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[PATCH] ASoC: rt5682: fix a little pop while playback
by derek.fang＠realtek.com 14 Oct '21

14 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> A little pop can be heard obviously from HP while playing a silent. This patch fixes it by using two functions: 1. Enable HP 1bit output mode. 2. Change the charge pump switch size during playback on and off. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 56 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt5682.h | 20 ++++++++++++++ 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 914fe7debc05..78b4cb5fb6c8 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -46,6 +46,8 @@ static const struct reg_sequence patch_list[] = { {RT5682_SAR_IL_CMD_1, 0x22b7}, {RT5682_SAR_IL_CMD_3, 0x0365}, {RT5682_SAR_IL_CMD_6, 0x0110}, + {RT5682_CHARGE_PUMP_1, 0x0210}, + {RT5682_HP_LOGIC_CTRL_2, 0x0007}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -1515,21 +1517,29 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_write(component, - RT5682_HP_LOGIC_CTRL_2, 0x0012); - snd_soc_component_write(component, - RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_AMP_MUTE, 0); + snd_soc_component_update_bits(component, RT5682_HP_LOGIC_CTRL_2, + RT5682_HP_LC2_SIG_SOUR2_MASK, RT5682_HP_LC2_SIG_SOUR2_REG); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN); + usleep_range(5000, 10000); + snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, + RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L); break; case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, 0); + snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, + RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_M); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x0); - snd_soc_component_write(component, - RT5682_HP_CTRL_2, 0x0000); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; @@ -1637,6 +1647,23 @@ static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); +static const unsigned int rt5682_hpo_sig_out_values[] = { + 2, + 7, +}; + +static const char * const rt5682_hpo_sig_out_mode[] = { + "Legacy", + "OneBit", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_hpo_sig_out_enum, + RT5682_HP_LOGIC_CTRL_2, 0, RT5682_HP_LC2_SIG_SOUR1_MASK, + rt5682_hpo_sig_out_mode, rt5682_hpo_sig_out_values); + +static const struct snd_kcontrol_new rt5682_hpo_sig_demux = + SOC_DAPM_ENUM("HPO Signal Demux", rt5682_hpo_sig_out_enum); + static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, 0, NULL, 0), @@ -1820,6 +1847,10 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, &hpor_switch), + SND_SOC_DAPM_OUT_DRV("HPO Legacy", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HPO OneBit", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_DEMUX("HPO Signal Demux", SND_SOC_NOPM, 0, 0, &rt5682_hpo_sig_demux), + /* CLK DET */ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1987,10 +2018,19 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "Vref1"}, - {"HPOL Playback", "Switch", "HP Amp"}, - {"HPOR Playback", "Switch", "HP Amp"}, + + {"HPO Signal Demux", NULL, "HP Amp"}, + + {"HPO Legacy", "Legacy", "HPO Signal Demux"}, + {"HPO OneBit", "OneBit", "HPO Signal Demux"}, + + {"HPOL Playback", "Switch", "HPO Legacy"}, + {"HPOR Playback", "Switch", "HPO Legacy"}, + {"HPOL", NULL, "HPOL Playback"}, {"HPOR", NULL, "HPOR Playback"}, + {"HPOL", NULL, "HPO OneBit"}, + {"HPOR", NULL, "HPO OneBit"}, }; static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 262512babc50..d93829c35585 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -375,6 +375,14 @@ #define RT5682_R_VOL_MASK (0x3f) #define RT5682_R_VOL_SFT 0 +/* Headphone Amp Control 2 (0x0003) */ +#define RT5682_HP_C2_DAC_AMP_MUTE_SFT 15 +#define RT5682_HP_C2_DAC_AMP_MUTE (0x1 << 15) +#define RT5682_HP_C2_DAC_L_EN_SFT 14 +#define RT5682_HP_C2_DAC_L_EN (0x1 << 14) +#define RT5682_HP_C2_DAC_R_EN_SFT 13 +#define RT5682_HP_C2_DAC_R_EN (0x1 << 13) + /*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ #define RT5682_G_HP (0xf << 8) #define RT5682_G_HP_SFT 8 @@ -1265,6 +1273,10 @@ #define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) /* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_SW_SIZE_MASK (0x7 << 8) +#define RT5682_CP_SW_SIZE_L (0x4 << 8) +#define RT5682_CP_SW_SIZE_M (0x2 << 8) +#define RT5682_CP_SW_SIZE_S (0x1 << 8) #define RT5682_CP_CLK_HP_MASK (0x3 << 4) #define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) #define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) @@ -1315,6 +1327,14 @@ #define RT5682_DEB_STO_DAC_MASK (0x7 << 4) #define RT5682_DEB_80_MS (0x0 << 4) +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5682_HP_LC2_SIG_SOUR2_MASK (0x1 << 4) +#define RT5682_HP_LC2_SIG_SOUR2_REG (0x1 << 4) +#define RT5682_HP_LC2_SIG_SOUR2_DC_CAL (0x0 << 4) +#define RT5682_HP_LC2_SIG_SOUR1_MASK (0x7) +#define RT5682_HP_LC2_SIG_SOUR1_1BIT (0x7) +#define RT5682_HP_LC2_SIG_SOUR1_LEGA (0x2) + /* SAR ADC Inline Command Control 1 (0x0210) */ #define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) #define RT5682_SAR_BUTT_DET_EN (0x1 << 15) -- 2.17.1

