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-rw-r--r--Documentation/devicetree/bindings/sound/st,stm32-sai.yaml2
-rw-r--r--Documentation/sound/hd-audio/notes.rst16
-rw-r--r--include/sound/tas2781-dsp.h7
-rw-r--r--sound/core/init.c12
-rw-r--r--sound/core/timer.c8
-rw-r--r--sound/pci/hda/patch_realtek.c12
-rw-r--r--sound/soc/codecs/es8326.c6
-rw-r--r--sound/soc/codecs/rt5645.c1
-rw-r--r--sound/soc/codecs/tas2552.c15
-rw-r--r--sound/soc/codecs/tas2781-fmwlib.c103
-rw-r--r--sound/soc/codecs/tas2781-i2c.c4
-rw-r--r--sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c2
-rw-r--r--sound/usb/quirks.c4
13 files changed, 86 insertions, 106 deletions
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
index b46a4778807d..68f97b462598 100644
--- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
@@ -72,7 +72,7 @@ patternProperties:
properties:
compatible:
description: Compatible for SAI sub-block A or B.
- pattern: "st,stm32-sai-sub-[ab]"
+ pattern: "^st,stm32-sai-sub-[ab]$"
"#sound-dai-cells":
const: 0
diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst
index a9e35b1f87bd..ef6a4513cce7 100644
--- a/Documentation/sound/hd-audio/notes.rst
+++ b/Documentation/sound/hd-audio/notes.rst
@@ -15,7 +15,7 @@ problem is broken BIOS, and the rest is the driver implementation.
This document explains the brief trouble-shooting and debugging
methods for the HD-audio hardware.
-The HD-audio component consists of two parts: the controller chip and
+The HD-audio component consists of two parts: the controller chip and
the codec chips on the HD-audio bus. Linux provides a single driver
for all controllers, snd-hda-intel. Although the driver name contains
a word of a well-known hardware vendor, it's not specific to it but for
@@ -81,7 +81,7 @@ the wake-up timing. It wakes up a few samples before actually
processing the data on the buffer. This caused a lot of problems, for
example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts
an artificial delay to the wake up timing. This delay is controlled
-via ``bdl_pos_adj`` option.
+via ``bdl_pos_adj`` option.
When ``bdl_pos_adj`` is a negative value (as default), it's assigned to
an appropriate value depending on the controller chip. For Intel
@@ -144,7 +144,7 @@ see a regression wrt the sound quality (stuttering, etc) or a lock-up
in the recent kernel, try to pass ``enable_msi=0`` option to disable
MSI. If it works, you can add the known bad device to the blacklist
defined in hda_intel.c. In such a case, please report and give the
-patch back to the upstream developer.
+patch back to the upstream developer.
HD-Audio Codec
@@ -375,7 +375,7 @@ HD-Audio Reconfiguration
------------------------
This is an experimental feature to allow you re-configure the HD-audio
codec dynamically without reloading the driver. The following sysfs
-files are available under each codec-hwdep device directory (e.g.
+files are available under each codec-hwdep device directory (e.g.
/sys/class/sound/hwC0D0):
vendor_id
@@ -433,7 +433,7 @@ re-configure based on that state, run like below:
::
# echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs
- # echo 1 > /sys/class/sound/hwC0D0/reconfig
+ # echo 1 > /sys/class/sound/hwC0D0/reconfig
Hint Strings
@@ -494,7 +494,7 @@ indep_hp (bool)
mixer control, if available
add_stereo_mix_input (bool)
add the stereo mix (analog-loopback mix) to the input mux if
- available
+ available
add_jack_modes (bool)
add "xxx Jack Mode" enum controls to each I/O jack for allowing to
change the headphone amp and mic bias VREF capabilities
@@ -504,7 +504,7 @@ power_save_node (bool)
stream states
power_down_unused (bool)
power down the unused widgets, a subset of power_save_node, and
- will be dropped in future
+ will be dropped in future
add_hp_mic (bool)
add the headphone to capture source if possible
hp_mic_detect (bool)
@@ -603,7 +603,7 @@ present.
The patch module option is specific to each card instance, and you
need to give one file name for each instance, separated by commas.
