diff options
29 files changed, 266 insertions, 92 deletions
diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index 8132d0c0f00a..35e669020296 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -378,6 +378,8 @@ examples: - | sound { compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; simple-audio-card,name = "rsnd-ak4643"; simple-audio-card,format = "left_j"; @@ -391,10 +393,12 @@ examples: "ak4642 Playback", "DAI1 Playback"; dpcmcpu: simple-audio-card,cpu@0 { + reg = <0>; sound-dai = <&rcar_sound 0>; }; simple-audio-card,cpu@1 { + reg = <1>; sound-dai = <&rcar_sound 1>; }; @@ -418,6 +422,8 @@ examples: - | sound { compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; simple-audio-card,routing = "pcm3168a Playback", "DAI1 Playback", @@ -426,6 +432,7 @@ examples: "pcm3168a Playback", "DAI4 Playback"; simple-audio-card,dai-link@0 { + reg = <0>; format = "left_j"; bitclock-master = <&sndcpu0>; frame-master = <&sndcpu0>; @@ -439,22 +446,23 @@ examples: }; simple-audio-card,dai-link@1 { + reg = <1>; format = "i2s"; bitclock-master = <&sndcpu1>; frame-master = <&sndcpu1>; convert-channels = <8>; /* TDM Split */ - sndcpu1: cpu@0 { + sndcpu1: cpu0 { sound-dai = <&rcar_sound 1>; }; - cpu@1 { + cpu1 { sound-dai = <&rcar_sound 2>; }; - cpu@2 { + cpu2 { sound-dai = <&rcar_sound 3>; }; - cpu@3 { + cpu3 { sound-dai = <&rcar_sound 4>; }; codec { @@ -466,6 +474,7 @@ examples: }; simple-audio-card,dai-link@2 { + reg = <2>; format = "i2s"; bitclock-master = <&sndcpu2>; frame-master = <&sndcpu2>; diff --git a/MAINTAINERS b/MAINTAINERS index d53db30d1365..0887816d125e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6956,6 +6956,7 @@ M: Timur Tabi <timur@kernel.org> M: Nicolin Chen <nicoleotsuka@gmail.com> M: Xiubo Li <Xiubo.Lee@gmail.com> R: Fabio Estevam <festevam@gmail.com> +R: Shengjiu Wang <shengjiu.wang@gmail.com> L: alsa-devel@alsa-project.org (moderated for non-subscribers) L: linuxppc-dev@lists.ozlabs.org S: Maintained @@ -11333,17 +11334,17 @@ F: drivers/iio/adc/at91-sama5d2_adc.c F: include/dt-bindings/iio/adc/at91-sama5d2_adc.h MICROCHIP SAMA5D2-COMPATIBLE SHUTDOWN CONTROLLER -M: Nicolas Ferre <nicolas.ferre@microchip.com> +M: Claudiu Beznea <claudiu.beznea@microchip.com> S: Supported F: drivers/power/reset/at91-sama5d2_shdwc.c MICROCHIP SPI DRIVER -M: Nicolas Ferre <nicolas.ferre@microchip.com> +M: Tudor Ambarus <tudor.ambarus@microchip.com> S: Supported F: drivers/spi/spi-atmel.* MICROCHIP SSC DRIVER -M: Nicolas Ferre <nicolas.ferre@microchip.com> +M: Codrin Ciubotariu <codrin.ciubotariu@microchip.com> L: linux-arm-kernel@lists.infradead.org (moderated for non-subscribers) S: Supported F: drivers/misc/atmel-ssc.c diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index f9024c7a1600..02e1d7778354 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -12,6 +12,7 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; bool dev_gpio; + bool gpio1_is_ext_spk_en; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 212257e84fac..71e178c89793 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -161,6 +161,7 @@ void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream); +void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); static inline void snd_soc_dai_activate(struct snd_soc_dai *dai, diff --git a/include/sound/soc.h b/include/sound/soc.h index 2756f9bcac3e..3ce7f0f5aa92 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -444,6 +444,8 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *component_driver, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver); struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, diff --git a/sound/core/info.c b/sound/core/info.c index 8c6bc5241df5..9fec3070f8ba 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -606,7 +606,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { int c; - if (snd_BUG_ON(!buffer || !buffer->buffer)) + if (snd_BUG_ON(!buffer)) + return 1; + if (!buffer->buffer) return 1; if (len <= 0 || buffer->stop || buffer->error) return 1; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b06c4261248..1b2d8e56390a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7587,6 +7587,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index f25ce50f1a90..ebf4388b6262 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -232,9 +232,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pm_runtime_set_autosuspend_delay(&pci->dev, 2000); pm_runtime_use_autosuspend(&pci->dev); - pm_runtime_set_active(&pci->dev); pm_runtime_put_noidle(&pci->dev); - pm_runtime_enable(&pci->dev); pm_runtime_allow(&pci->dev); return 0; @@ -303,7 +301,7 @@ static void snd_acp3x_remove(struct pci_dev *pci) ret = acp3x_deinit(adata->acp3x_base); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); - pm_runtime_disable(&pci->dev); + pm_runtime_forbid(&pci->dev); pm_runtime_get_noresume(&pci->dev); pci_disable_msi(pci); pci_release_regions(pci); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 96718e3a1ad0..d87402a86c88 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -779,13 +779,6 @@ static int max98373_probe(struct snd_soc_component *component) regmap_write(max98373->regmap, MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2, 0x1); - /* Set inital volume (0dB) */ - regmap_write(max98373->regmap, - MAX98373_R203D_AMP_DIG_VOL_CTRL, - 0x00); - regmap_write(max98373->regmap, - MAX98373_R203E_AMP_PATH_GAIN, - 0x00); /* Enable DC blocker */ regmap_write(max98373->regmap, MAX98373_R203F_AMP_DSP_CFG, @@ -869,7 +862,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98373 = { .num_dapm_widgets = ARRAY_SIZE(max98373_dapm_widgets), .dapm_routes = max98373_audio_map, .num_dapm_routes = ARRAY_SIZE(max98373_audio_map), - .