2 1

[PATCH] ALSA: usb-audio: add Schiit Hel device to quirk table
by Greg Kroah-Hartman 14 Oct '21

14 Oct '21

The Shciit Hel device responds to the ctl message for the mic capture switch with a timeout of -EPIPE: usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 This seems safe to ignore as the device works properly with the control message quirk, so add it to the quirk table so all is good. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-usb(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6ee6d24c847f..ec11be366dc0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1884,6 +1884,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ + QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ -- 2.33.0

2 1

[RFC] sound: cs35l41: Add support for Legion 7 16ACHg6 laptop
by Lucas Tanure 14 Oct '21

14 Oct '21

Hi, I would like to get some guidance about this solution to support the 16ACHg6 laptop. Hardware: - The 16ACHg6 laptop has two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - To load the firmware for CS35L41, this solution will require the wm_adsp library, which requires regmap, header definitions and register tables. - To minimize the duplication of the code, the HDA functions will be placed inside the ASoC CS35L41 driver. - Finally, HDA patch_realtek will access exposed functions from ASoC CS35L41 driver to initialize the amplifiers, start and stop streams and load firmware. Notes: - This is a work in progress, so the code is not functional, its only intent is to demonstrate the overall solution - If accepted, this will be split into a couple of patches for a new patch chain Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- drivers/acpi/scan.c | 1 + drivers/platform/x86/i2c-multi-instantiate.c | 7 ++ include/sound/cs35l41.h | 4 ++ sound/pci/hda/Kconfig | 1 + sound/pci/hda/patch_realtek.c | 21 +++++- sound/soc/codecs/cs35l41.c | 75 ++++++++++++++++---- 6 files changed, 95 insertions(+), 14 deletions(-) diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 5b54c80b9d32..c1c27a408420 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1703,6 +1703,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"BSG2150", }, {"INT33FE", }, {"INT3515", }, + {"CLSA0100", }, {} }; diff --git a/drivers/platform/x86/i2c-multi-instantiate.c b/drivers/platform/x86/i2c-multi-instantiate.c index a50153ecd560..b61f7e30d42a 100644 --- a/drivers/platform/x86/i2c-multi-instantiate.c +++ b/drivers/platform/x86/i2c-multi-instantiate.c @@ -139,6 +139,12 @@ static const struct i2c_inst_data bsg2150_data[] = { {} }; +static const struct i2c_inst_data clsa0100_data[] = { + { "cs35l41", IRQ_RESOURCE_GPIO, 0 }, + { "cs35l41", IRQ_RESOURCE_GPIO, 0 }, + {} +}; + /* * Device with _HID INT3515 (TI PD controllers) has some unresolved interrupt * issues. The most common problem seen is interrupt flood. @@ -170,6 +176,7 @@ static const struct i2c_inst_data bsg2150_data[] = { static const struct acpi_device_id i2c_multi_inst_acpi_ids[] = { { "BSG1160", (unsigned long)bsg1160_data }, { "BSG2150", (unsigned long)bsg2150_data }, + { "CLSA0100", (unsigned long)clsa0100_data }, { } }; MODULE_DEVICE_TABLE(acpi, i2c_multi_inst_acpi_ids); diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h index 1f1e3c6c9be1..4d665b7dbfdf 100644 --- a/include/sound/cs35l41.h +++ b/include/sound/cs35l41.h @@ -23,6 +23,8 @@ struct cs35l41_irq_cfg { }; struct cs35l41_platform_data { + bool no_bst; + bool hda; int bst_ind; int bst_ipk; int bst_cap; @@ -31,4 +33,6 @@ struct cs35l41_platform_data { struct cs35l41_irq_cfg irq_config2; }; +void cs35l41_hda_init(void); + #endif /* __CS35L41_H */ diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ab9d2746e804..37202466f033 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -95,6 +95,7 @@ config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC select SND_HDA_GENERIC_LEDS + select SND_SOC_CS35L41_I2C help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..2a0ac9a1b613 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -21,6 +21,7 @@ #include <sound/core.h> #include <sound/jack.h> #include <sound/hda_codec.h> +#include <sound/cs35l41.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -6443,6 +6444,18 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, } } +static void alc287_fixup_lenovo_y760(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PROBE) { + codec_info(cdc, "HDA_FIXUP_ACT_PROBE\n"); + cs35l41_hda_init(); + } else if (action == HDA_FIXUP_ACT_INIT) { + codec_info(cdc, "HDA_FIXUP_ACT_INIT\n"); + } else if (action == HDA_FIXUP_ACT_FREE) { + codec_info(cdc, "HDA_FIXUP_ACT_FREE\n"); + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6663,7 +6676,8 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, - ALC287_FIXUP_13S_GEN2_SPEAKERS + ALC287_FIXUP_13S_GEN2_SPEAKERS, + ALC287_FIXUP_LENOVO_Y760 }; static const struct hda_fixup alc269_fixups[] = { @@ -8361,6 +8375,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC287_FIXUP_LENOVO_Y760] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_lenovo_y760, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8755,6 +8773,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3847, "Legion Y760", ALC287_FIXUP_LENOVO_Y760), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index b16eb6610c0e..f643ed1b48c0 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -21,9 +21,17 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include <linux/acpi.h> #include "cs35l41.h" +static struct list_head *cs35l41_hda_lst; + +struct cs35l41_hda_node { + struct list_head node; + struct cs35l41_private *cs35l41; +}; + static const char * const cs35l41_supplies[CS35L41_NUM_SUPPLIES] = { "VA", "VP", @@ -1039,9 +1047,7 @@ static int cs35l41_set_pdata(struct cs35l41_private *cs35l41) { int ret; - /* Set Platform Data */ - /* Required */ - if (cs35l41->pdata.bst_ipk && + if (!cs35l41->pdata.no_bst && cs35l41->pdata.bst_ipk && cs35l41->pdata.bst_ind && cs35l41->pdata.bst_cap) { ret = cs35l41_boost_config(cs35l41, cs35l41->pdata.bst_ind, cs35l41->pdata.bst_cap, @@ -1051,8 +1057,7 @@ static