-For example, if you have two cards, one for an on-board analog and one
+For example, if you have two cards, one for an on-board analog and one
for an HDMI video board, you may pass patch option like below:
::
diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h
index ea9af2726a53..7fba7ea26a4b 100644
--- a/include/sound/tas2781-dsp.h
+++ b/include/sound/tas2781-dsp.h
@@ -2,7 +2,7 @@
//
// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
//
-// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
+// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
// https://www.ti.com
//
// The TAS2781 driver implements a flexible and configurable
@@ -13,8 +13,8 @@
// Author: Kevin Lu <kevin-lu@ti.com>
//
-#ifndef __TASDEVICE_DSP_H__
-#define __TASDEVICE_DSP_H__
+#ifndef __TAS2781_DSP_H__
+#define __TAS2781_DSP_H__
#define MAIN_ALL_DEVICES 0x0d
#define MAIN_DEVICE_A 0x01
@@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context);
int tasdevice_select_tuningprm_cfg(void *context, int prm,
int cfg_no, int rca_conf_no);
int tasdevice_prmg_load(void *context, int prm_no);
-int tasdevice_prmg_calibdata_load(void *context, int prm_no);
void tasdevice_tuning_switch(void *context, int state);
int tas2781_load_calibration(void *context, char *file_name,
unsigned short i);
diff --git a/sound/core/init.c b/sound/core/init.c
index 6b127864a1a3..4e52bbe32786 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -50,7 +50,7 @@ MODULE_PARM_DESC(slots, "Module names assigned to the slots.");
static int module_slot_match(struct module *module, int idx)
{
int match = 1;
-#ifdef MODULE
+#ifdef CONFIG_MODULES
const char *s1, *s2;
if (!module || !*module->name || !slots[idx])
@@ -77,7 +77,7 @@ static int module_slot_match(struct module *module, int idx)
if (!c1)
break;
}
-#endif /* MODULE */
+#endif /* CONFIG_MODULES */
return match;
}
@@ -311,10 +311,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent,
}
card->dev = parent;
card->number = idx;
-#ifdef MODULE
- WARN_ON(!module);
+ WARN_ON(IS_MODULE(CONFIG_SND) && !module);
card->module = module;
-#endif
INIT_LIST_HEAD(&card->devices);
init_rwsem(&card->controls_rwsem);
rwlock_init(&card->ctl_files_rwlock);
@@ -969,7 +967,7 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer)
#endif
-#ifdef MODULE
+#ifdef CONFIG_MODULES
static void snd_card_module_info_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
@@ -997,7 +995,7 @@ int __init snd_card_info_init(void)
if (snd_info_register(entry) < 0)
return -ENOMEM; /* freed in error path */
-#ifdef MODULE
+#ifdef CONFIG_MODULES
entry = snd_info_create_module_entry(THIS_MODULE, "modules", NULL);
if (!entry)
return -ENOMEM;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 4d2ee99c12a3..d104adc75a8b 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -544,6 +544,14 @@ static int snd_timer_start1(struct snd_timer_instance *timeri,
SNDRV_TIMER_IFLG_START))
return -EBUSY;
+ /* check the actual time for the start tick;
+ * bail out as error if it's way too low (< 100us)
+ */
+ if (start) {
+ if ((u64)snd_timer_hw_resolution(timer) * ticks < 100000)
+ return -EINVAL;
+ }
+
if (start)
timeri->ticks = timeri->cticks = ticks;
else if (!timeri->cticks)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a696943aec0d..e3c0b9d5552d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10194,8 +10194,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8c89, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c8a, "HP EliteBook 630", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c8c, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8c8d, "HP ProBook 440 G11", ALC236_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8c8e, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c90, "HP EliteBook 640", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c91, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
@@ -12028,6 +12031,7 @@ enum {
ALC897_FIXUP_LENOVO_HEADSET_MODE,
ALC897_FIXUP_HEADSET_MIC_PIN2,
ALC897_FIXUP_UNIS_H3C_X500S,
+ ALC897_FIXUP_HEADSET_MIC_PIN3,
};
static const struct hda_fixup alc662_fixups[] = {
@@ -12474,10 +12478,18 @@ static const struct hda_fixup alc662_fixups[] = {
{}
},
},
+ [ALC897_FIXUP_HEADSET_MIC_PIN3] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a11050 }, /* use as headset mic */
+ { }
+ },
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1019, 0x9859, "JP-IK LEAP W502", ALC897_FIXUP_HEADSET_MIC_PIN3),
SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 833ea52638ab..03b539ba540f 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -829,8 +829,8 @@ static void es8326_jack_detect_handler(struct work_struct *work)
/* mute adc when mic path switch */
regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44);
regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
- es8326->hp = 0;
}
+ es8326->hp = 0;
regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a);
regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03);
@@ -981,7 +981,7 @@ static int es8326_resume(struct snd_soc_component *component)
regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
usleep_range(10000, 15000);
regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);
- regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb);
+ regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xd8);
/* set headphone default type and detect pin */
regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83);
regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05);
@@ -1018,7 +1018,7 @@ static int es8326_resume(struct snd_soc_component *component)
regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F);
/* select vdda as micbias source */
- regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x23);
+ regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x03);
/* set dac dsmclip = 1 */
regmap_write(es8326->regmap, ES8326_DAC_DSM, 0x08);
regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 05f574bf8b8f..cdb7ff7020e9 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -81,6 +81,7 @@ static const struct reg_sequence init_list[] = {
static const struct reg_sequence rt5650_init_list[] = {
{0xf6, 0x0100},
{RT5645_PWR_ANLG1, 0x02},
+ {RT5645_IL_CMD3, 0x0018},
};
static const struct reg_default rt5645_reg[] = {
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 40f5f27e74c0..a7ed59ec49a6 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -2,7 +2,8 @@
/*
* tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
*
- * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
+ * Copyright (C) 2014 - 2024 Texas Instruments Incorporated -
+ * https://www.ti.com
*
* Author: Dan Murphy <dmurphy@ti.com>
*/
@@ -119,12 +120,14 @@ static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] =
&tas2552_input_mux_control),
SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("ASI OUT", "DAC Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0),
SND_SOC_DAPM_POST("Post Event", tas2552_post_event),
- SND_SOC_DAPM_OUTPUT("OUT")
+ SND_SOC_DAPM_OUTPUT("OUT"),
+ SND_SOC_DAPM_INPUT("DMIC")
};
static const struct snd_soc_dapm_route tas2552_audio_map[] = {
@@ -134,6 +137,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = {
{"ClassD", NULL, "Input selection"},
{"OUT", NULL, "ClassD"},
{"ClassD", NULL, "PLL"},
+ {"ASI OUT", NULL, "DMIC"}
};
#ifdef CONFIG_PM
@@ -538,6 +542,13 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = TAS2552_FORMATS,
},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = TAS2552_FORMATS,
+ },
.ops = &tas2552_speaker_dai_ops,
},
};
diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c
index a6be81adcb83..265a8ca25cbb 100644
--- a/sound/soc/codecs/tas2781-fmwlib.c
+++ b/sound/soc/codecs/tas2781-fmwlib.c
@@ -2151,6 +2151,24 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv,
return ret;
}
+static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i)
+{
+ struct tasdevice_calibration *cal;
+ struct tasdevice_fw *cal_fmw;
+
+ cal_fmw = priv->tasdevice[i].cali_data_fmw;
+
+ /* No calibrated data for current devices, playback will go ahead. */
+ if (!cal_fmw)
+ return;
+
+ cal = cal_fmw->calibrations;
+ if (cal)
+ return;
+
+ load_calib_data(priv, &cal->dev_data);
+}
+
int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
int cfg_no, int rca_conf_no)
{
@@ -2210,21 +2228,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
for (i = 0; i < tas_priv->ndev; i++) {
if (tas_priv->tasdevice[i].is_loaderr == true)
continue;
- else if (tas_priv->tasdevice[i].is_loaderr == false
- && tas_priv->tasdevice[i].is_loading == true) {
- struct tasdevice_fw *cal_fmw =
- tas_priv->tasdevice[i].cali_data_fmw;
-
- if (cal_fmw) {
- struct tasdevice_calibration
- *cal = cal_fmw->calibrations;
-
- if (cal)
- load_calib_data(tas_priv,
- &(cal->dev_data));
- }
+ if (tas_priv->tasdevice[i].is_loaderr == false &&
+ tas_priv->tasdevice[i].is_loading == true)
tas_priv->tasdevice[i].cur_prog = prm_no;
- }
}
}
@@ -2245,11 +2251,15 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
