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efc09808c6ce98a243b02796f65dbcafd67f97c6
omarchy-pkgs/pkgbuilds/linux-omarchy-eevdf/0523-asoc-fixes-2.patch
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diff --git a/sound/hda/core/ext/controller.c b/sound/hda/core/ext/controller.c
--- a/sound/hda/core/ext/controller.c
+++ b/sound/hda/core/ext/controller.c
@@ -90,8 +90,10 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus)
for (idx = 0; idx < link_count; idx++) {
hlink = kzalloc_obj(*hlink);
- if (!hlink)
+ if (!hlink) {
+ snd_hdac_ext_link_free_all(bus);
return -ENOMEM;
+ }
hlink->index = idx;
hlink->bus = bus;
hlink->ml_addr = bus->mlcap + AZX_ML_BASE +
diff --git a/sound/hda/core/ext/stream.c b/sound/hda/core/ext/stream.c
--- a/sound/hda/core/ext/stream.c
+++ b/sound/hda/core/ext/stream.c
@@ -102,8 +102,10 @@ int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx,
for (i = 0; i < num_stream; i++) {
struct hdac_ext_stream *hext_stream = kzalloc_obj(*hext_stream);
- if (!hext_stream)
+ if (!hext_stream) {
+ snd_hdac_ext_stream_free_all(bus);
return -ENOMEM;
+ }
tag = ++stream_tag;
snd_hdac_ext_stream_init(bus, hext_stream, idx, dir, tag);
idx++;
@@ -111,7 +113,6 @@ int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx,
}
return 0;
-
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all);
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -388,6 +388,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Nitro ANV15-41"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "RB"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Nitro AN17-41"),
+ }
+ },
{
.driver_data = &acp6x_card,
.matches = {
@@ -899,6 +906,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_BOARD_NAME, "TM2423"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "Hewlett-Packard"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "HP 255R 15.6 inch G10 Notebook PC"),
+ }
+ },
{}
};
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -14,17 +14,14 @@
* for ST-Ericsson.
*/
-#include <linux/cleanup.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
-#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/platform_device.h>
-#include <linux/mutex.h>
#include <linux/mfd/abx500/ab8500.h>
#include <linux/mfd/abx500.h>
#include <linux/mfd/abx500/ab8500-sysctrl.h>
@@ -54,32 +51,9 @@
/* Macrocell register definitions */
#define AB8500_GPIO_DIR4_REG 0x13 /* Bank AB8500_MISC */
-/* Nr of FIR/IIR-coeff banks in ANC-block */
-#define AB8500_NR_OF_ANC_COEFF_BANKS 2
-
-/* Minimum duration to keep ANC IIR Init bit high or
-low before proceeding with the configuration sequence */
-#define AB8500_ANC_SM_DELAY 2000
-
-/* Sidetone states */
-static const char * const enum_sid_state[] = {
- "Unconfigured",
- "Apply FIR",
- "FIR is configured",
-};
-enum sid_state {
- SID_UNCONFIGURED = 0,
- SID_APPLY_FIR = 1,
- SID_FIR_CONFIGURED = 2,
-};
-
/* Private data for AB8500 device-driver */
struct ab8500_codec_drvdata {
struct regmap *regmap;
- struct mutex ctrl_lock;
-
- /* Sidetone */
- enum sid_state sid_status;
};
static inline const char *amic_micbias_str(enum amic_micbias micbias)
@@ -642,9 +616,6 @@ static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = {
NULL, 0),
/* Acoustical Noise Cancellation path */
- SND_SOC_DAPM_INPUT("ANC Configure Input"),
- SND_SOC_DAPM_OUTPUT("ANC Configure Output"),
-
SND_SOC_DAPM_MUX("ANC Source",
SND_SOC_NOPM, 0, 0,
dapm_anc_in_select),
@@ -687,10 +658,6 @@ static const struct snd_soc_dapm_route ab8500_dapm_routes[] = {
{"Main Supply", NULL, "Audio Power"},
{"Main Supply", NULL, "Audio Analog Power"},
- /* ANC Configure */
- {"ANC Configure Input", NULL, "Main Supply"},
- {"ANC Configure Output", NULL, "ANC Configure Input"},
-
/* Powerup charge pump if DA1/2 is in use */
{"DA_IN1", NULL, "ab8500_0p"},
@@ -972,75 +939,6 @@ static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = {
{"MIC2 V-AMICx Enable", NULL, "V-AMIC2"},
};
-/*
- * Control-events
- */
-
-static int sid_status_control_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
- struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(component->dev);
-
- guard(mutex)(&drvdata->ctrl_lock);
- ucontrol->value.enumerated.item[0] = drvdata->sid_status;
-
- return 0;
-}
-
-/* Write sidetone FIR-coefficients configuration sequence */
-static int sid_status_control_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
- struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(component->dev);
- unsigned int param, sidconf;
- int status = 1;
-
- dev_dbg(component->dev, "%s: Enter\n", __func__);
-
- if (ucontrol->value.enumerated.item[0] != SID_APPLY_FIR) {
- dev_err(component->dev,
- "%s: ERROR: This control supports '%s' only!\n",
- __func__, enum_sid_state[SID_APPLY_FIR]);
- return -EIO;
- }
-
- guard(mutex)(&drvdata->ctrl_lock);
-
- sidconf = snd_soc_component_read(component, AB8500_SIDFIRCONF);
- if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) {
- if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) {
- dev_err(component->dev, "%s: Sidetone busy while off!\n",
- __func__);
- status = -EPERM;
- } else {
- status = -EBUSY;
- }
- dev_dbg(component->dev, "%s: Exit\n", __func__);
- return status;
- }
-
- snd_soc_component_write(component, AB8500_SIDFIRADR, 0);
-
- for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) {
- snd_soc_component_write(component, AB8500_SIDFIRCOEF1, 0);
- snd_soc_component_write(component, AB8500_SIDFIRCOEF2, 0);
- }
-
- snd_soc_component_update_bits(component, AB8500_SIDFIRADR,
- BIT(AB8500_SIDFIRADR_FIRSIDSET),
- BIT(AB8500_SIDFIRADR_FIRSIDSET));
- snd_soc_component_update_bits(component, AB8500_SIDFIRADR,
- BIT(AB8500_SIDFIRADR_FIRSIDSET), 0);
-
- drvdata->sid_status = SID_FIR_CONFIGURED;
-
- dev_dbg(component->dev, "%s: Exit\n", __func__);
-
- return status;
-}
-
/*
* Controls - Non-DAPM ASoC
*/
@@ -1324,9 +1222,6 @@ static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast,
AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT,
enum_slavemaster);
-/* Sidetone */
-static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state);
-
/* ANC */
static struct snd_kcontrol_new ab8500_ctrls[] = {
@@ -1617,8 +1512,6 @@ static struct snd_kcontrol_new ab8500_ctrls[] = {
AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0),
/* Sidetone */
- SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate,
- sid_status_control_get, sid_status_control_put),
SOC_SINGLE_STROBE("Sidetone Reset",
AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0),
};
@@ -2148,10 +2041,8 @@ static void ab8500_codec_of_probe(struct device *dev, struct device_node *np,
static int ab8500_codec_probe(struct snd_soc_component *component)
{
- struct snd_soc_dapm_context *dapm = snd_soc_component_to_dapm(component);
struct device *dev = component->dev;
struct device_node *np = dev->of_node;
- struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
struct ab8500_codec_platform_data codec_pdata;
int status;
@@ -2184,10 +2075,6 @@ static int ab8500_codec_probe(struct snd_soc_component *component)
snd_soc_component_write(component, AB8500_SHORTCIRCONF,
BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
- snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
-
- mutex_init(&drvdata->ctrl_lock);
-
return status;
}
@@ -2216,7 +2103,6 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev)
GFP_KERNEL);
if (!drvdata)
return -ENOMEM;
- drvdata->sid_status = SID_UNCONFIGURED;
dev_set_drvdata(&pdev->dev, drvdata);
drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev,
diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c
--- a/sound/soc/codecs/cs35l56-sdw.c
+++ b/sound/soc/codecs/cs35l56-sdw.c
@@ -386,11 +386,8 @@ static int __maybe_unused cs35l56_sdw_system_suspend(struct device *dev)
{
struct cs35l56_private *cs35l56 = dev_get_drvdata(dev);
- if (!cs35l56->base.init_done)
- return 0;
-
- /* runtime_resume unmasks the interrupt */
- cs35l56_mask_soundwire_interrupts(cs35l56);
+ if (cs35l56->sdw_attached)
+ cs35l56_mask_soundwire_interrupts(cs35l56);
return cs35l56_system_suspend(dev);
}
diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c
--- a/sound/soc/codecs/cs35l56.c
+++ b/sound/soc/codecs/cs35l56.c
@@ -18,9 +18,11 @@
#include <linux/interrupt.h>
#include <linux/math.h>
#include <linux/module.h>
+#include <linux/mutex.h>
#include <linux/pm.h>
#include <linux/pm_runtime.h>
#include <linux/property.h>
+#include <linux/reboot.h>
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
@@ -37,6 +39,13 @@
#include "wm_adsp.h"
#include "cs35l56.h"
+/*
+ * snd_soc_register_component() can call component_probe() on all instances
+ * in a card, so deferred registration must be protected across all instances.