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..e8d14eefc41b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - if (!*mic) { + + if (!*hp) { snd_soc_dapm_disable_pin(dapm, "HV"); snd_soc_dapm_disable_pin(dapm, "VREF"); - } - if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + } return 0; } diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 70fee6849ab0..dfbc0ca38ff7 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -31,18 +31,19 @@ #include "rt5670.h" #include "rt5670-dsp.h" -#define RT5670_DEV_GPIO BIT(0) -#define RT5670_IN2_DIFF BIT(1) -#define RT5670_DMIC_EN BIT(2) -#define RT5670_DMIC1_IN2P BIT(3) -#define RT5670_DMIC1_GPIO6 BIT(4) -#define RT5670_DMIC1_GPIO7 BIT(5) -#define RT5670_DMIC2_INR BIT(6) -#define RT5670_DMIC2_GPIO8 BIT(7) -#define RT5670_DMIC3_GPIO5 BIT(8) -#define RT5670_JD_MODE1 BIT(9) -#define RT5670_JD_MODE2 BIT(10) -#define RT5670_JD_MODE3 BIT(11) +#define RT5670_DEV_GPIO BIT(0) +#define RT5670_IN2_DIFF BIT(1) +#define RT5670_DMIC_EN BIT(2) +#define RT5670_DMIC1_IN2P BIT(3) +#define RT5670_DMIC1_GPIO6 BIT(4) +#define RT5670_DMIC1_GPIO7 BIT(5) +#define RT5670_DMIC2_INR BIT(6) +#define RT5670_DMIC2_GPIO8 BIT(7) +#define RT5670_DMIC3_GPIO5 BIT(8) +#define RT5670_JD_MODE1 BIT(9) +#define RT5670_JD_MODE2 BIT(10) +#define RT5670_JD_MODE3 BIT(11) +#define RT5670_GPIO1_IS_EXT_SPK_EN BIT(12) static unsigned long rt5670_quirk; static unsigned int quirk_override; @@ -602,9 +603,9 @@ int rt5670_set_jack_detect(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(rt5670_set_jack_detect); static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ @@ -1447,6 +1448,33 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5670_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component); + + if (!rt5670->pdata.gpio1_is_ext_spk_en) + return 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO); + break; + + default: + return 0; + } + + return 0; +} + static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1860,7 +1888,9 @@ static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = { - SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + rt5670_spk_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), SND_SOC_DAPM_OUTPUT("SPORP"), @@ -2857,14 +2887,14 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { }, { .callback = rt5670_quirk_cb, - .ident = "Lenovo Thinkpad Tablet 10", + .ident = "Lenovo Miix 2 10", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"), }, .driver_data = (unsigned long *)(RT5670_DMIC_EN | RT5670_DMIC1_IN2P | - RT5670_DEV_GPIO | + RT5670_GPIO1_IS_EXT_SPK_EN | RT5670_JD_MODE2), }, { @@ -2924,6 +2954,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, rt5670->pdata.dev_gpio = true; dev_info(&i2c->dev, "quirk dev_gpio\n"); } + if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) { + rt5670->pdata.gpio1_is_ext_spk_en = true; + dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n"); + } if (rt5670_quirk & RT5670_IN2_DIFF) { rt5670->pdata.in2_diff = true; dev_info(&i2c->dev, "quirk IN2_DIFF\n"); @@ -3023,6 +3057,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); } + if (rt5670->pdata.gpio1_is_ext_spk_en) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); + } + if (rt5670->pdata.jd_mode) { regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a8c3e44770b8..de0203369b7c 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -757,7 +757,7 @@ #define RT5670_PWR_VREF2_BIT 4 #define RT5670_PWR_FV2 (0x1 << 3) #define RT5670_PWR_FV2_BIT 3 -#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_MASK (0x7) #define RT5670_LDO_SEL_SFT 0 /* Power Management for Analog 2 (0x64) */ diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7d6670abdb08..d503b5bef4ba 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -967,13 +967,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); - else + RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -992,16 +991,17 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, rt5682->hs_jack = hs_jack; - if (!rt5682->is_sdw) { - if (!hs_jack) { - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - cancel_delayed_work_sync(&rt5682->jack_detect_work); - return 0; - } + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + return 0; + } + + if (!rt5682->is_sdw) { switch (rt5682->pdata.jd_src) { case RT5682_JD1: snd_soc_component_update_bits(component, @@ -1082,7 +1082,8 @@ void rt5682_jack_detect_handler(struct work_struct *work) /* jack was out, report jack type */ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 1); - } else { + } else if ((rt5682->jack_type & SND_JACK_HEADSET) == + SND_JACK_HEADSET) { /* jack is already in, report button