diff options
Diffstat (limited to 'sound/soc')
50 files changed, 401 insertions, 162 deletions
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 3ec15b46fa35..15a864dcd7bd 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -217,6 +217,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82QF"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "82TL"), } }, @@ -224,12 +231,26 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82UG"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "82V2"), } }, { .driver_data = &acp6x_card, .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82YM"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"), } @@ -265,6 +286,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { { .driver_data = &acp6x_card, .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"), DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"), } diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 8ee1baa03269..87dd0ccade4c 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -452,11 +452,13 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, if ((aw_bin->all_bin_parse_num != 1) || (aw_bin->header_info[0].bin_data_type != DATA_TYPE_REGISTER)) { dev_err(aw_dev->dev, "bin num or type error"); + ret = -EINVAL; goto parse_bin_failed; } if (aw_bin->header_info[0].valid_data_len % 4) { dev_err(aw_dev->dev, "bin data len get error!"); + ret = -EINVAL; goto parse_bin_failed; } diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 9f4f2f4f23f5..d10e0e2380e8 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return -ENOMEM; cs35l56->base.dev = dev; - cs35l56->base.can_hibernate = true; i2c_set_clientdata(client, cs35l56); cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 600b79c62ec4..f9059780b7a7 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -706,7 +706,7 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56) mutex_lock(&cs35l56->base.irq_lock); - init_completion(&cs35l56->init_completion); + reinit_completion(&cs35l56->init_completion); cs35l56->soft_resetting = true; cs35l56_system_reset(&cs35l56->base, !!cs35l56->sdw_peripheral); @@ -1186,6 +1186,12 @@ post_soft_reset: /* Registers could be dirty after soft reset or SoundWire enumeration */ regcache_sync(cs35l56->base.regmap); + /* Set ASP1 DOUT to high-impedance when it is not transmitting audio data. */ + ret = regmap_set_bits(cs35l56->base.regmap, CS35L56_ASP1_CONTROL3, + CS35L56_ASP1_DOUT_HIZ_CTRL_MASK); + if (ret) + return dev_err_probe(cs35l56->base.dev, ret, "Failed to write ASP1_CONTROL3\n"); + cs35l56->base.init_done = true; complete(&cs35l56->init_completion); @@ -1207,6 +1213,7 @@ void cs35l56_remove(struct cs35l56_private *cs35l56) flush_workqueue(cs35l56->dsp_wq); destroy_workqueue(cs35l56->dsp_wq); + pm_runtime_dont_use_autosuspend(cs35l56->base.dev); pm_runtime_suspend(cs35l56->base.dev); pm_runtime_disable(cs35l56->base.dev); diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index eeab07c850f9..94a66a325303 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -6,6 +6,7 @@ #include <linux/acpi.h> #include <linux/device.h> +#include <linux/gpio/consumer.h> #include <linux/iopoll.h> #include <linux/module.h> #include <linux/mod_devicetable.h> @@ -344,6 +345,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral, switch (status) { case SDW_SLAVE_ATTACHED: dev_dbg(cs42l42->dev, "ATTACHED\n"); + + /* + * The SoundWire core can report stale ATTACH notifications + * if we hard-reset CS42L42 in probe() but it had already been + * enumerated. Reject the ATTACH if we haven't yet seen an + * UNATTACH report for the device being in reset. + */ + if (cs42l42->sdw_waiting_first_unattach) + break; + /* * Initialise codec, this only needs to be done once. * When resuming from suspend, resume callback will handle re-init of codec, @@ -354,6 +365,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral, break; case SDW_SLAVE_UNATTACHED: dev_dbg(cs42l42->dev, "UNATTACHED\n"); + + if (cs42l42->sdw_waiting_first_unattach) { + /* + * SoundWire core has seen that CS42L42 is not on + * the bus so release RESET and wait for ATTACH. + */ + cs42l42->sdw_waiting_first_unattach = false; + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + } + break; default: break; diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a0de0329406a..2961340f15e2 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2320,7 +2320,26 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42, if (cs42l42->reset_gpio) { dev_dbg(cs42l42->dev, "Found reset GPIO\n"); - gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + + /* + * ACPI can override the default GPIO state we requested + * so ensure that we start with RESET low. + */ + gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); + + /* Ensure minimum reset pulse width */ + usleep_range(10, 500); + + /* + * On SoundWire keep the chip in reset until we get an UNATTACH + * notification from the SoundWire core. This acts as a + * synchronization point to reject stale ATTACH notifications + * if the chip was already enumerated before we reset it. + */ + if (cs42l42->sdw_peripheral) + cs42l42->sdw_waiting_first_unattach = true; + else + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 4bd7b85a5747..7785125b73ab 