June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.15 02/41] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

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[PATCH AUTOSEL 5.17 27/43] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 112ecc256b14..7e7e517d32c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10092,6 +10104,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11542,6 +11555,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

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[PATCH AUTOSEL 5.17 18/43] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..93677b9f3a68 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -328,7 +328,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

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[PATCH AUTOSEL 5.17 17/43] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..ca3b1c00fa78 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

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[PATCH AUTOSEL 5.17 16/43] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a5584ba962dc..e97eaca7049f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3858,6 +3858,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

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[PATCH AUTOSEL 5.17 12/43] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.17 11/43] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.17 10/43] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.17 09/43] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

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[PATCH AUTOSEL 5.17 08/43] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

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