int cs35l41_set_pdata(struct cs35l41_private *cs35l41) return ret; } } else { - dev_err(cs35l41->dev, "Incomplete Boost component DT config\n"); - return -EINVAL; + dev_info(cs35l41->dev, "Boost disabled\n"); } /* Optional */ @@ -1148,9 +1153,31 @@ static int cs35l41_handle_pdata(struct device *dev, { struct cs35l41_irq_cfg *irq_gpio1_config = &pdata->irq_config1; struct cs35l41_irq_cfg *irq_gpio2_config = &pdata->irq_config2; + struct acpi_device *adev; + struct device *phys_dev; unsigned int val; int ret; + if (memcmp(dev_name(cs35l41->dev), "i2c-CLSA0100", 12) == 0) { + pdata->no_bst = true; + pdata->hda = true; + adev = acpi_dev_get_first_match_dev("CLSA0100", "1", -1); + if (!adev) { + dev_err(dev, "Failed to find an ACPI device\n"); + return -ENODEV; + } + + phys_dev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + if (!phys_dev) { + dev_err(dev, "Failed to find a physical device\n"); + return -ENODEV; + } + cs35l41->reset_gpio = gpiod_get_index(phys_dev, NULL, 0, GPIOD_ASIS); + return 0; + } + ret = device_property_read_u32(dev, "cirrus,boost-peak-milliamp", &val); if (ret >= 0) pdata->bst_ipk = val; @@ -1237,10 +1264,22 @@ static const struct reg_sequence cs35l41_revb2_errata_patch[] = { { 0x00000040, 0x00003333 }, }; +void cs35l41_hda_init(void) +{ + struct list_head *p; + int i = 0; + + list_for_each(p, cs35l41_hda_lst) { + pr_info("%s %d\n", __func__, i++); + } +} +EXPORT_SYMBOL_GPL(cs35l41_hda_init); + int cs35l41_probe(struct cs35l41_private *cs35l41, struct cs35l41_platform_data *pdata) { u32 regid, reg_revid, i, mtl_revid, int_status, chipid_match; + struct cs35l41_hda_node *cs35l41_hda; int irq_pol = 0; int ret; @@ -1269,8 +1308,8 @@ int cs35l41_probe(struct cs35l41_private *cs35l41, } /* returning NULL can be an option if in stereo mode */ - cs35l41->reset_gpio = devm_gpiod_get_optional(cs35l41->dev, "reset", - GPIOD_OUT_LOW); + if (!cs35l41->reset_gpio) + cs35l41->reset_gpio = devm_gpiod_get_optional(cs35l41->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l41->reset_gpio)) { ret = PTR_ERR(cs35l41->reset_gpio); cs35l41->reset_gpio = NULL; @@ -1413,12 +1452,22 @@ int cs35l41_probe(struct cs35l41_private *cs35l41, goto err; } - ret = devm_snd_soc_register_component(cs35l41->dev, - &soc_component_dev_cs35l41, - cs35l41_dai, ARRAY_SIZE(cs35l41_dai)); - if (ret < 0) { - dev_err(cs35l41->dev, "Register codec failed: %d\n", ret); - goto err; + if (!cs35l41->pdata.hda) { + ret = devm_snd_soc_register_component(cs35l41->dev, + &soc_component_dev_cs35l41, + cs35l41_dai, ARRAY_SIZE(cs35l41_dai)); + if (ret < 0) { + dev_err(cs35l41->dev, "Register codec failed: %d\n", ret); + goto err; + } + } else { + if (!cs35l41_hda_lst) { + cs35l41_hda_lst = devm_kzalloc(cs35l41->dev, sizeof(*cs35l41_hda_lst), + GFP_KERNEL); + INIT_LIST_HEAD(cs35l41_hda_lst); + } + cs35l41_hda = devm_kzalloc(cs35l41->dev, sizeof(*cs35l41_hda), GFP_KERNEL); + list_add(&cs35l41_hda->node, cs35l41_hda_lst); } dev_info(cs35l41->dev, "Cirrus Logic CS35L41 (%x), Revision: %02X\n", -- 2.33.0