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -53,6 +53,7 @@ struct cs42l42_private { u8 stream_use; bool hp_adc_up_pending; bool suspended; + bool sdw_waiting_first_unattach; bool init_done; }; diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 92e37bc1df9d..9f5f1a92561d 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -34,7 +34,7 @@ static const unsigned int cs42l43_accdet_db_ms[] = { static const unsigned int cs42l43_accdet_ramp_ms[] = { 10, 40, 90, 170 }; static const unsigned int cs42l43_accdet_bias_sense[] = { - 14, 23, 41, 50, 60, 68, 86, 95, 0, + 14, 24, 43, 52, 61, 71, 90, 99, 0, }; static int cs42l43_find_index(struct cs42l43_codec *priv, const char * const prop, diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 1a95c370fc4c..5643c666d7d0 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2077,7 +2077,8 @@ static const struct cs42l43_irq cs42l43_irqs[] = { static int cs42l43_request_irq(struct cs42l43_codec *priv, struct irq_domain *dom, const char * const name, - unsigned int irq, irq_handler_t handler) + unsigned int irq, irq_handler_t handler, + unsigned long flags) { int ret; @@ -2087,8 +2088,8 @@ static int cs42l43_request_irq(struct cs42l43_codec *priv, dev_dbg(priv->dev, "Request IRQ %d for %s\n", ret, name); - ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, IRQF_ONESHOT, - name, priv); + ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, + IRQF_ONESHOT | flags, name, priv); if (ret) return dev_err_probe(priv->dev, ret, "Failed to request IRQ %s\n", name); @@ -2124,11 +2125,11 @@ static int cs42l43_shutter_irq(struct cs42l43_codec *priv, return 0; } - ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler); + ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler, IRQF_SHARED); if (ret) return ret; - return cs42l43_request_irq(priv, dom, open_name, open_irq, handler); + return cs42l43_request_irq(priv, dom, open_name, open_irq, handler, IRQF_SHARED); } static int cs42l43_codec_probe(struct platform_device *pdev) @@ -2178,7 +2179,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev) for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) { ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name, - cs42l43_irqs[i].irq, cs42l43_irqs[i].handler); + cs42l43_irqs[i].irq, + cs42l43_irqs[i].handler, 0); if (ret) goto err_pm; } diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 581b334a6631..3bbe85091649 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,9 +59,6 @@ static void da7219_aad_btn_det_work(struct work_struct *work) bool micbias_up = false; int retries = 0; - /* Disable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); - /* Drive headphones/lineout */ snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, @@ -155,9 +152,6 @@ static void da7219_aad_hptest_work(struct work_struct *work) tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ_INT_OSC); } - /* Disable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); - /* Ensure gain ramping at fastest rate */ gain_ramp_ctrl = snd_soc_component_read(component, DA7219_GAIN_RAMP_CTRL); snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_X8); @@ -421,6 +415,11 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) * handle a removal, and we can check at the end of * hptest if we have a valid result or not. */ + + cancel_delayed_work_sync(&da7219_aad->jack_det_work); + /* Disable ground switch */ + snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); + if (statusa & DA7219_JACK_TYPE_STS_MASK) { report |= SND_JACK_HEADSET; mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 13689e718d36..09eef6042aad 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -531,7 +531,10 @@ static int hdmi_codec_fill_codec_params(struct snd_soc_dai *dai, hp->sample_rate = sample_rate; hp->channels = channels; - hcp->chmap_idx = idx; + if (pcm_audio) + hcp->chmap_idx = ca_id; + else + hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; return 0; } diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index ec6859ec0d38..fff4a8b862a7 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1675,12 +1675,12 @@ static int wsa_macro_spk_boost_event(struct snd_soc_dapm_widget *w, u16 boost_path_ctl, boost_path_cfg1; u16 reg, reg_mix; - if (!strcmp(w->name, "WSA_RX INT0 CHAIN")) { + if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT0 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST0_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX0_RX_PATH_CFG1; reg = CDC_WSA_RX0_RX_PATH_CTL; reg_mix = CDC_WSA_RX0_RX_PATH_MIX_CTL; - } else if (!strcmp(w->name, "WSA_RX INT1 CHAIN")) { + } else if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT1 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST1_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX1_RX_PATH_CFG1; reg = CDC_WSA_RX1_RX_PATH_CTL; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 15e1a62b9e57..e8cdc166bdaa 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2403,13 +2403,11 @@ static irqreturn_t rt5640_irq(int irq, void *data) struct rt5640_priv *rt5640 = data; int delay = 0; - if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { - cancel_delayed_work_sync(&rt5640->jack_work); + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) delay = 100; - } if (rt5640->jack) - queue_delayed_work(system_long_wq, &rt5640->jack_work, delay); + mod_delayed_work(system_long_wq, &rt5640->jack_work, delay); return IRQ_HANDLED; } @@ -2565,10 +2563,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, if (jack_data && jack_data->use_platform_clock) rt5640->use_platform_clock = jack_data->use_platform_clock; - ret = devm_request_threaded_irq(component->dev, rt5640->irq, - NULL, rt5640_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "rt5640", rt5640); + ret = request_irq(rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640_disable_jack_detect(component); @@ -2621,14 +2618,14 @@ static void rt5640_enable_hda_jack_detect( rt5640->jack = jack; - ret = devm_request_threaded_irq(component->dev, rt5640->irq, - NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT, - "rt5640", rt5640); + ret = request_irq(rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); - rt5640->irq = -ENXIO; + rt5640->jack = NULL; return; } + rt5640->irq_requested = true; /* sync initial jack state */ queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); @@ -2801,12 +2798,12 @@ static int rt5640_suspend(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); - if (rt5640->irq) { + if (rt5640->jack) { /* disable jack interrupts during system suspend */ disable_irq(rt5640->irq); + rt5640_cancel_work(rt5640); } - rt5640_cancel_work(rt5640); snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); rt5640_reset(component); regcache_cache_only(rt5640->regmap, true); @@ -2829,9 +2826,6 @@ static int rt5640_resume(struct snd_soc_component *component) regcache_cache_only(rt5640->regmap, false); regcache_sync(rt5640->regmap); - if (rt5640->irq) - enable_irq(rt5640->irq); - if (rt5640->jack) { if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { snd_soc_component_update_bits(component, @@ -2859,6 +2853,7 @@ static int rt5640_resume(struct snd_soc_component *component) } } + enable_irq(rt5640->irq); queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1a137ca3f496..7938b52d741d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3257,6 +3257,8 @@ int rt5645_set_jack_detect(struct snd_soc_component *component, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); } rt5645_irq(0, rt5645); diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index b05b4f73d8aa..fbad1ed06626 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -157,11 +157,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators, - rt5682); - if (ret) - return ret; - ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); if (ret) { @@ -169,6 +164,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators, + rt5682); + if (ret) + return ret; + ret = rt5682_get_ldo1(rt5682, &i2c->dev); if (ret) return ret; diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 86bd6c18a944..41076be23854 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -39,7 +39,7 @@ static void tas2780_reset(struct tas2780_priv *tas2780) usleep_range(2000, 2050); } - snd_soc_component_write(tas2780->component, TAS2780_SW_RST, + ret = snd_soc_component_write(tas2780->component, TAS2780_SW_RST, TAS2780_RST); if (ret) dev_err(tas2780->dev, "%s:errCode:0x%x Reset error!\n", diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index b976c1946286..420bbf588efe 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -293,7 +293,7 @@ #define ADC3XXX_BYPASS_RPGA 0x80 /* MICBIAS control bits */ -#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS_MASK 0x3 #define ADC3XXX_MICBIAS1_SHIFT 5 #define ADC3XXX_MICBIAS2_SHIFT 3 @@ -1099,7 +1099,7 @@ static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, unsigned int val; if (!of_property_read_u32(np, propname, &val)) { - if (val >= ADC3XXX_MICBIAS_AVDD) { + if (val > ADC3XXX_MICBIAS_AVDD) { dev_err(dev, "Invalid property value for '%s'\n", propname); return -EINVAL; } diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index 6951120057e5..a1f04010da95 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1278,7 +1278,31 @@ static int wcd9380_probe(struct sdw_slave *pdev, pm_runtime_set_active(dev); pm_runtime_enable(dev); - return component_add(dev, &wcd938x_sdw_component_ops); + ret = component_add(dev, &wcd938x_sdw_component_ops); + if (ret) + goto err_disable_rpm; + + return 0; + +err_disable_rpm: + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); + + return ret; +} + +static int wcd9380_remove(struct sdw_slave *pdev) +{ + struct device *dev = &pdev->dev; + + component_del(dev, &wcd938x_sdw_component_ops); + + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); + + return 0; } static const struct sdw_device_id wcd9380_slave_id[] = { @@ -1320,6 +1344,7 @@ static const struct dev_pm_ops wcd938x_sdw_pm_ops = { static struct sdw_driver wcd9380_codec_driver = { .probe = wcd9380_probe, + .remove = wcd9380_remove, .ops = &wcd9380_slave_ops, .id_table = wcd9380_slave_id, .driver = { diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index a3c680661377..d27b919c63b4 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3325,8 +3325,10 @@ static int wcd938x_populate_dt_data(struct wcd938x_priv *wcd938x, struct device return dev_err_probe(dev, ret, "Failed to get supplies\n"); ret = regulator_bulk_enable(WCD938X_MAX_SUPPLY, wcd938x->supplies); - if (ret) + if (ret) { + regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); return dev_err_probe(dev, ret, "Failed to enable supplies\n"); + } wcd938x_dt_parse_micbias_info(dev, wcd938x); @@ -3435,7 +3437,8 @@ static int wcd938x_bind(struct device *dev) wcd938x->rxdev = wcd938x_sdw_device_get(wcd938x->rxnode); if (!wcd938x->rxdev) { dev_err(dev, "could not find slave with matching of node\n"); - return -EINVAL; + ret = -EINVAL; + goto err_unbind; } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; @@ -3443,46 +3446,47 @@ static int wcd938x_bind(struct device *dev) wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { dev_err(dev, "could not find txslave with matching of node\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put_rxdev; } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); - if (!wcd938x->tx_sdw_dev) { - dev_err(dev, "could not get txslave with matching of dev\n"); - return -EINVAL; - } /* As TX is main CSR reg interface, which should not be suspended first. * expicilty add the dependency link */ if (!device_link_add(wcd938x->rxdev, wcd938x->txdev, DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME)) { dev_err(dev, "could not devlink tx and rx\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put_txdev; } if (!device_link_add(dev, wcd938x->txdev, DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME)) { dev_err(dev, "could not devlink wcd and tx\n"); - return -EINVAL; + ret = -EINVAL; + goto err_remove_rxtx_link; } if (!device_link_add(dev, wcd938x->rxdev, DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME)) { dev_err(dev, "could not devlink wcd and rx\n"); - return -EINVAL; + ret = -EINVAL; + goto err_remove_tx_link; } wcd938x->regmap = dev_get_regmap(&wcd938x->tx_sdw_dev->dev, NULL); if (!wcd938x->regmap) { dev_err(dev, "could not get TX device regmap\n"); - return -EINVAL; + ret = -EINVAL; + goto err_remove_rx_link; } ret = wcd938x_irq_init(wcd938x, dev); if (ret) { dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); - return ret; + goto err_remove_rx_link; } wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; @@ -3491,27 +3495,45 @@ static int wcd938x_bind(struct device *dev) ret = wcd938x_set_micbias_data(wcd938x); if (ret < 0) { dev_err(dev, "%s: bad micbias pdata\n", __func__); - return ret; + goto err_remove_rx_link; } ret = snd_soc_register_component(dev, &soc_codec_dev_wcd938x, wcd938x_dais, ARRAY_SIZE(wcd938x_dais)); - if (ret) + if (ret) { dev_err(dev, "%s: Codec registration failed\n", __func__); + goto err_remove_rx_link; + } - return ret; + return 0; +err_remove_rx_link: + device_link_remove(dev, wcd938x->rxdev); +err_remove_tx_link: + device_link_remove(dev, wcd938x->txdev); +err_remove_rxtx_link: + device_link_remove(wcd938x->rxdev, wcd938x->txdev); +err_put_txdev: + put_device(wcd938x->txdev); +err_put_rxdev: + put_device(wcd938x->rxdev); +err_unbind: + component_unbind_all(dev, wcd938x); + + return ret; } static void wcd938x_unbind(struct device *dev) { struct wcd938x_priv *wcd938x = dev_get_drvdata(dev); + snd_soc_unregister_component(dev); device_link_remove(dev, wcd938x->txdev); device_link_remove(dev, wcd938x->rxdev); device_link_remove(wcd938x->rxdev, wcd938x->txdev); - snd_soc_unregister_component(dev); + put_device(wcd938x->txdev); + put_device(wcd938x->rxdev); component_unbind_all(dev, wcd938x); } @@ -3572,13 +3594,13 @@ static int wcd938x_probe(struct platform_device *pdev) ret = wcd938x_add_slave_components(wcd938x, dev, &match); if (ret) - return ret; + goto err_disable_regulators; wcd938x_reset(wcd938x); ret = component_master_add_with_match(dev, &wcd938x_comp_ops, match); if (ret) - return ret; + goto err_disable_regulators; pm_runtime_set_autosuspend_delay(dev, 1000); pm_runtime_use_autosuspend(dev); @@ -3588,11 +3610,27 @@ static int wcd938x_probe(struct platform_device *pdev) pm_runtime_idle(dev); return 0; + +err_disable_regulators: + regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies); + regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); + + return ret; } static void wcd938x_remove(struct platform_device *pdev) { - component_master_del(&pdev->dev, &wcd938x_comp_ops); + struct device *dev = &pdev->dev; + struct wcd938x_priv *wcd938x = dev_get_drvdata(dev); + + component_master_del(dev, &wcd938x_comp_ops); + + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); + + regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies); + regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); } #if defined(CONFIG_OF) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a50180750e8..7689fe3cc86d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1468,8 +1468,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) } wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); - if (IS_ERR(wm8960->regmap)) - return PTR_ERR(wm8960->regmap); + if (IS_ERR(wm8960->regmap)) { + ret = PTR_ERR(wm8960->regmap); + goto bulk_disable; + } if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); @@ -1479,13 +1481,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) ret = i2c_master_recv(i2c, &val, sizeof(val)); if (ret >= 0) { dev_err(&i2c->dev, "Not wm8960, wm8960 reg can not read by i2c\n"); - return -EINVAL; + ret = -EINVAL; + goto