event */ rt5682->jack_type = SND_JACK_HEADSET; btn_type = rt5682_button_detect(rt5682->component); @@ -1608,8 +1609,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ @@ -2492,6 +2492,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, RT5682_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2, + RT5682_PWR_VREF2); + usleep_range(55000, 60000); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); @@ -2517,9 +2526,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); if (!rt5682->jack_type) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2 | RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 06ba36595ddd..7cfc89602fc3 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0), /* Boost mixer */ static const struct snd_kcontrol_new wm8974_boost_mixer[] = { -SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1), }; /* Input PGA */ @@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0008; break; case SND_SOC_DAIFMT_DSP_A: + if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF || + (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) { + return -EINVAL; + } iface |= 0x00018; break; default: diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 9ad35d9940fe..97b4f5480a31 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -317,8 +317,8 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (ret < 0) goto out_put_node; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + snd_soc_dai_link_set_capabilities(dai_link); + dai_link->ops = &graph_ops; dai_link->init = asoc_simple_dai_init; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 55e9f8800b3e..04d4d28ed511 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -231,8 +231,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (ret < 0) goto out_put_node; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + snd_soc_dai_link_set_capabilities(dai_link); + dai_link->ops = &simple_ops; dai_link->init = asoc_simple_dai_init; diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 5f96d7ac0a22..bed4d5f73d9c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -354,6 +354,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { { .name = "Codec DSP", .stream_name = "Wake on Voice", + .capture_only = 1, .ops = &bdw_rt5677_dsp_ops, SND_SOC_DAILINK_REG(dsp), }, diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 9e5fc9430628..ecbc58e8a37f 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -543,8 +543,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (cnt) { ret = device_add_properties(codec_dev, props); - if (ret) + if (ret) { + put_device(codec_dev); return ret; + } } devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 7a43c70a1378..22e432768edb 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,21 +253,20 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot + * Default mode for SSP configuration is TDM 4 slot. One board/design, + * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The + * second piggy-backed, output-only codec is inside the keyboard-dock + * (which has extra speakers). Unlike the main rt5672 codec, we cannot + * configure this codec, it is hard coded to use 2 channel 24 bit I2S. + * Since we only support 2 channels anyways, there is no need for TDM + * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. */ - ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_IB_NF | + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) { - dev_err(rtd->dev, "can't set format to TDM %d\n", ret); - return ret; - } - - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; } diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..92f51d0e9fe2 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" - depends on QCOM_APR && HAS_DMA + depends on QCOM_APR select SND_SOC_QDSP6_COMMON select SND_SOC_QDSP6_CORE select SND_SOC_QDSP6_AFE diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f45e5aaa4b30..9539b0d024fe 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -219,19 +219,32 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } +static int rockchip_sound_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, + 8000, 96000); +} + static const struct snd_soc_ops rockchip_sound_max98357a_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_max98357a_hw_params, }; static const struct snd_soc_ops rockchip_sound_rt5514_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_rt5514_hw_params, }; static const struct snd_soc_ops rockchip_sound_da7219_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_da7219_hw_params, }; static const struct snd_soc_ops rockchip_sound_dmic_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_dmic_hw_params, }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0f30f5aabaa8..2b8abf88ec60 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2573,6 +2573,33 @@ int snd_soc_register_component(struct device *dev, EXPORT_SYMBOL_GPL(snd_soc_register_component); /** + * snd_soc_unregister_component_by_driver - Unregister component using a given driver + * from the ASoC core + * + * @dev: The device to unregister + * @component_driver: The component driver to unregister + */ +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver) +{ + struct