+ */
+static DEFINE_MUTEX(cs35l56_component_register_lock);
+static bool cs35l56_shutting_down;
+
void cs35l56_mask_soundwire_interrupts(struct cs35l56_private *cs35l56)
{
/*
@@ -1957,6 +1966,11 @@ static void cs35l56_component_register_work(struct work_struct *work)
component_register_work);
int ret;
+ guard(mutex)(&cs35l56_component_register_lock);
+
+ if (cs35l56_shutting_down)
+ return;
+
PM_RUNTIME_ACQUIRE_AUTOSUSPEND(cs35l56->base.dev, pm_err);
ret = PM_RUNTIME_ACQUIRE_ERR(&pm_err);
if (ret) {
@@ -2217,6 +2231,37 @@ EXPORT_NS_GPL_DEV_PM_OPS(cs35l56_pm_ops_i2c_spi, SND_SOC_CS35L56_CORE) = {
};
#endif
+static int cs35l56_reboot_notify(struct notifier_block *nb,
+ unsigned long action, void *data)
+{
+ guard(mutex)(&cs35l56_component_register_lock);
+ cs35l56_shutting_down = true;
+
+ return NOTIFY_DONE;
+}
+
+static struct notifier_block cs35l56_reboot_notifier = {
+ .notifier_call = cs35l56_reboot_notify,
+};
+
+static int __init cs35l56_modinit(void)
+{
+ /*
+ * Use reboot notifier to prevent race between shutdown and
+ * snd_soc_register_component(). Driver shutdown() callback would
+ * run too late, after device_shutdown() is already walking the
+ * device list that component registration can modify.
+ */
+ return register_reboot_notifier(&cs35l56_reboot_notifier);
+}
+module_init(cs35l56_modinit);
+
+static void __exit cs35l56_modexit(void)
+{
+ unregister_reboot_notifier(&cs35l56_reboot_notifier);
+}
+module_exit(cs35l56_modexit);
+
MODULE_DESCRIPTION("ASoC CS35L56 driver");
MODULE_IMPORT_NS("SND_SOC_CS35L56_SHARED");
MODULE_IMPORT_NS("SND_SOC_CS_AMP_LIB");
diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h
--- a/sound/soc/codecs/cs35l56.h
+++ b/sound/soc/codecs/cs35l56.h
@@ -32,6 +32,7 @@ struct sdw_slave;
struct cs35l56_private {
struct wm_adsp dsp; /* must be first member */
struct cs35l56_base base;
+ struct work_struct component_register_work;
struct work_struct dsp_work;
struct workqueue_struct *dsp_wq;
struct snd_soc_component *component;
@@ -41,6 +42,7 @@ struct cs35l56_private {
const char *fallback_fw_suffix;
bool soft_resetting;
bool sdw_attached;
+ bool component_registered;
struct completion init_completion;
int speaker_id;
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -26,6 +26,7 @@ struct es8326_priv {
struct snd_soc_component *component;
struct delayed_work jack_detect_work;
struct delayed_work button_press_work;
+ struct delayed_work capture_pop_work;
struct snd_soc_jack *jack;
int irq;
/* The lock protects the situation that an irq is generated
@@ -628,6 +629,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF,
0x30, 0x00);
} else {
+ cancel_delayed_work_sync(&es8326->capture_pop_work);
regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE,
0x0F, 0x0F);
if (es8326->version > ES8326_VERSION_B) {
@@ -666,8 +668,9 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70);
regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00);
}
- regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE,
- 0x0F, 0x00);
+
+ queue_delayed_work(system_dfl_wq, &es8326->capture_pop_work,
+ msecs_to_jiffies(40));
}
}
return 0;
@@ -773,6 +776,15 @@ static void es8326_disable_micbias(struct snd_soc_component *component)
snd_soc_dapm_mutex_unlock(dapm);
}
+static void es8326_capture_pop_handler(struct work_struct *work)
+{
+ struct es8326_priv *es8326 =
+ container_of(work, struct es8326_priv, capture_pop_work.work);
+
+ regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE,
+ 0x0F, 0x00);
+}
+
/*
* For button detection, set the following in soundcard
* snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
@@ -1140,6 +1152,7 @@ static int es8326_suspend(struct snd_soc_component *component)
struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component);
cancel_delayed_work_sync(&es8326->jack_detect_work);