bulk_disable; } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); - return ret; + goto bulk_disable; } if (wm8960->pdata.shared_lrclk) { @@ -1494,7 +1497,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) if (ret != 0) { dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", ret); - return ret; + goto bulk_disable; } } @@ -1528,7 +1531,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8960, &wm8960_dai, 1); + if (ret) + goto bulk_disable; + return 0; + +bulk_disable: + regulator_bulk_disable(ARRAY_SIZE(wm8960->supplies), wm8960->supplies); return ret; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6fc34f41b175..d1b9238d391e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -687,7 +687,10 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, struct wm_coeff_ctl *ctl; int ret; + mutex_lock(&dsp->cs_dsp.pwr_lock); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); + mutex_unlock(&dsp->cs_dsp.pwr_lock); + if (ret < 0) return ret; @@ -703,8 +706,14 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), - 0, buf, len); + int ret; + + mutex_lock(&dsp->cs_dsp.pwr_lock); + ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), + 0, buf, len); + mutex_unlock(&dsp->cs_dsp.pwr_lock); + + return ret; } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 22c004179214..9ea4be56d3b7 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -917,7 +917,7 @@ static int jh7110_i2stx0_clk_cfg(struct i2s_clk_config_data *config) static int dw_i2s_probe(struct platform_device *pdev) { - const struct i2s_platform_data *pdata = of_device_get_match_data(&pdev->dev); + const struct i2s_platform_data *pdata = pdev->dev.platform_data; struct dw_i2s_dev *dev; struct resource *res; int ret, irq; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 76b5bfc288fd..bab7d34cf585 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -52,8 +52,8 @@ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; - u32 fll_id; - u32 pll_id; + int fll_id; + int pll_id; }; /** @@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } /* Specific configuration for PLL */ - if (codec_priv->pll_id && codec_priv->fll_id) { + if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * 384; else @@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) priv->streams &= ~BIT(substream->stream); - if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), codec_priv->mclk_id, @@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; + + priv->codec_priv.fll_id = -1; + priv->codec_priv.pll_id = -1; + /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1e4020fae05a..8a9a30dd31e2 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -710,10 +710,15 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) { unsigned int ofs = sai->soc_data->reg_offset; bool tx = dir == TX; - u32 xcsr, count = 100; + u32 xcsr, count = 100, mask; + + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output) + mask = FSL_SAI_CSR_TERE; + else + mask = FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE; regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), - FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0); + mask, 0); /* TERE will remain set till the end of current frame */ do { diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 0b58df56f4da..aeb81aa61184 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -315,7 +315,7 @@ static int imx_audmix_probe(struct platform_device *pdev) if (IS_ERR(priv->cpu_mclk)) { ret = PTR_ERR(priv->cpu_mclk); dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); - return -EINVAL; + return ret; } priv->audmix_pdev = audmix_pdev; diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index d63782b8bdef..bb736d45c9e0 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -19,6 +19,7 @@ static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 3c7b95db2eac..b578f9a32d7f 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -89,6 +89,14 @@ static int imx_rpmsg_probe(struct platform_device *pdev) SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + /* + * i.MX rpmsg sound cards work on codec slave mode. MCLK will be + * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK + * present at power up/down sequence. So need to set ignore_pmdown_time + * to power down codec immediately before MCLK is turned off. + */ + data->dai.ignore_pmdown_time = 1; + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 5b18a4af022f..2588ec735dbd 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -310,7 +310,8 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) if (fixed_sysclk % props->mclk_fs) { dev_err(rtd->dev, "fixed sysclk %u not divisible by mclk_fs %u\n", fixed_sysclk, props->mclk_fs); - return -EINVAL; + ret = -EINVAL; + goto codec_err; } ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, fixed_rate, fixed_rate); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 190f11366e84..274417e39e7d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -759,10 +759,12 @@ static int asoc_simple_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link = priv->dai_link; struct simple_dai_props *dai_props = priv->dai_props; + ret = -EINVAL; + cinfo = dev->platform_data; if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + goto err; } if (!cinfo->name || @@ -771,7 +773,7 @@ static int