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-rw-r--r--sound/core/timer.c4
-rw-r--r--sound/hda/codecs/realtek/alc269.c31
-rw-r--r--sound/hda/codecs/side-codecs/tas2781_hda_i2c.c4
-rw-r--r--sound/hda/codecs/side-codecs/tas2781_hda_spi.c6
-rw-r--r--sound/soc/codecs/cs35l56-sdw.c69
-rw-r--r--sound/soc/codecs/cs35l56-shared.c29
-rw-r--r--sound/soc/codecs/cs35l56.c2
-rw-r--r--sound/soc/codecs/cs35l56.h3
-rw-r--r--sound/soc/codecs/es8389.c2
-rw-r--r--sound/soc/codecs/tas2781-i2c.c4
-rw-r--r--sound/usb/stream.c2
-rw-r--r--sound/usb/validate.c2
12 files changed, 62 insertions, 96 deletions
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 3ce12264eed8..d9fff5c87613 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -2139,14 +2139,14 @@ static int snd_utimer_create(struct snd_timer_uinfo *utimer_info,
goto err_take_id;
}
+ utimer->id = utimer_id;
+
utimer->name = kasprintf(GFP_KERNEL, "snd-utimer%d", utimer_id);
if (!utimer->name) {
err = -ENOMEM;
goto err_get_name;
}
- utimer->id = utimer_id;
-
tid.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION;
tid.dev_class = SNDRV_TIMER_CLASS_GLOBAL;
tid.card = -1;
diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c
index e90c4047ea62..0323606b3d6d 100644
--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -510,6 +510,15 @@ static void alc256_shutup(struct hda_codec *codec)
hp_pin = 0x21;
alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */
+
+ /* 3k pull low control for Headset jack. */
+ /* NOTE: call this before clearing the pin, otherwise codec stalls */
+ /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
+ * when booting with headset plugged. So skip setting it for the codec alc257
+ */
+ if (spec->en_3kpull_low)
+ alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
+
hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
if (hp_pin_sense) {
@@ -520,14 +529,6 @@ static void alc256_shutup(struct hda_codec *codec)
msleep(75);
- /* 3k pull low control for Headset jack. */
- /* NOTE: call this before clearing the pin, otherwise codec stalls */
- /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
- * when booting with headset plugged. So skip setting it for the codec alc257
- */
- if (spec->en_3kpull_low)
- alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
-
if (!spec->no_shutup_pins)
snd_hda_codec_write(codec, hp_pin, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
@@ -3579,6 +3580,7 @@ enum {
ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
ALC294_FIXUP_ASUS_MIC,
ALC294_FIXUP_ASUS_HEADSET_MIC,
+ ALC294_FIXUP_ASUS_I2C_HEADSET_MIC,
ALC294_FIXUP_ASUS_SPK,
ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE,
ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE,
@@ -4889,6 +4891,15 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MIC
},
+ [ALC294_FIXUP_ASUS_I2C_HEADSET_MIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a19020 }, /* use as headset mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC287_FIXUP_CS35L41_I2C_2
+ },
[ALC294_FIXUP_ASUS_SPK] = {
.type = HDA_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -6368,6 +6379,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+ SND_PCI_QUIRK(0x103c, 0x8548, "HP EliteBook x360 830 G6", ALC285_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x854a, "HP EliteBook 830 G6", ALC285_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11),
SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360),
SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT),
@@ -6728,7 +6741,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2),
+ SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC294_FIXUP_ASUS_I2C_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2),
diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c
index 06c7bc2b9e9d..e34b17f0c9b9 100644
--- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c
+++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c
@@ -267,7 +267,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = {
static const struct snd_kcontrol_new tas2781_snd_controls[] = {
ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL,
1, 0, 20, 0, tas2781_amp_getvol,
- tas2781_amp_putvol, amp_vol_tlv),
+ tas2781_amp_putvol, tas2781_amp_tlv),
ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0,
tas2781_force_fwload_get, tas2781_force_fwload_put),
};
@@ -305,7 +305,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h)
efi_char16_t efi_name[TAS2563_CAL_VAR_NAME_MAX];
unsigned long max_size = TAS2563_CAL_DATA_SIZE;
unsigned char var8[TAS2563_CAL_VAR_NAME_MAX];
- struct tasdevice_priv *p = h->hda_priv;
+ struct tasdevice_priv *p = h->priv;
struct calidata *cd = &p->cali_data;
struct cali_reg *r = &cd->cali_reg_array;
unsigned int offset = 0;
diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c
index 09a5d0f131b2..b9a55672bf15 100644
--- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c
+++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c
@@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol,
static struct snd_kcontrol_new tas2781_snd_ctls[] = {
ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0,
- tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv),
+ tas2781_amp_getvol, tas2781_amp_putvol,
+ tas2781_amp_tlv),
ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1,
- tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv),
+ tas2781_digital_getvol, tas2781_digital_putvol,
+ tas2781_dvc_tlv),
ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get,
tas2781_force_fwload_put),
};
diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c
index ee14031695a1..3905c9cb188a 100644
--- a/sound/soc/codecs/cs35l56-sdw.c
+++ b/sound/soc/codecs/cs35l56-sdw.c
@@ -393,74 +393,6 @@ static int cs35l56_sdw_update_status(struct sdw_slave *peripheral,
return 0;
}
-static int cs35l63_sdw_kick_divider(struct cs35l56_private *cs35l56,
- struct sdw_slave *peripheral)
-{
- unsigned int curr_scale_reg, next_scale_reg;
- int curr_scale, next_scale, ret;
-
- if (!cs35l56->base.init_done)
- return 0;
-
- if (peripheral->bus->params.curr_bank) {
- curr_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B1;
- next_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B0;
- } else {
- curr_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B0;
- next_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B1;
- }
-
- /*
- * Current clock scale value must be different to new value.
