MUX tristate "Simple Audio Mux" select GPIOLIB -config SND_SOC_SIRF_AUDIO_CODEC - tristate "SiRF SoC internal audio codec" - select REGMAP_MMIO - config SND_SOC_SPDIF tristate "S/PDIF CODEC" diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h deleted file mode 100644 index a7fe2680f4c7..000000000000 --- a/sound/soc/codecs/sirf-audio-codec.h +++ /dev/null @@ -1,124 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ -/* - * SiRF inner codec controllers define - * - * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. - */ - -#ifndef _SIRF_AUDIO_CODEC_H -#define _SIRF_AUDIO_CODEC_H - - -#define AUDIO_IC_CODEC_PWR (0x00E0) -#define AUDIO_IC_CODEC_CTRL0 (0x00E4) -#define AUDIO_IC_CODEC_CTRL1 (0x00E8) -#define AUDIO_IC_CODEC_CTRL2 (0x00EC) -#define AUDIO_IC_CODEC_CTRL3 (0x00F0) - -#define MICBIASEN (1 << 3) - -#define IC_RDACEN (1 << 0) -#define IC_LDACEN (1 << 1) -#define IC_HSREN (1 << 2) -#define IC_HSLEN (1 << 3) -#define IC_SPEN (1 << 4) -#define IC_CPEN (1 << 5) - -#define IC_HPRSELR (1 << 6) -#define IC_HPLSELR (1 << 7) -#define IC_HPRSELL (1 << 8) -#define IC_HPLSELL (1 << 9) -#define IC_SPSELR (1 << 10) -#define IC_SPSELL (1 << 11) - -#define IC_MONOR (1 << 12) -#define IC_MONOL (1 << 13) - -#define IC_RXOSRSEL (1 << 28) -#define IC_CPFREQ (1 << 29) -#define IC_HSINVEN (1 << 30) - -#define IC_MICINREN (1 << 0) -#define IC_MICINLEN (1 << 1) -#define IC_MICIN1SEL (1 << 2) -#define IC_MICIN2SEL (1 << 3) -#define IC_MICDIFSEL (1 << 4) -#define IC_LINEIN1SEL (1 << 5) -#define IC_LINEIN2SEL (1 << 6) -#define IC_RADCEN (1 << 7) -#define IC_LADCEN (1 << 8) -#define IC_ALM (1 << 9) - -#define IC_DIGMICEN (1 << 22) -#define IC_DIGMICFREQ (1 << 23) -#define IC_ADC14B_12 (1 << 24) -#define IC_FIRDAC_HSL_EN (1 << 25) -#define IC_FIRDAC_HSR_EN (1 << 26) -#define IC_FIRDAC_LOUT_EN (1 << 27) -#define IC_POR (1 << 28) -#define IC_CODEC_CLK_EN (1 << 29) -#define IC_HP_3DB_BOOST (1 << 30) - -#define IC_ADC_LEFT_GAIN_SHIFT 16 -#define IC_ADC_RIGHT_GAIN_SHIFT 10 -#define IC_ADC_GAIN_MASK 0x3F -#define IC_MIC_MAX_GAIN 0x39 - -#define IC_RXPGAR_MASK 0x3F -#define IC_RXPGAR_SHIFT 14 -#define IC_RXPGAL_MASK 0x3F -#define IC_RXPGAL_SHIFT 21 -#define IC_RXPGAR 0x7B -#define IC_RXPGAL 0x7B - -#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F -#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 - -#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_SC_OFFSET) -#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_LC_OFFSET) -#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_HC_OFFSET) - -#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F -#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 - -#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_SC_OFFSET) -#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_LC_OFFSET) -#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_HC_OFFSET) -#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) -#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) - -#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) -#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) -#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) -#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) -#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) - -#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) -#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) -#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) -#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) -#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) - -#define AUDIO_FIFO_START (1 << 0) -#define AUDIO_FIFO_RESET (1 << 1) - -#define AUDIO_FIFO_FULL (1 << 0) -#define AUDIO_FIFO_EMPTY (1 << 1) -#define AUDIO_FIFO_OFLOW (1 << 2) -#define AUDIO_FIFO_UFLOW (1 << 3) - -#define IC_TX_ENABLE (0x03) -#define IC_RX_ENABLE_MONO (0x01) -#define IC_RX_ENABLE_STEREO (0x03) - -#endif /*__SIRF_AUDIO_CODEC_H*/ -- cgit v1.2.3 From 1c668e1c0a0f74472469cd514f40c9012b324c31 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 9 Mar 2021 14:21:27 +0000 Subject: ASoC: qcom: sdm845: Fix array out of bounds access Static analysis Coverity had detected a potential array out-of-bounds write issue due to the fact that MAX AFE port Id was set to 16 instead of using AFE_PORT_MAX macro. Fix this by properly using AFE_PORT_MAX macro. Fixes: 1b93a8843147 ("ASoC: qcom: sdm845: handle soundwire stream") Reported-by: John Stultz Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210309142129.14182-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 6c2760e27ea6..1e2c2d0902ea 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -33,12 +33,12 @@ struct sdm845_snd_data { struct snd_soc_jack jack; bool jack_setup; - bool stream_prepared[SLIM_MAX_RX_PORTS]; + bool stream_prepared[AFE_PORT_MAX]; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count; uint32_t quat_tdm_clk_count; - struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS]; + struct sdw_stream_runtime *sruntime[AFE_PORT_MAX]; }; static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; -- cgit v1.2.3 From 4800fe6ea1022eb240215b1743d2541adad8efc7 