3 3

[PATCH] ALSA: usb-audio: allow -EPIPE errors for some v2 messages
by Greg Kroah-Hartman 14 Oct '21

14 Oct '21

The Schiit Hel device does not like to respond to all get_ctl_value_v2() requests for some reason. This used to work in older kernels, but now with more strict checking, this failure causes the device to fail to work. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- This fixes the Shiit Hel device that I have. It used to work on older kernels (a year ago?), but stopped working for some reason and I didn't take the time to track it down. This change fixes the issue for me, but feels wrong for some reason. At least now I can use the device as a headphone driver, much better than the built-in one for my current machine... If needed, I can take the time to do bisection to track down the real issue here, it might be due to stricter endpoint checking in the USB core, but that feels wrong somehow. Here's the debugfs USB output for this device, showing the endpoints: T: Bus=07 Lev=02 Prnt=02 Port=01 Cnt=01 Dev#= 5 Spd=480 MxCh= 0 D: Ver= 2.00 Cls=ef(misc ) Sub=02 Prot=01 MxPS=64 #Cfgs= 1 P: Vendor=30be ProdID=0101 Rev=01.02 S: Manufacturer=Schiit Audio S: Product=Schiit Hel C: #Ifs= 4 Cfg#= 1 Atr=c0 MxPwr=0mA I: If#= 0 Alt= 0 #EPs= 1 Cls=01(audio) Sub=01 Prot=20 Driver=snd-usb-audio E: Ad=8f(I) Atr=03(Int.) MxPS= 6 Ivl=1ms I: If#= 1 Alt= 1 #EPs= 2 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=05(O) Atr=05(Isoc) MxPS= 104 Ivl=125us E: Ad=85(I) Atr=11(Isoc) MxPS= 4 Ivl=1ms I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=88(I) Atr=05(Isoc) MxPS= 156 Ivl=125us I: If#= 3 Alt= 0 #EPs= 1 Cls=03(HID ) Sub=00 Prot=00 Driver=usbhid E: Ad=84(I) Atr=03(Int.) MxPS= 16 Ivl=1ms Any other suggestions to fix this are welcome. sound/usb/mixer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a2ce535df14b..37d3d697776b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -371,11 +371,11 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, validx, idx, buf, size); snd_usb_unlock_shutdown(chip); - if (ret < 0) { + if ((ret < 0) && (ret != -EPIPE)) { error: usb_audio_err(chip, - "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, idx, cval->val_type); + "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, ret = %d\n", + request, validx, idx, cval->val_type, ret); return ret; } -- 2.33.0

2 11

[PATCH] ASoC: soc-compress: prevent the potentially use of null pointer
by Jiasheng Jiang 14 Oct '21

14 Oct '21

The codec_dai and the cpu_dai are from rtd. When the rtd->dais[0] or rtd->dais[rtd->num_cpus] is None, it could lead to the use of null pointer in the snd_soc_dai_stream_valid(). So it might be better to add the check before the use of snd_soc_dai_stream_valid(). Fixes: 467fece ("ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/soc-compress.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b4f5935..67c3df1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,12 +535,14 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - playback = 1; - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - capture = 1; + if (codec_dai && cpu_dai) { + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + playback = 1; + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + capture = 1; + } /* * Compress devices are unidirectional so only one of the directions -- 2.7.4

2 1

[PATCH] ASoC: dt-bindings: rockchip: i2s-tdm: Fix rockchip, i2s-[rt]x-route
by Geert Uytterhoeven 13 Oct '21