+ cancel_delayed_work_sync(&es8326->capture_pop_work);
es8326_disable_micbias(component);
es8326->calibrated = false;
regmap_write(es8326->regmap, ES8326_CLK_MUX, 0x2d);
@@ -1291,6 +1304,8 @@ static int es8326_i2c_probe(struct i2c_client *i2c)
es8326_jack_detect_handler);
INIT_DELAYED_WORK(&es8326->button_press_work,
es8326_jack_button_handler);
+ INIT_DELAYED_WORK(&es8326->capture_pop_work,
+ es8326_capture_pop_handler);
/* ES8316 is level-based while ES8326 is edge-based */
ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq,
IRQF_TRIGGER_RISING | IRQF_ONESHOT,
diff --git a/sound/soc/codecs/es8389.c b/sound/soc/codecs/es8389.c
--- a/sound/soc/codecs/es8389.c
+++ b/sound/soc/codecs/es8389.c
@@ -812,6 +812,23 @@ static int es8389_pcm_hw_free(struct snd_pcm_substream *substream,
return 0;
}
+static void es8389_standby(struct snd_soc_component *component)
+{
+ struct es8389_private *es8389 = snd_soc_component_get_drvdata(component);
+
+ regmap_update_bits(es8389->regmap, ES8389_ADC_HPF1, 0x0f, 0x04);
+ regmap_update_bits(es8389->regmap, ES8389_ADC_HPF2, 0x0f, 0x04);
+ regmap_write(es8389->regmap, ES8389_CSM_JUMP, 0xD4);
+ usleep_range(70000, 72000);
+ 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, 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);
+}
+
static int es8389_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
@@ -834,18 +851,7 @@ static int es8389_set_bias_level(struct snd_soc_component *component,
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- regmap_update_bits(es8389->regmap, ES8389_ADC_HPF1, 0x0f, 0x04);
- regmap_update_bits(es8389->regmap, ES8389_ADC_HPF2, 0x0f, 0x04);
- regmap_write(es8389->regmap, ES8389_CSM_JUMP, 0xD4);
- usleep_range(70000, 72000);
- 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, 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);
-
+ es8389_standby(component);
clk_disable_unprepare(es8389->mclk);
break;
case SND_SOC_BIAS_OFF:
@@ -1015,7 +1021,7 @@ static int es8389_suspend(struct snd_soc_component *component)
{
struct es8389_private *es8389 = snd_soc_component_get_drvdata(component);
- es8389_set_bias_level(component, SND_SOC_BIAS_STANDBY);
+ es8389_standby(component);
regcache_cache_only(es8389->regmap, true);
regcache_mark_dirty(es8389->regmap);
@@ -1084,7 +1090,7 @@ static int es8389_probe(struct snd_soc_component *component)
es8389->hpf_freq = ES8389_HPF_DEFAULT;
es8389_init(component);
- es8389_set_bias_level(component, SND_SOC_BIAS_STANDBY);
+ es8389_standby(component);
return 0;
}
diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c
--- a/sound/soc/codecs/rt1318-sdw.c
+++ b/sound/soc/codecs/rt1318-sdw.c
@@ -16,6 +16,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc-dapm.h>
+#include <sound/sdw.h>
#include <sound/initval.h>
#include "rt1318-sdw.h"
@@ -564,11 +565,10 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct rt1318_sdw_priv *rt1318 =
snd_soc_component_get_drvdata(component);
- struct sdw_stream_config stream_config;
+ struct sdw_stream_config stream_config = {0};
struct sdw_port_config port_config;
- enum sdw_data_direction direction;
struct sdw_stream_runtime *sdw_stream;
- int retval, port, num_channels, ch_mask;
+ int retval, port;
unsigned int sampling_rate;
dev_dbg(dai->dev, "%s %s", __func__, dai->name);
@@ -582,23 +582,13 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
/* SoundWire specific configuration */
/* port 1 for playback */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- direction = SDW_DATA_DIR_RX;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
port = 1;
- } else {
- direction = SDW_DATA_DIR_TX;
+ else
port = 2;
- }
- num_channels = params_channels(params);
- ch_mask = (1 << num_channels) - 1;
-
- stream_config.frame_rate = params_rate(params);
- stream_config.ch_count = num_channels;