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 9 Mar 2021 14:21:28 +0000 Subject: ASoC: qcom: sdm845: Fix array out of range on rx slim channels WCD934x has only 13 RX SLIM ports however we are setting it as 16 in set_channel_map, this will lead to array out of bounds error! Orignally caught by enabling USBAN array out of bounds check: Fixes: 5caf64c633a3 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga") Reported-by: John Stultz Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210309142129.14182-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 1e2c2d0902ea..153e9b2de0b5 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -27,7 +27,7 @@ #define SPK_TDM_RX_MASK 0x03 #define NUM_TDM_SLOTS 8 #define SLIM_MAX_TX_PORTS 16 -#define SLIM_MAX_RX_PORTS 16 +#define SLIM_MAX_RX_PORTS 13 #define WCD934X_DEFAULT_MCLK_RATE 9600000 struct sdm845_snd_data { -- cgit v1.2.3 From 3bb4852d598f0275ed5996a059df55be7318ac2f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 9 Mar 2021 14:21:29 +0000 Subject: ASoC: codecs: wcd934x: add a sanity check in set channel map set channel map can be passed with a channel maps, however if the number of channels that are passed are more than the actual supported channels then we would be accessing array out of bounds. So add a sanity check to validate these numbers! Fixes: a61f3b4f476e ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec") Reported-by: John Stultz Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 40f682f5dab8..d18ae5e3ee80 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1873,6 +1873,12 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai, wcd = snd_soc_component_get_drvdata(dai->component); + if (tx_num > WCD934X_TX_MAX || rx_num > WCD934X_RX_MAX) { + dev_err(wcd->dev, "Invalid tx %d or rx %d channel count\n", + tx_num, rx_num); + return -EINVAL; + } + if (!tx_slot || !rx_slot) { dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n", tx_slot, rx_slot); -- cgit v1.2.3 From fd8299181995093948ec6ca75432e797b4a39143 Mon Sep 17 00:00:00 2001 From: Pan Xiuli Date: Mon, 8 Mar 2021 18:41:27 -0600 Subject: ASoC: SOF: intel: fix wrong poll bits in dsp power down The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered down, we need to check ADSPCS_CPA, the Current Power Active bit. Fixes: 747503b1813a3 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations") Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210309004127.4940-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 5788fe356960..c3b757cf01a0 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -207,7 +207,7 @@ int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask) ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, adspcs, - !(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)), + !(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)), HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC); if (ret < 0) -- cgit v1.2.3 From ea94191e584b146878f0b7fd4b767500d7aae870 Mon Sep 17 00:00:00 2001 From: Meng Li Date: Thu, 11 Mar 2021 17:12:20 +0800 Subject: spi: cadence: set cqspi to the driver_data field of struct device When initialize cadence qspi controller, it is need to set cqspi to the driver_data field of struct device, because it will be used in function cqspi_remove/suspend/resume(). Otherwise, there will be a crash trace as below when invoking these finctions. Fixes: 31fb632b5d43 ("spi: Move cadence-quadspi driver to drivers/spi/") Cc: stable@vger.kernel.org Signed-off-by: Meng Li Link: https://lore.kernel.org/r/20210311091220.3615-1-Meng.Li@windriver.com Signed-off-by: Mark Brown --- drivers/spi/spi-cadence-quadspi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/spi/spi-cadence-quadspi.c b/drivers/spi/spi-cadence-quadspi.c index 442cc7c53a47..52ddb3255d88 100644 --- a/drivers/spi/spi-cadence-quadspi.c +++ b/drivers/spi/spi-cadence-quadspi.c @@ -1433,6 +1433,7 @@ static int cqspi_probe(struct platform_device *pdev) cqspi = spi_master_get_devdata(master); cqspi->pdev = pdev; + platform_set_drvdata(pdev, cqspi); /* Obtain configuration from OF. */ ret = cqspi_of_get_pdata(cqspi); -- cgit v1.2.3 From 9922f50f7178496e709d3d064920b5031f0d9061 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 11 Mar 2021 21:15:57 +0530 Subject: ASoC: qcom: lpass-cpu: Fix lpass dai ids parse The max boundary check while parsing dai ids makes sound card registration fail after common up dai ids. Fixes: cd3484f7f138 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY") Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/20210311154557.24978-1-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c642e5f8f28c..be360a402b67 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -739,7 +739,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id < 0 || id >= data->variant->num_dai) { + if (ret || id < 0) { dev_err(dev, "valid dai id not found: %d\n", ret); continue; } -- cgit v1.2.3 From 8d06b9633a66f41fed520f6eebd163189518ba79 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Fri, 12 Mar 2021 10:26:45 +0800 Subject: ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3]. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/1615516005-781-1-git-send-email-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 +++- sound/soc/mediatek/mt8192/mt8192-reg.h | 8 +++++--- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c index f5de1d769679..f3bebed2428a 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c @@ -555,7 +555,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, /* set tdm */ if (tdm_priv->bck_invert) - tdm_con |= 1 << BCK_INVERSE_SFT; + regmap_update_bits(afe->regmap, AUDIO_TOP_CON3, + BCK_INVERSE_MASK_SFT, + 0x1 << BCK_INVERSE_SFT); if (tdm_priv->lck_invert) tdm_con |= 1 << LRCK_INVERSE_SFT; diff --git a/sound/soc/mediatek/mt8192/mt8192-reg.h b/sound/soc/mediatek/mt8192/mt8192-reg.h index 562f25c79c34..b9fb80d4afec 100644 --- a/sound/soc/mediatek/mt8192/mt8192-reg.h +++ b/sound/soc/mediatek/mt8192/mt8192-reg.h @@ -21,6 +21,11 @@ enum { /***************************************************************************** * R E G I S T E R D E F I N I T I O N *****************************************************************************/ +/* AUDIO_TOP_CON3 */ +#define BCK_INVERSE_SFT 3 +#define BCK_INVERSE_MASK 0x1 +#define BCK_INVERSE_MASK_SFT (0x1 << 3) + /* AFE_DAC_CON0 */ #define VUL12_ON_SFT 31 #define VUL12_ON_MASK 0x1 @@ -2079,9 +2084,6 @@ enum { #define TDM_EN_SFT 0 #define TDM_EN_MASK 0x1 #define TDM_EN_MASK_SFT (0x1 << 0) -#define BCK_INVERSE_SFT 1 -#define BCK_INVERSE_MASK 0x1 -#define BCK_INVERSE_MASK_SFT (0x1 << 1) #define LRCK_INVERSE_SFT 2 #define LRCK_INVERSE_MASK 0x1 #define LRCK_INVERSE_MASK_SFT (0x1 << 2) -- cgit v1.2.3 From 8ca88d53351cc58d535b2bfc7386835378fb0db2 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Mon, 15 Mar 2021 23:01:31 +0530 Subject: ASoC: simple-card-utils: Do not handle device clock This reverts commit 1e30f642cf29 ("ASoC: simple-card-utils: Fix device module clock"). The original patch ended up breaking following platform, which depends on set_sysclk() to configure internal PLL on wm8904 codec and expects simple-card-utils to not update the MCLK rate. - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts" It would be best if codec takes care of setting MCLK clock via DAI set_sysclk() callback. Reported-by: Michael Walle Suggested-by: Mark Brown Suggested-by: Michael Walle Fixes: 1e30f642cf29 ("ASoC: simple-card-utils: Fix device module clock") Signed-off-by: Sameer Pujar Tested-by: Michael Walle Link: https://lore.kernel.org/r/1615829492-8972-2-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index ab31045cfc95..6cada4c1e283 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -172,15 +172,16 @@ int asoc_simple_parse_clk(struct device *dev, * or device's module clock. */ clk = devm_get_clk_from_child(dev, node, NULL); - if (IS_ERR(clk)) - clk = devm_get_clk_from_child(dev, dlc->of_node, NULL); - if (!IS_ERR(clk)) { - simple_dai->clk = clk; simple_dai->sysclk = clk_get_rate(clk); - } else if (!of_property_read_u32(node, "system-clock-frequency", - &val)) { + + simple_dai->clk = clk; + } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; + } else { + clk = devm_get_clk_from_child(dev, dlc->of_node, NULL); + if (!IS_ERR(clk)) + simple_dai->sysclk = clk_get_rate(clk); } if (of_property_read_bool(node, "system-clock-direction-out")) -- cgit v1.2.3 From dbf54a9534350d6aebbb34f5c1c606b81a4f35dd Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Mon, 15 Mar 2021 23:01:32 +0530 Subject: ASoC: rt5659: Update MCLK rate in set_sysclk() Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown Suggested-by: Michael Walle Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 41e5917b16a5..91a4ef7f620c 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- cgit v1.2.3 From 899b12542b0897f92de9ba30944937c39ebb246d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 16 Mar 2021 08:52:54 +0800 Subject: ASoC: rt711: add snd_soc_component remove callback We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Rander Wang Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 85f744184a60..047f4e677d78 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } +static void rt711_remove(struct snd_soc_component *component) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt711->regmap, true); +} + static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, + .remove = rt711_remove, }; static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, -- cgit v1.2.3 From 9deef665f5811a7ad22b5e6eb80fe2a14ba4494c Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 16 Mar 2021 16:40:53 +0800 Subject: ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms Add compatible string for new added platforms which support spdif module. They are i.MX8QXP, i.MX8MM, i.MX8MN, i.MX8MQ. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1615884053-4264-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 50449b6d1048..4454aca34d56 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -21,6 +21,10 @@ properties: - fsl,vf610-spdif - fsl,imx6sx-spdif - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8mq-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif reg: maxItems: 1 -- cgit v1.2.3