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 #include #include #include #include #include #include -#include #include #include -#include #include #include #include @@ -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 #include #include +#include #include #include #include +#include #include #include #include @@ -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 #include #include +#include #include #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 #include #include +#include #include #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 #include #include +#include #include #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 #include #include +#include #include #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 #include #include +#include #include #include #include @@ -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 */ {