- * Modify current to guarantee this. If next still has the dummy
- * value we wrote when it was current, the core code has not set
- * a new scale so restore its original good value
- */
- curr_scale = sdw_read_no_pm(peripheral, curr_scale_reg);
- if (curr_scale < 0) {
- dev_err(cs35l56->base.dev, "Failed to read current clock scale: %d\n", curr_scale);
- return curr_scale;
- }
-
- next_scale = sdw_read_no_pm(peripheral, next_scale_reg);
- if (next_scale < 0) {
- dev_err(cs35l56->base.dev, "Failed to read next clock scale: %d\n", next_scale);
- return next_scale;
- }
-
- if (next_scale == CS35L56_SDW_INVALID_BUS_SCALE) {
- next_scale = cs35l56->old_sdw_clock_scale;
- ret = sdw_write_no_pm(peripheral, next_scale_reg, next_scale);
- if (ret < 0) {
- dev_err(cs35l56->base.dev, "Failed to modify current clock scale: %d\n",
- ret);
- return ret;
- }
- }
-
- cs35l56->old_sdw_clock_scale = curr_scale;
- ret = sdw_write_no_pm(peripheral, curr_scale_reg, CS35L56_SDW_INVALID_BUS_SCALE);
- if (ret < 0) {
- dev_err(cs35l56->base.dev, "Failed to modify current clock scale: %d\n", ret);
- return ret;
- }
-
- dev_dbg(cs35l56->base.dev, "Next bus scale: %#x\n", next_scale);
-
- return 0;
-}
-
-static int cs35l56_sdw_bus_config(struct sdw_slave *peripheral,
- struct sdw_bus_params *params)
-{
- struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev);
-
- if ((cs35l56->base.type == 0x63) && (cs35l56->base.rev < 0xa1))
- return cs35l63_sdw_kick_divider(cs35l56, peripheral);
-
- return 0;
-}
-
static int __maybe_unused cs35l56_sdw_clk_stop(struct sdw_slave *peripheral,
enum sdw_clk_stop_mode mode,
enum sdw_clk_stop_type type)
@@ -476,7 +408,6 @@ static const struct sdw_slave_ops cs35l56_sdw_ops = {
.read_prop = cs35l56_sdw_read_prop,
.interrupt_callback = cs35l56_sdw_interrupt,
.update_status = cs35l56_sdw_update_status,
- .bus_config = cs35l56_sdw_bus_config,
#ifdef DEBUG
.clk_stop = cs35l56_sdw_clk_stop,
#endif
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index ba653f6ccfae..850fcf385996 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/sound/soc/codecs/cs35l56-shared.c
@@ -838,6 +838,15 @@ const struct cirrus_amp_cal_controls cs35l56_calibration_controls = {
};
EXPORT_SYMBOL_NS_GPL(cs35l56_calibration_controls, "SND_SOC_CS35L56_SHARED");
+static const struct cirrus_amp_cal_controls cs35l63_calibration_controls = {
+ .alg_id = 0xbf210,
+ .mem_region = WMFW_ADSP2_YM,
+ .ambient = "CAL_AMBIENT",
+ .calr = "CAL_R",
+ .status = "CAL_STATUS",
+ .checksum = "CAL_CHECKSUM",
+};
+
int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base)
{
u64 silicon_uid = 0;
@@ -912,19 +921,31 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_read_prot_status, "SND_SOC_CS35L56_SHARED");
void cs35l56_log_tuning(struct cs35l56_base *cs35l56_base, struct cs_dsp *cs_dsp)
{
__be32 pid, sid, tid;
+ unsigned int alg_id;
int ret;
+ switch (cs35l56_base->type) {
+ case 0x54:
+ case 0x56:
+ case 0x57:
+ alg_id = 0x9f212;
+ break;
+ default:
+ alg_id = 0xbf212;
+ break;
+ }
+
scoped_guard(mutex, &cs_dsp->pwr_lock) {
ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_PRJCT_ID",
- WMFW_ADSP2_XM, 0x9f212),
+ WMFW_ADSP2_XM, alg_id),
0, &pid, sizeof(pid));
if (!ret)
ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_CHNNL_ID",
- WMFW_ADSP2_XM, 0x9f212),
+ WMFW_ADSP2_XM, alg_id),
0, &sid, sizeof(sid));
if (!ret)
ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_SNPSHT_ID",
- WMFW_ADSP2_XM, 0x9f212),
+ WMFW_ADSP2_XM, alg_id),
0, &tid, sizeof(tid));
}
@@ -974,8 +995,10 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base)
case 0x35A54:
case 0x35A56:
case 0x35A57:
+ cs35l56_base->calibration_controls = &cs35l56_calibration_controls;
break;
case 0x35A630:
+ cs35l56_base->calibration_controls = &cs35l63_calibration_controls;
devid = devid >> 4;
break;
default:
diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c
index b1c65d8331e7..2c1edbd636ef 100644
--- a/sound/soc/codecs/cs35l56.c
+++ b/sound/soc/codecs/cs35l56.c
@@ -695,7 +695,7 @@ static int cs35l56_write_cal(struct cs35l56_private *cs35l56)
return ret;
ret = cs_amp_write_cal_coeffs(&cs35l56->dsp.cs_dsp,
- &cs35l56_calibration_controls,
+ cs35l56->base.calibration_controls,
&cs35l56->base.cal_data);
wm_adsp_stop(&cs35l56->dsp);
diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h
index bd77a57249d7..40a1800a4585 100644
--- a/sound/soc/codecs/cs35l56.h
+++ b/sound/soc/codecs/cs35l56.h
@@ -20,8 +20,6 @@
#define CS35L56_SDW_GEN_INT_MASK_1 0xc1
#define CS35L56_SDW_INT_MASK_CODEC_IRQ BIT(0)
-#define CS35L56_SDW_INVALID_BUS_SCALE 0xf
-
#define CS35L56_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
#define CS35L56_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \
| SNDRV_PCM_FMTBIT_S32_LE)
@@ -52,7 +50,6 @@ struct cs35l56_private {
u8 asp_slot_count;
bool tdm_mode;
bool sysclk_set;
- u8 old_sdw_clock_scale;
u8 sdw_link_num;
u8 sdw_unique_id;
};
diff --git a/sound/soc/codecs/es8389.c b/sound/soc/codecs/es8389.c
index ba1763f36f17..6e4c75d288ef 100644
--- a/sound/soc/codecs/es8389.c
+++ b/sound/soc/codecs/es8389.c
@@ -636,7 +636,7 @@ static int es8389_set_bias_level(struct snd_soc_component *component,
regmap_write(es8389->regmap, ES8389_ANA_CTL1, 0x59);
regmap_write(es8389->regmap, ES8389_ADC_EN, 0x00);
regmap_write(es8389->regmap, ES8389_CLK_OFF1, 0x00);
- regmap_write(es8389->regmap, ES8389_RESET, 0x7E);
+ regmap_write(es8389->regmap, ES8389_RESET, 0x3E);
regmap_update_bits(es8389->regmap, ES8389_DAC_INV, 0x80, 0x80);
usleep_range(8000, 8500);
regmap_update_bits(es8389->regmap, ES8389_DAC_INV, 0x80, 0x00);
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index 676130f4cf3e..0e09d794516f 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -910,10 +910,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = {
static const struct snd_kcontrol_new tas2781_snd_controls[] = {
SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL,
1, 0, 20, 0, tas2781_amp_getvol,
- tas2781_amp_putvol, amp_vol_tlv),
+ tas2781_amp_putvol, tas2781_amp_tlv),
SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL,
0, 0, 200, 1, tas2781_digital_getvol,
- tas2781_digital_putvol, dvc_tlv),
+ tas2781_digital_putvol, tas2781_dvc_tlv),
};
static const struct snd_kcontrol_new tas2781_cali_controls[] = {
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index acf3dc2d79e0..5c235a5ba7e1 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -349,7 +349,7 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor
u16 cs_len;
u8 cs_type;
- if (len < sizeof(*p))
+ if (len < sizeof(*cs_desc))
break;
cs_len = le16_to_cpu(cs_desc->wLength);
if (len < cs_len)
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
index 4f4e8e87a14c..a0d55b77c994 100644
--- a/sound/usb/validate.c
+++ b/sound/usb/validate.c
@@ -285,7 +285,7 @@ static const struct usb_desc_validator audio_validators[] = {
/* UAC_VERSION_3, UAC3_EXTENDED_TERMINAL: not implemented yet */
FUNC(UAC_VERSION_3, UAC3_MIXER_UNIT, validate_mixer_unit),
FUNC(UAC_VERSION_3, UAC3_SELECTOR_UNIT, validate_selector_unit),
- FUNC(UAC_VERSION_3, UAC_FEATURE_UNIT, validate_uac3_feature_unit),
+ FUNC(UAC_VERSION_3, UAC3_FEATURE_UNIT, validate_uac3_feature_unit),
/* UAC_VERSION_3, UAC3_EFFECT_UNIT: not implemented yet */
FUNC(UAC_VERSION_3, UAC3_PROCESSING_UNIT, validate_processing_unit),
FUNC(UAC_VERSION_3, UAC3_EXTENSION_UNIT, validate_processing_unit),
generated by cgit v1.2.3 (git 2.39.0) at 2025-08-23 12:59:42 +0300