13 Oct '21

make dt_binding_check: Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml: properties:rockchip,i2s-rx-route: {'required': ['maxItems']} is not allowed for {'$ref': '/schemas/types.yaml#/definitions/uint32-array', 'description': 'Defines the mapping of I2S RX sdis to I2S data bus lines. By default, they are mapped one-to-one. rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0.', 'maxItems': 4, 'items': [{'enum': [0, 1, 2, 3]}]} hint: "maxItems" is not needed with an "items" list from schema $id: http://devicetree.org/meta-schemas/items.yaml# Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml: properties:rockchip,i2s-tx-route: {'required': ['maxItems']} is not allowed for {'$ref': '/schemas/types.yaml#/definitions/uint32-array', 'description': 'Defines the mapping of I2S TX sdos to I2S data bus lines. By default, they are mapped one-to-one. rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0.', 'maxItems': 4, 'items': [{'enum': [0, 1, 2, 3]}]} hint: "maxItems" is not needed with an "items" list from schema $id: http://devicetree.org/meta-schemas/items.yaml# Drop the "-" from the enums to fix this. Fixes: 510f1c133aedcf69 ("ASoC: dt-bindings: rockchip: add i2s-tdm bindings") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- v4/v5 differed slightly from what Rob suggested in his review of v3 https://lore.kernel.org/all/YS6kHkeOKCJMOOIf@robh.at.kernel.org/ --- Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml index dce8b4136ec6db4e..ce3e18b50230dc9b 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -112,7 +112,7 @@ properties: rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0. maxItems: 4 items: - - enum: [0, 1, 2, 3] + enum: [0, 1, 2, 3] rockchip,i2s-tx-route: $ref: /schemas/types.yaml#/definitions/uint32-array @@ -122,7 +122,7 @@ properties: rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0. maxItems: 4 items: - - enum: [0, 1, 2, 3] + enum: [0, 1, 2, 3] rockchip,io-multiplex: description: -- 2.25.1

2 1

[PATCH v2] ASoC: rt1011: Fix 'I2S Reference' enum control
by Peter Ujfalusi 13 Oct '21

13 Oct '21

There are several things the patch adding the support for 'I2S Reference' got wrong: - "None" selection is in fact equals to last selected reference - The custom put overrides RX/TX len, TDM slot sizes, etc - the enum is useless in most part for the reference tracking - there is no need for EXT control as there is a single bit in RT1011_TDM1_SET_1 register (bit 7) which selects the reference - it was using ucontrol->value.integer.value[0] in the put/get callbacks which causesed access to 'I2S Reference' enum with alsamixer to fail Complements: c3de683c4d1d ("ASoC: rt1011: Fix 'I2S Reference' enum control caused error") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v1: - Rebased on top of next - changed the tag from Fixes to Complements to make sure that the crash fix and the fix for the implementation stays together. This patch is an iteration on https://lore.kernel.org/alsa-devel/20211011144518.2518-1-peter.ujfalusi@lin… to fix 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011"). In essence it is reverting the original patch to use SOC_ENUM_SINGLE_DECL/SOC_ENUM to handle the bit to select the I2S reference. Regards, Peter sound/soc/codecs/rt1011.c | 53 ++++----------------------------------- sound/soc/codecs/rt1011.h | 7 ------ 2 files changed, 5 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index bdfcbb81fa19..297af7ff824c 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1312,54 +1312,12 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, } static const char * const rt1011_i2s_ref_texts[] = { - "None", "Left Channel", "Right Channel" + "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_EXT_DECL(rt1011_i2s_ref_enum, rt1011_i2s_ref_texts); - -static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.enumerated.item[0]; - - switch (i2s_ref_ch) { - case RT1011_I2S_REF_LEFT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - case RT1011_I2S_REF_RIGHT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - default: - dev_info(component->dev, "I2S Reference: Do nothing\n"); - } - - rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; - - return 0; -} - -static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - - ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; - - return 0; -} +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, + RT1011_TDM1_SET_1, 7, + rt1011_i2s_ref_texts); static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ @@ -1400,8 +1358,7 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), /* I2S Reference */ - SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, - rt1011_i2s_ref_get, rt1011_i2s_ref_put), + SOC_ENUM("I2S Reference", rt1011_i2s_ref_enum), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index afb2fad94216..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,12 +654,6 @@ enum { RT1011_AIFS }; -enum { - RT1011_I2S_REF_NONE, - RT1011_I2S_REF_LEFT_CH, - RT1011_I2S_REF_RIGHT_CH, -}; - /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -698,7 +692,6 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; - unsigned int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- 2.33.0

2 1

[PATCH] ASoC: rt1011: Fix 'I2S Reference' enum control
by Peter Ujfalusi 13 Oct '21