- stream_config.bps = snd_pcm_format_width(params_format(params));
- stream_config.direction = direction;
-
- port_config.ch_mask = ch_mask;
+ /* SoundWire specific configuration */
+ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
port_config.num = port;
retval = sdw_stream_add_slave(rt1318->sdw_slave, &stream_config,
diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c
--- a/sound/soc/codecs/rt1320-sdw.c
+++ b/sound/soc/codecs/rt1320-sdw.c
@@ -3246,7 +3246,7 @@ static int rt1320_sdw_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct rt1320_sdw_priv *rt1320 =
snd_soc_component_get_drvdata(component);
- struct sdw_stream_config stream_config;
+ struct sdw_stream_config stream_config = {0};
struct sdw_port_config port_config;
struct sdw_port_config dmic_port_config[2];
struct sdw_stream_runtime *sdw_stream;
diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c
--- a/sound/soc/codecs/rt712-sdca.c
+++ b/sound/soc/codecs/rt712-sdca.c
@@ -23,6 +23,7 @@
#include <linux/soundwire/sdw_registers.h>
#include <linux/slab.h>
#include <sound/soc-dapm.h>
+#include <sound/sdw.h>
#include <sound/tlv.h>
#include "rt712-sdca.h"
@@ -1449,11 +1450,10 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_component *component = dai->component;
struct rt712_sdca_priv *rt712 = snd_soc_component_get_drvdata(component);
- struct sdw_stream_config stream_config;
+ struct sdw_stream_config stream_config = {0};
struct sdw_port_config port_config;
- enum sdw_data_direction direction;
struct sdw_stream_runtime *sdw_stream;
- int retval, port, num_channels;
+ int retval, port;
unsigned int sampling_rate;
dev_dbg(dai->dev, "%s %s id %d", __func__, dai->name, dai->id);
@@ -1471,7 +1471,6 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
/* SoundWire specific configuration */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- direction = SDW_DATA_DIR_RX;
if (dai->id == RT712_AIF1)
port = 1;
else if (dai->id == RT712_AIF2)
@@ -1479,7 +1478,6 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
else
return -EINVAL;
} else {
- direction = SDW_DATA_DIR_TX;
if (dai->id == RT712_AIF1)
port = 4;
else if (dai->id == RT712_AIF3)
@@ -1488,13 +1486,8 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- stream_config.frame_rate = params_rate(params);
- stream_config.ch_count = params_channels(params);
- stream_config.bps = snd_pcm_format_width(params_format(params));
- stream_config.direction = direction;
-
- num_channels = params_channels(params);
- port_config.ch_mask = GENMASK(num_channels - 1, 0);
+ /* SoundWire specific configuration */
+ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
port_config.num = port;
retval = sdw_stream_add_slave(rt712->slave, &stream_config,
diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c
--- a/sound/soc/codecs/rt721-sdca.c
+++ b/sound/soc/codecs/rt721-sdca.c
@@ -21,6 +21,7 @@
#include <linux/soundwire/sdw_registers.h>
#include <linux/slab.h>
#include <sound/soc-dapm.h>
+#include <sound/sdw.h>
#include <sound/tlv.h>
#include "rt721-sdca.h"
@@ -206,6 +207,7 @@ static void rt721_sdca_amp_preset(struct rt721_sdca_priv *rt721)
regmap_write(rt721->regmap,
SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55,
RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00);
+ regmap_write(rt721->regmap, 0x2f5d, 0x1);
}
static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721)
@@ -1268,11 +1270,10 @@ static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_component *component = dai->component;
struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component);
- struct sdw_stream_config stream_config;
+ struct sdw_stream_config stream_config = {0};
struct sdw_port_config port_config;
- enum sdw_data_direction direction;
struct sdw_stream_runtime *sdw_stream;
- int retval, port, num_channels;