snd_soc_component *component; + + if (!component_driver) + return; + + mutex_lock(&client_mutex); + component = snd_soc_lookup_component_nolocked(dev, component_driver->name); + if (!component) + goto out; + + snd_soc_del_component_unlocked(component); + +out: + mutex_unlock(&client_mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver); + +/** * snd_soc_unregister_component - Unregister all related component * from the ASoC core * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index b05e18b63a1c..457159975b01 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -391,6 +391,44 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) return stream->channels_min; } +/* + * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs + */ +void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *cpu; + struct snd_soc_dai_link_component *codec; + struct snd_soc_dai *dai; + bool supported[SNDRV_PCM_STREAM_LAST + 1]; + int direction; + int i; + + for_each_pcm_streams(direction) { + supported[direction] = true; + + for_each_link_cpus(dai_link, i, cpu) { + dai = snd_soc_find_dai(cpu); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + if (!supported[direction]) + continue; + for_each_link_codecs(dai_link, i, codec) { + dai = snd_soc_find_dai(codec); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + } + + dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; + dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); + void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action) { diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 11e5d7962370..4534a1c03e8e 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -48,7 +48,9 @@ EXPORT_SYMBOL_GPL(devm_snd_soc_register_dai); static void devm_component_release(struct device *dev, void *res) { - snd_soc_unregister_component(*(struct device **)res); + const struct snd_soc_component_driver **cmpnt_drv = res; + + snd_soc_unregister_component_by_driver(dev, *cmpnt_drv); } /** @@ -65,7 +67,7 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai) { - struct device **ptr; + const struct snd_soc_component_driver **ptr; int ret; ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); @@ -74,7 +76,7 @@ int devm_snd_soc_register_component(struct device *dev, ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); if (ret == 0) { - *ptr = dev; + *ptr = cmpnt_drv; devres_add(dev, ptr); } else { devres_free(ptr); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 80a4e71f2d95..61844403f181 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -478,7 +478,7 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_component_to_pcm(component); - snd_soc_unregister_component(dev); + snd_soc_unregister_component_by_driver(dev, component->driver); dmaengine_pcm_release_chan(pcm); kfree(pcm); } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 43e5745b06aa..6eaa00c21011 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1261,17 +1261,29 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list); ret = soc_tplg_add_route(tplg, routes[i]); - if (ret < 0) + if (ret < 0) { + /* + * this route was added to the list, it will + * be freed in remove_route() so increment the + * counter to skip it in the error handling + * below. + */ + i++; break; + } /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, routes[i], 1); } - /* free memory allocated for all dapm routes in case of error */ - if (ret < 0) - for (i = 0; i < count ; i++) - kfree(routes[i]); + /* + * free memory allocated for all dapm routes not added to the + * list in case of error + */ + if (ret < 0) { + while (i < count) + kfree(routes[i++]); + } /* * free pointer to array of dapm routes as this is no longer needed. @@ -1359,7 +1371,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( if (err < 0) { dev_err(tplg->dev, "ASoC: failed to init %s\n", mc->hdr.name); - soc_tplg_free_tlv(tplg, &kc[i]); goto err_sm; } } @@ -1367,6 +1378,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( err_sm: for (; i >= 0; i--) { + soc_tplg_free_tlv(tplg, &kc[i]); sm = (struct soc_mixer_control *)kc[i].private_value; kfree(sm); kfree(kc[i].name); diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 339c4930b0c0..adc7c37145d6 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev) struct snd_sof_pdata *pdata = sdev->pdata; int ret; - ret = snd_sof_dsp_power_down_notify(sdev); - if (ret < 0) - dev_warn(dev, "error: %d failed to prepare DSP for device removal", - ret); - if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { + ret = snd_sof_dsp_power_down_notify(sdev); + if (ret < 0) + dev_warn(dev, "error: %d failed to prepare DSP for device removal", + ret); + snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 63f9c20a1bac..a4fa8451d8cb 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, static struct snd_soc_dai_driver imx8_dai[] = { { .name = "esai-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index fa86a9e2990f..287114a37688 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev, static struct snd_soc_dai_driver imx8m_dai[] = { { .name = "sai-port", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, }, }; 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