tasdevice_load_data(tas_priv, &(conf->dev_data));
for (i = 0; i < tas_priv->ndev; i++) {
if (tas_priv->tasdevice[i].is_loaderr == true) {
- status |= 1 << (i + 4);
+ status |= BIT(i + 4);
continue;
- } else if (tas_priv->tasdevice[i].is_loaderr == false
- && tas_priv->tasdevice[i].is_loading == true)
+ }
+
+ if (tas_priv->tasdevice[i].is_loaderr == false &&
+ tas_priv->tasdevice[i].is_loading == true) {
+ tasdev_load_calibrated_data(tas_priv, i);
tas_priv->tasdevice[i].cur_conf = cfg_no;
+ }
}
} else
dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n",
@@ -2308,65 +2318,6 @@ out:
}
EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB);
-int tasdevice_prmg_calibdata_load(void *context, int prm_no)
-{
- struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context;
- struct tasdevice_fw *tas_fmw = tas_priv->fmw;
- struct tasdevice_prog *program;
- int prog_status = 0;
- int i;
-
- if (!tas_fmw) {
- dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__);
- goto out;
- }
-
- if (prm_no >= tas_fmw->nr_programs) {
- dev_err(tas_priv->dev,
- "%s: prm(%d) is not in range of Programs %u\n",
- __func__, prm_no, tas_fmw->nr_programs);
- goto out;
- }
-
- for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) {
- if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) {
- tas_priv->tasdevice[i].cur_conf = -1;
- tas_priv->tasdevice[i].is_loading = true;
- prog_status++;
- }
- tas_priv->tasdevice[i].is_loaderr = false;
- }
-
- if (prog_status) {
- program = &(tas_fmw->programs[prm_no]);
- tasdevice_load_data(tas_priv, &(program->dev_data));
- for (i = 0; i < tas_priv->ndev; i++) {
- if (tas_priv->tasdevice[i].is_loaderr == true)
- continue;
- else if (tas_priv->tasdevice[i].is_loaderr == false
- && tas_priv->tasdevice[i].is_loading == true) {
- struct tasdevice_fw *cal_fmw =
- tas_priv->tasdevice[i].cali_data_fmw;
-
- if (cal_fmw) {
- struct tasdevice_calibration *cal =
- cal_fmw->calibrations;
-
- if (cal)
- load_calib_data(tas_priv,
- &(cal->dev_data));
- }
- tas_priv->tasdevice[i].cur_prog = prm_no;
- }
- }
- }
-
-out:
- return prog_status;
-}
-EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load,
- SND_SOC_TAS2781_FMWLIB);
-
void tasdevice_tuning_switch(void *context, int state)
{
struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context;
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index b5abff230e43..9350972dfefe 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -2,7 +2,7 @@
//
// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
//
-// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
+// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
// https://www.ti.com
//
// The TAS2563/TAS2781 driver implements a flexible and configurable
@@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw,
__func__, tas_priv->cal_binaryname[i]);
}
- tasdevice_prmg_calibdata_load(tas_priv, 0);
+ tasdevice_prmg_load(tas_priv, 0);
tas_priv->cur_prog = 0;
out:
if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) {
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 701b0372f59e..012195c50519 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -109,7 +109,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *d
return -ENOMEM;
for (i = 0; i < ARRAY_SIZE(need_sdca_suffix); i++) {
- if (strstr(codec_dai->name, need_sdca_suffix[i])) {
+ if (strstr(component->name_prefix, need_sdca_suffix[i])) {
/* Add -sdca suffix for existing UCMs */
card->components = devm_kasprintf(card->dev, GFP_KERNEL,
"%s-sdca", card->components);
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 2f961f0e9378..58156fbca02c 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1740,7 +1740,7 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
u32 current_rate;
// Get current rate from card and check if changing it is needed
- snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+ snd_usb_ctl_msg(subs->dev, usb_rcvctrlpipe(subs->dev, 0),
0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
current_rate = le32_to_cpu(buff4);
dev_dbg(&subs->dev->dev,
@@ -1765,7 +1765,7 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
// Check whether the change was successful
buff4 = 0;
- snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+ snd_usb_ctl_msg(subs->dev, usb_rcvctrlpipe(subs->dev, 0),
0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
if (new_rate != le32_to_cpu(buff4))
dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate");
generated by cgit v1.2.3-70-g09d2 (git 2.47.1) at 2024-12-28 05:24:34 +0000