+ int retval, port;
unsigned int sampling_rate;
dev_dbg(dai->dev, "%s %s", __func__, dai->name);
@@ -1291,7 +1292,6 @@ static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
* RT721_AIF3 with port = 6 for digital-mic capture
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- direction = SDW_DATA_DIR_RX;
if (dai->id == RT721_AIF1)
port = 1;
else if (dai->id == RT721_AIF2)
@@ -1299,7 +1299,6 @@ static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
else
return -EINVAL;
} else {
- direction = SDW_DATA_DIR_TX;
if (dai->id == RT721_AIF1)
port = 2;
else if (dai->id == RT721_AIF3)
@@ -1307,13 +1306,9 @@ static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
else
return -EINVAL;
}
- stream_config.frame_rate = params_rate(params);
- stream_config.ch_count = params_channels(params);
- stream_config.bps = snd_pcm_format_width(params_format(params));
- stream_config.direction = direction;
- num_channels = params_channels(params);
- port_config.ch_mask = GENMASK(num_channels - 1, 0);
+ /* SoundWire specific configuration */
+ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
port_config.num = port;
retval = sdw_stream_add_slave(rt721->slave, &stream_config,
diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c
--- a/sound/soc/codecs/rt722-sdca.c
+++ b/sound/soc/codecs/rt722-sdca.c
@@ -24,6 +24,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc-dapm.h>
+#include <sound/sdw.h>
#include <sound/tlv.h>
#include "rt722-sdca.h"
@@ -1442,11 +1443,10 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_component *component = dai->component;
struct rt722_sdca_priv *rt722 = snd_soc_component_get_drvdata(component);
- struct sdw_stream_config stream_config;
+ struct sdw_stream_config stream_config = {0};
struct sdw_port_config port_config;
- enum sdw_data_direction direction;
struct sdw_stream_runtime *sdw_stream;
- int retval, port, num_channels;
+ int retval, port;
unsigned int sampling_rate;
dev_dbg(dai->dev, "%s %s", __func__, dai->name);
@@ -1465,7 +1465,6 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
* RT722_AIF3 with port = 6 for digital-mic capture
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- direction = SDW_DATA_DIR_RX;
if (dai->id == RT722_AIF1)
port = 1;
else if (dai->id == RT722_AIF2)
@@ -1473,7 +1472,6 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
else
return -EINVAL;
} else {
- direction = SDW_DATA_DIR_TX;
if (dai->id == RT722_AIF1)
port = 2;
else if (dai->id == RT722_AIF3)
@@ -1481,13 +1479,9 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
else
return -EINVAL;
}
- stream_config.frame_rate = params_rate(params);
- stream_config.ch_count = params_channels(params);
- stream_config.bps = snd_pcm_format_width(params_format(params));
- stream_config.direction = direction;
- num_channels = params_channels(params);
- port_config.ch_mask = GENMASK(num_channels - 1, 0);
+ /* SoundWire specific configuration */
+ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
port_config.num = port;
retval = sdw_stream_add_slave(rt722->slave, &stream_config,
diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c
--- a/sound/soc/codecs/tas2783-sdw.c
+++ b/sound/soc/codecs/tas2783-sdw.c
@@ -1001,6 +1001,31 @@ static s32 tas_sdw_hw_params(struct snd_pcm_substream *substream,
/* SoundWire specific configuration */
snd_sdw_params_to_config(substream, params,
&stream_config, &port_config);
+
+ /*
+ * The two mono amps each render one channel of the stereo stream:
+ * snd_sdw_params_to_config() hands every codec the full mask for
+ * playback, which leaves the pair in mirror mode and one channel
+ * unreproduced. Claim a single channel instead, keyed off the
+ * machine-assigned component prefix rather than the SoundWire
+ * address, which is board-specific: soc_sdw_ti_amp.c names the amps
+ * tas2783-1..4.