13 Oct '21

There are several things the patch adding the support for 'I2S Reference' got wrong: - "None" selection is in fact equals to last selected reference - The custom put overrides RX/TX len, TDM slot sizes, etc - the enum is useless in most part for the reference tracking - there is no need for EXT control as there is a single bit in RT1011_TDM1_SET_1 register (bit 7) which selects the reference - it was using ucontrol->value.integer.value[0] in the put/get callbacks which causes access to 'I2S Reference' enum with alsamixer to fail: $ alsamixer cannot load mixer controls: Invalid argument cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow It should have used ucontrol->value.enumerated.item[0], but since there is no need for the custom code, it does not really matter. Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, This patch is an iteration on https://lore.kernel.org/alsa-devel/20211011144518.2518-1-peter.ujfalusi@lin… to fix 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011"). In essence it is reverting the original patch to use SOC_ENUM_SINGLE_DECL/SOC_ENUM to handle the bit to select the I2S reference. Regards, Peter sound/soc/codecs/rt1011.c | 56 +++++---------------------------------- sound/soc/codecs/rt1011.h | 7 ----- 2 files changed, 6 insertions(+), 57 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 508597866dff..297af7ff824c 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,56 +1311,13 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref[] = { - "None", "Left Channel", "Right Channel" +static const char * const rt1011_i2s_ref_texts[] = { + "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, - rt1011_i2s_ref); - -static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.integer.value[0]; - - switch (i2s_ref_ch) { - case RT1011_I2S_REF_LEFT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - case RT1011_I2S_REF_RIGHT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - default: - dev_info(component->dev, "I2S Reference: Do nothing\n"); - } - - rt1011->i2s_ref = ucontrol->value.integer.value[0]; - - return 0; -} - -static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - - ucontrol->value.integer.value[0] = rt1011->i2s_ref; - - return 0; -} +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, + RT1011_TDM1_SET_1, 7, + rt1011_i2s_ref_texts); static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ @@ -1401,8 +1358,7 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), /* I2S Reference */ - SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, - rt1011_i2s_ref_get, rt1011_i2s_ref_put), + SOC_ENUM("I2S Reference", rt1011_i2s_ref_enum), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index afb2fad94216..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,12 +654,6 @@ enum { RT1011_AIFS }; -enum { - RT1011_I2S_REF_NONE, - RT1011_I2S_REF_LEFT_CH, - RT1011_I2S_REF_RIGHT_CH, -}; - /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -698,7 +692,6 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; - unsigned int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- 2.33.0

2 1

[PATCH v4 0/2] ASoC: rt9120: Add Richtek RT9120 supprot
by cy_huang 13 Oct '21

13 Oct '21

From: ChiYuan Huang <cy_huang(a)richtek.com> This patch series Add the Richtek RT9120 support. In v4: - Add 'classd_tlv' for 'SPK Gain Volume' control item. - Unify the tlv declaration to the postfix '_tlv'. - Fix 'digital_tlv' mute as 1 to declare the minimum is muted. In v3: - Add dvdd regulator binding to check the dvdd voltage domain. - Refine sdo_select_text. - Use switch case in 'internal_power_event' function. - Remove the volume and mute initially write in component probe. - Remove the mute API. It's no need by HW design. In v2: - Add missing #sound-dai-cells property. ChiYuan Huang (2): ASoC: dt-bindings: rt9120: Add initial bindings ASoC: rt9120: Add rt9210 audio amplifier support .../devicetree/bindings/sound/richtek,rt9120.yaml | 59 +++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt9120.c | 495 +++++++++++++++++++++ 4 files changed, 566 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml create mode 100644 sound/soc/codecs/rt9120.c -- 2.7.4

2 3

[PATCH v2] ASoC: wm8960: Fix clock configuration on slave mode
by Shengjiu Wang 13 Oct '21

13 Oct '21

There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - change error message to warning - add more comments sound/soc/codecs/wm8960.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392..499604f1e178 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,9 +742,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; } -- 2.17.1

3 2