asoc_simple_probe(struct platform_device *pdev) !cinfo->platform || !cinfo->cpu_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + goto err; } cpus = dai_link->cpus; diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index cb00bc86ac94..8876558f19a1 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -55,6 +55,9 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int return -ENOMEM; dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + if (!dl[i].codecs->name) + return -ENOMEM; + dl[i].codecs->dai_name = pcm->name; dl[i].num_codecs = 1; dl[i].num_cpus = 1; diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index f8a3e8a91761..9904a9e33ccc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -808,6 +808,16 @@ static const struct platform_device_id board_ids[] = { SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | SOF_ES8336_JD_INVERTED), }, + { + .name = "mtl_es83x6_c1_h02", + .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | + SOF_NO_OF_HDMI_CAPTURE_SSP(2) | + SOF_HDMI_CAPTURE_1_SSP(0) | + SOF_HDMI_CAPTURE_2_SSP(2) | + SOF_SSP_HDMI_CAPTURE_PRESENT | + SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | + SOF_ES8336_JD_INVERTED), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5a1c750e6ae6..842649501e30 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -380,6 +380,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, + + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B29"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 8e995edf4c10..5103e75ac830 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -656,18 +656,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l3.tplg", }, { - .link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */ - .links = adl_sdw_rt1316_link1_rt714_link0, - .drv_name = "sof_sdw", - .sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg", - }, - { .link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */ .links = adl_sdw_rt1316_link12_rt714_link0, .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt1316-l12-rt714-l0.tplg", }, { + .link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */ + .links = adl_sdw_rt1316_link1_rt714_link0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg", + }, + { .link_mask = 0x5, /* 2 active links required */ .links = adl_sdw_rt1316_link2_rt714_link0, .drv_name = "sof_sdw", diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 0304246d2922..92498d1d6c8d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -30,6 +30,16 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + +static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = { + .num_codecs = 1, + .codecs = {"INTC10B0"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -52,6 +62,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "mtl_es83x6_c1_h02", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mtl_lt6911_hdmi, + .sof_tplg_filename = "sof-mtl-es83x6-ssp1-hdmi-ssp02.tplg", + }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-mtl-es8336", /* 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 | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d86880169075..bc2f2849ecfb 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -112,34 +112,6 @@ static int axg_spdifin_prepare(struct snd_pcm_substream *substream, return 0; } -static int axg_spdifin_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = clk_prepare_enable(priv->refclk); - if (ret) { - dev_err(dai->dev, - "failed to enable spdifin reference clock\n"); - return ret; - } - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, - SPDIFIN_CTRL0_EN); - - return 0; -} - -static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); - clk_disable_unprepare(priv->refclk); -} - static void axg_spdifin_write_mode_param(struct regmap *map, int mode, unsigned int val, unsigned int num_per_reg, @@ -251,17 +223,32 @@ static int axg_spdifin_dai_probe(struct snd_soc_dai *dai) ret = axg_spdifin_sample_mode_config(dai, priv); if (ret) { dev_err(dai->dev, "mode configuration failed\n"); - clk_disable_unprepare(priv->pclk); - return ret; + goto pclk_err; } + ret = clk_prepare_enable(priv->refclk); + if (ret) { + dev_err(dai->dev, + "failed to enable spdifin reference clock\n"); + goto pclk_err; + } + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, + SPDIFIN_CTRL0_EN); + return 0; + +pclk_err: + clk_disable_unprepare(priv->pclk); + return ret; } static int axg_spdifin_dai_remove(struct snd_soc_dai *dai) { struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); + clk_disable_unprepare(priv->refclk); clk_disable_unprepare(priv->pclk); return 0; } @@ -270,8 +257,6 @@ static const struct snd_soc_dai_ops axg_spdifin_ops = { .probe = axg_spdifin_dai_probe, .remove = axg_spdifin_dai_remove, .prepare = axg_spdifin_prepare, - .startup = axg_spdifin_startup, - .shutdown = axg_spdifin_shutdown, }; static int axg_spdifin_iec958_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b70034c07eee..b8a3cb8b7597 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -773,7 +773,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (IS_ERR(priv->extclk)) { ret = PTR_ERR(priv->extclk); if (ret == -EPROBE_DEFER) - return ret; + goto err_priv; priv->extclk = NULL; } diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e29c2fee9521..1bd7114c472a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1303,6 +1303,7 @@ audio_graph: if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); of_node_put(node); + of_node_put(ports); break; } } diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ba7c0ae82e00..566033f7dd2e 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -242,6 +242,7 @@ int snd_soc_component_notify_control(struct snd_soc_component *component, char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; struct snd_kcontrol *kctl; + /* When updating, change also snd_soc_dapm_widget_name_cmp() */ if (component->name_prefix) snprintf(name, ARRAY_SIZE(name), "%s %s", component->name_prefix, ctl); else diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cc442c52cdea..9de98c01d815 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1347,7 +1347,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, snd_soc_runtime_get_dai_fmt(rtd); ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); if (ret) - return ret; + goto err; /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); @@ -1372,17 +1372,26 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* create compress_device if possible */ ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); if (ret != -ENOTSUPP) - return ret; + goto err; /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); - return ret; + goto err; } - return snd_soc_pcm_dai_new(rtd); + ret = snd_soc_pcm_dai_new(rtd); + if (ret < 0) + goto err; + + rtd->initialized = true; + + return 0; +err: + snd_soc_link_exit(rtd); + return ret; } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1445,8 +1454,8 @@ static int soc_probe_component(struct snd_soc_card *card, if (component->card) { if (component->card != card) { dev_err(component->dev, - "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", - card->name, component->card->name); + "Trying to bind component \"%s\" to card \"%s\" but is already bound to card \"%s\"\n", + component->name, card->name, component->card->name); return -ENODEV; } return 0; @@ -1980,7 +1989,8 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) /* release machine specific resources */ for_each_card_rtds(card, rtd) - snd_soc_link_exit(rtd); + if (rtd->initialized) + snd_soc_link_exit(rtd); /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f07e83678373..312e55579831 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2728,6 +2728,18 @@ int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai); +int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm); + const char *wname = widget->name; + + if (component->name_prefix) + wname += strlen(component->name_prefix) + 1; /* plus space */ + + return strcmp(wname, s); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_widget_name_cmp); + /* * dapm_update_widget_flags() - Re-compute widget sink and source flags * @w: The widget for which to update the flags diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d0653d775c87..cad222eb9a29 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -44,8 +44,8 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, * platforms which make use of the snd_dmaengine_dai_dma_data struct for their * DAI DMA data. Internally the function will first call * snd_hwparams_to_dma_slave_config to fill in the slave config based on the - * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the - * remaining fields based on the DAI DMA data. + * hw_params, followed by snd_dmaengine_pcm_set_config_from_dai_data to fill in + * the remaining fields based on the DAI DMA data. */ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb0723876851..54704250c0a2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -985,6 +985,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; + struct snd_pcm_hw_params tmp_params; int i, ret = 0; snd_soc_dpcm_mutex_assert_held(rtd); @@ -998,7 +999,6 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { - struct snd_pcm_hw_params codec_params; unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream); /* @@ -1019,23 +1019,22 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, continue; /* copy params for each codec */ - codec_params = *params; + tmp_params = *params; /* fixup params based on TDM slot masks */ if (tdm_mask) - soc_pcm_codec_params_fixup(&codec_params, tdm_mask); + soc_pcm_codec_params_fixup(&tmp_params, tdm_mask); ret = snd_soc_dai_hw_params(codec_dai, substream, - &codec_params); + &tmp_params); if(ret < 0) goto out; - soc_pcm_set_dai_params(codec_dai, &codec_params); - snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); + soc_pcm_set_dai_params(codec_dai, &tmp_params); + snd_soc_dapm_update_dai(substream, &tmp_params, codec_dai); } for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - struct snd_pcm_hw_params cpu_params; unsigned int ch_mask = 0; int j; @@ -1047,7 +1046,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, continue; /* copy params for each cpu */ - cpu_params = *params; + tmp_params = *params; if (!rtd->dai_link->codec_ch_maps) goto hw_params; @@ -1062,16 +1061,16 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, /* fixup cpu channel number */ if (ch_mask) - soc_pcm_codec_params_fixup(&cpu_params, ch_mask); + soc_pcm_codec_params_fixup(&tmp_params, ch_mask); hw_params: - ret = snd_soc_dai_hw_params(cpu_dai, substream, &cpu_params); + ret = snd_soc_dai_hw_params(cpu_dai, substream, &tmp_params); if (ret < 0) goto out; /* store the parameters for each DAI */ - soc_pcm_set_dai_params(cpu_dai, &cpu_params); - snd_soc_dapm_update_dai(substream, &cpu_params, cpu_dai); + soc_pcm_set_dai_params(cpu_dai, &tmp_params); + snd_soc_dapm_update_dai(substream, &tmp_params, cpu_dai); } ret = snd_soc_pcm_component_hw_params(substream, params); diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 11607c5f5d5a..9c746e4edef7 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -217,6 +217,7 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) return 1; return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy); int snd_soc_component_is_dummy(struct snd_soc_component *component) { diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 9935e457b467..a7ae76efc2dd 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -35,7 +35,6 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = { .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, - .acp_clkmux_sel = ACP6X_CLKMUX_SEL, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, .probe_reg_offset = ACP6X_FUTURE_REG_ACLK_0, }; diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 30db685cc5f4..2d1616b81485 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -486,10 +486,9 @@ int snd_sof_device_remove(struct device *dev) snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_remove(sdev); + sof_ops_free(sdev); } - sof_ops_free(sdev); - /* release firmware */ snd_sof_fw_unload(sdev); diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index b84ca58da9d5..f9412517eaf2 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -460,7 +460,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) /* step 3: wait for IPC DONE bit from ROM */ ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->ipc_ack, status, ((status & chip->ipc_ack_mask) == chip->ipc_ack_mask), - HDA_DSP_REG_POLL_INTERVAL_US, MTL_DSP_PURGE_TIMEOUT_US); + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_INIT_TIMEOUT_US); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, "timeout waiting for purge IPC done\n"); diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 02181490f12a..95696b3d7c4c 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -62,7 +62,6 @@ #define MTL_DSP_IRQSTS_IPC BIT(0) #define MTL_DSP_IRQSTS_SDW BIT(6) -#define MTL_DSP_PURGE_TIMEOUT_US 20000000 /* 20s */ #define MTL_DSP_REG_POLL_INTERVAL_US 10 /* 10 us */ /* Memory windows */ diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f2a30cd31378..7cb63e6b24dc 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -231,7 +231,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp, ret = sof_update_ipc_object(scomp, available_fmt, SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(available_fmt), 1); + swidget->num_tuples, sizeof(*available_fmt), 1); if (ret) { dev_err(scomp->dev, "Failed to parse audio format token count\n"); return ret; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index e7ef77012c35..e5405f854a91 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -212,7 +212,8 @@ widget_free: sof_widget_free_unlocked(sdev, swidget); use_count_decremented = true; core_put: - snd_sof_dsp_core_put(sdev, swidget->core); + if (!use_count_decremented) + snd_sof_dsp_core_put(sdev, swidget->core); pipe_widget_free: if (swidget->id != snd_soc_dapm_scheduler) sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget); diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index 1f2c5018bf5a..4737e776d383 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -10,6 +10,7 @@ #include <linux/platform_device.h> #include <sound/graph_card.h> #include <sound/pcm_params.h> +#include <sound/soc-dai.h> #define MAX_PLLA_OUT0_DIV 128 @@ -44,6 +45,21 @@ struct tegra_audio_cdata { unsigned int plla_out0_rates[NUM_RATE_TYPE]; }; +static bool need_clk_update(struct snd_soc_dai *dai) +{ + if (snd_soc_dai_is_dummy(dai) || + !dai->driver->ops || + !dai->driver->name) + return false; + + if (strstr(dai->driver->name, "I2S") || + strstr(dai->driver->name, "DMIC") || + strstr(dai->driver->name, "DSPK")) + return true; + + return false; +} + /* Setup PLL clock as per the given sample rate */ static int tegra_audio_graph_update_pll(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -140,19 +156,7 @@ static int tegra_audio_graph_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; - /* - * This gets called for each DAI link (FE or BE) when DPCM is used. - * We may not want to update PLLA rate for each call. So PLLA update - * must be restricted to external I/O links (I2S, DMIC or DSPK) since - * they actually depend on it. I/O modules update their clocks in - * hw_param() of their respective component driver and PLLA rate - * update here helps them to derive appropriate rates. - * - * TODO: When more HW accelerators get added (like sample rate - * converter, volume gain controller etc., which don't really - * depend on PLLA) we need a better way to filter here. - */ - if (cpu_dai->driver->ops && rtd->dai_link->no_pcm) { + if (need_clk_update(cpu_dai)) { err = tegra_audio_graph_update_pll(substream, params); if (err) return err; |