+ *
+ * Which side an amp then renders does not follow from the bit that
+ * is set - sdw_compute_slave_ports() advances the payload offset by
+ * the popcount of ch_mask and never looks at which bit it is - but
+ * from the amp's position in the codec order of the DAI link, which
+ * on these boards matches the prefix numbering.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ params_channels(params) == 2 && component->name_prefix) {
+ const char *idx_str = strrchr(component->name_prefix, '-');
+ unsigned long idx;
+
+ if (idx_str && !kstrtoul(idx_str + 1, 10, &idx) && idx)
+ port_config.ch_mask = (idx & 1) ? BIT(0) : BIT(1);
+ }
+
/* port 1 for playback */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
port_config.num = 1;
diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c
--- a/sound/soc/intel/avs/boards/hdaudio.c
+++ b/sound/soc/intel/avs/boards/hdaudio.c
@@ -15,6 +15,22 @@
#include "../../../codecs/hda.h"
#include "../utils.h"
+static int avs_link_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ const struct snd_soc_pcm_stream *stream_info;
+ struct snd_soc_dai *codec_dai;
+
+ codec_dai = snd_soc_rtd_to_codec(rtd, 0);
+ stream_info = snd_soc_dai_get_pcm_stream(codec_dai, substream->stream);
+
+ return snd_pcm_hw_constraint_msbits(substream->runtime, 0, 0, stream_info->sig_bits);
+}
+
+static const struct snd_soc_ops avs_link_ops = {
+ .startup = avs_link_startup,
+};
+
static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int pcm_count,
struct snd_soc_dai_link **links)
{
@@ -43,6 +59,7 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int
dl[i].platforms = platform;
dl[i].num_platforms = 1;
dl[i].ignore_pmdown_time = 1;
+ dl[i].ops = &avs_link_ops;
dl[i].codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL);
dl[i].cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL);
diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -92,16 +92,28 @@ static int avs_hdac_bus_init_streams(struct hdac_bus *bus)
{
unsigned int cp_streams, pb_streams;
unsigned int gcap;
+ int ret;
gcap = snd_hdac_chip_readw(bus, GCAP);
cp_streams = (gcap >> 8) & 0x0F;
pb_streams = (gcap >> 12) & 0x0F;
bus->num_streams = cp_streams + pb_streams;
- snd_hdac_ext_stream_init_all(bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE);
- snd_hdac_ext_stream_init_all(bus, cp_streams, pb_streams, SNDRV_PCM_STREAM_PLAYBACK);
+ ret = snd_hdac_ext_stream_init_all(bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ return ret;
+ ret = snd_hdac_ext_stream_init_all(bus, cp_streams, pb_streams, SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto err;
- return snd_hdac_bus_alloc_stream_pages(bus);
+ ret = snd_hdac_bus_alloc_stream_pages(bus);
+ if (ret)
+ goto err;
+
+ return 0;
+err:
+ snd_hdac_ext_stream_free_all(bus);
+ return ret;
}
static bool avs_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
@@ -383,6 +395,18 @@ static int avs_bus_init(struct avs_dev *adev, struct pci_dev *pci, const struct
struct device *dev = &pci->dev;
int ret;
+ ipc = devm_kzalloc(dev, sizeof(*ipc), GFP_KERNEL);
+ if (!ipc)
+ return -ENOMEM;
+
+ adev->modcfg_buf = devm_kzalloc(dev, AVS_MAILBOX_SIZE, GFP_KERNEL);
+ if (!adev->modcfg_buf)
+ return -ENOMEM;
+
+ ret = avs_ipc_init(ipc, dev);
+ if (ret < 0)
+ return ret;
+
ret = snd_hdac_ext_bus_init(&bus->core, dev, NULL, &soc_hda_ext_bus_ops);
if (ret < 0)
return ret;
@@ -394,17 +418,6 @@ static int avs_bus_init(struct avs_dev *adev, struct pci_dev *pci, const struct
bus->mixer_assigned = -1;
mutex_init(&bus->prepare_mutex);
- ipc = devm_kzalloc(dev, sizeof(*ipc), GFP_KERNEL);
- if (!ipc)
- return -ENOMEM;
- ret = avs_ipc_init(ipc, dev);
- if (ret < 0)
- return ret;
-
- adev->modcfg_buf = devm_kzalloc(dev, AVS_MAILBOX_SIZE, GFP_KERNEL);
- if (!adev->modcfg_buf)
- return -ENOMEM;
-
adev->dev = dev;
adev->spec = (const struct avs_spec *)id->driver_data;
adev->ipc = ipc;
@@ -456,13 +469,14 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id)
ret = pcim_request_all_regions(pci, "AVS HDAudio");
if (ret < 0)
- return ret;
+ goto err_request_regions;
bus->addr = pci_resource_start(pci, 0);
bus->remap_addr = pci_ioremap_bar(pci, 0);
if (!bus->remap_addr) {
dev_err(bus->dev, "ioremap error\n");
- return -ENXIO;
+ ret = -ENXIO;
+ goto err_request_regions;
}
adev->dsp_ba = pci_ioremap_bar(pci, 4);
@@ -526,6 +540,8 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id)
iounmap(adev->dsp_ba);
err_remap_bar4:
iounmap(bus->remap_addr);
+err_request_regions:
+ snd_hdac_ext_bus_exit(bus);
return ret;
}
diff --git a/sound/soc/intel/avs/debugfs.c b/sound/soc/intel/avs/debugfs.c
--- a/sound/soc/intel/avs/debugfs.c
+++ b/sound/soc/intel/avs/debugfs.c
@@ -9,6 +9,7 @@
#include <linux/cleanup.h>
#include <linux/debugfs.h>
#include <linux/kfifo.h>
+#include <linux/module.h>
#include <linux/wait.h>
#include <linux/sched/signal.h>
#include <linux/string_helpers.h>
@@ -236,15 +237,20 @@ static int strace_open(struct inode *inode, struct file *file)
if (!try_module_get(adev->dev->driver->owner))
return -ENODEV;
- if (kfifo_initialized(&adev->trace_fifo))
- return -EBUSY;
+ if (kfifo_initialized(&adev->trace_fifo)) {
+ ret = -EBUSY;
+ goto err;
+ }
ret = kfifo_alloc(&adev->trace_fifo, PAGE_SIZE, GFP_KERNEL);
if (ret < 0)
- return ret;
+ goto err;
file->private_data = adev;
return 0;
+err:
+ module_put(adev->dev->driver->owner);
+ return ret;
}
static int strace_release(struct inode *inode, struct file *file)
diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c
--- a/sound/soc/intel/avs/ipc.c
+++ b/sound/soc/intel/avs/ipc.c
@@ -172,7 +172,7 @@ static void avs_dsp_exception_caught(struct avs_dev *adev, union avs_notify_msg
/* Avoid deadlock as the exception may be the response to SET_D0IX. */
if (current_work() != &ipc->d0ix_work.work)
- cancel_delayed_work_sync(&ipc->d0ix_work);
+ cancel_delayed_work(&ipc->d0ix_work);
ipc->in_d0ix = false;
/* Re-enabled on recovery completion. */
pm_runtime_disable(adev->dev);
@@ -395,11 +395,11 @@ static int avs_dsp_do_send_msg(struct avs_dev *adev, struct avs_ipc_msg *request
struct avs_ipc *ipc = adev->ipc;
int ret;
+ guard(mutex)(&ipc->msg_mutex);
+
if (!ipc->ready)
return -EPERM;
- guard(mutex)(&ipc->msg_mutex);
-
spin_lock(&ipc->rx_lock);
avs_ipc_msg_init(ipc, reply);
avs_dsp_send_tx(adev, request, true);
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -533,12 +533,14 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
select SND_SOC_CS35L56_SPI
select SND_SOC_CS35L56_SDW
select SND_SOC_ES9356
+ imply SND_SOC_TAC5XX2_SDW
select SND_SOC_DMIC
select SND_SOC_INTEL_HDA_DSP_COMMON
imply SND_SOC_SDW_MOCKUP
help
Add support for Intel SoundWire-based platforms connected to
- MAX98373, RT700, RT711, RT1308 and RT715
+ MAX98373, RT700, RT711, RT1308, RT715, TAC5XX2_SDW family (including
+ TAC5572, TAC5682).
If unsure select "N".
endif
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -908,6 +908,14 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_PORT_BT_OFFLOAD(2) |
SOF_BT_OFFLOAD_PRESENT),
},
+ {
+ .name = "nvl_rt5682_def",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
+ },
{
.name = "ptl_rt5682_c1_h02",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
diff --git a/sound/soc/intel/common/soc-acpi-intel-nvl-match.c b/sound/soc/intel/common/soc-acpi-intel-nvl-match.c
--- a/sound/soc/intel/common/soc-acpi-intel-nvl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-nvl-match.c
@@ -51,10 +51,10 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_machines[] = {
},
{
.comp_ids = &nvl_rt5682_rt5682s_hp,
- .drv_name = "sof_rt5682",
- .sof_tplg_filename = "sof-nvl-rt5682", /* the tplg suffix is added at run time */
- .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER |
- SND_SOC_ACPI_TPLG_INTEL_SSP_MSB,
+ .drv_name = "nvl_rt5682_def",
+ .sof_tplg_filename = "sof-nvl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
/* place amp/hdmi-in only boards in the end of table */
{
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