diff options
23 files changed, 182 insertions, 41 deletions
diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml index 81a0215050e0..fe5d938ca310 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8316.yaml +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -49,6 +49,10 @@ properties: items: - const: mclk + interrupts: + maxItems: 1 + description: Headphone detect interrupt + port: $ref: audio-graph-port.yaml# unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml index 3f4f59287c1c..2eb631950963 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml @@ -47,6 +47,12 @@ properties: reg: maxItems: 1 + clocks: + maxItems: 1 + + clock-names: + const: mclk + interrupts: maxItems: 1 description: The CODEC's interrupt output. @@ -98,6 +104,7 @@ properties: - 4 # Use GPIO2 for jack-detect - 5 # Use GPIO3 for jack-detect - 6 # Use GPIO4 for jack-detect + - 7 # Use HDA header for jack-detect realtek,jack-detect-not-inverted: description: @@ -121,6 +128,10 @@ properties: - 2 # Scale current by 1.0 - 3 # Scale current by 1.5 + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - reg diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index 32dea7392e8d..56c755c22945 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -70,6 +70,9 @@ properties: "#sound-dai-cells": const: 0 + port: + $ref: /schemas/graph.yaml#/properties/port + required: - compatible - reg diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 3d4d00188c26..d933cebad52b 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -23,6 +23,7 @@ const struct device_type sdw_slave_type = { .release = sdw_slave_release, .uevent = sdw_slave_uevent, }; +EXPORT_SYMBOL_GPL(sdw_slave_type); int sdw_slave_add(struct sdw_bus *bus, struct sdw_slave_id *id, struct fwnode_handle *fwnode) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 58fd6e84f961..a7860c047503 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1402,7 +1402,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif -void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime); +int snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime); /** * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a82dd155e1d3..b12df5b5ddfc 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1074,7 +1074,9 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) runtime->oss.params = 0; runtime->oss.prepare = 1; runtime->oss.buffer_used = 0; - snd_pcm_runtime_buffer_set_silence(runtime); + err = snd_pcm_runtime_buffer_set_silence(runtime); + if (err < 0) + goto failure; runtime->oss.period_frames = snd_pcm_alsa_frames(substream, oss_period_size); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 68bee40c9ada..932a9bf98cbc 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -730,13 +730,18 @@ static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime) } /* fill the PCM buffer with the current silence format; called from pcm_oss.c */ -void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime) +int snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime) { - snd_pcm_buffer_access_lock(runtime); + int err; + + err = snd_pcm_buffer_access_lock(runtime); + if (err < 0) + return err; if (runtime->dma_area) snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes)); snd_pcm_buffer_access_unlock(runtime); + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_runtime_buffer_set_silence); diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 61c7372e6307..29469e549791 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6613,6 +6613,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a34, "HP Pavilion x360 2-in-1 Laptop 14-ek0xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a4f, "HP Victus 15-fa0xxx (MB 8A4F)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x103c, 0x8a74, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), @@ -6817,6 +6818,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8f42, "HP ZBook 8 G2a 14W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8f57, "HP Trekker G7JC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8f62, "HP ZBook 8 G2a 16W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x1024, "ASUS Zephyrus G14 2025", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1032, "ASUS VivoBook X513EA", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1034, "ASUS GU605C", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), diff --git a/sound/hda/codecs/side-codecs/cirrus_scodec_test.c b/sound/hda/codecs/side-codecs/cirrus_scodec_test.c index 3cca750857b6..dc35932b6b22 100644 --- a/sound/hda/codecs/side-codecs/cirrus_scodec_test.c +++ b/sound/hda/codecs/side-codecs/cirrus_scodec_test.c @@ -103,6 +103,7 @@ static int cirrus_scodec_test_gpio_probe(struct platform_device *pdev) /* GPIO core modifies our struct gpio_chip so use a copy */ gpio_priv->chip = cirrus_scodec_test_gpio_chip; + gpio_priv->chip.parent = &pdev->dev; ret = devm_gpiochip_add_data(&pdev->dev, &gpio_priv->chip, gpio_priv); if (ret) return dev_err_probe(&pdev->dev, ret, "Failed to add gpiochip\n"); @@ -319,7 +320,7 @@ static struct kunit_case cirrus_scodec_test_cases[] = { }; static struct kunit_suite cirrus_scodec_test_suite = { - .name = "snd-hda-scodec-cs35l56-test", + .name = "snd-hda-cirrus-scodec-test", .init = cirrus_scodec_test_case_init, .test_cases = cirrus_scodec_test_cases, }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index f7a7f216d586..624a822341bb 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -2,7 +2,7 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 - 2025 Texas Instruments, Inc. +// Copyright 2023 - 2026 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding@ti.com> // Current maintainer: Baojun Xu <baojun.xu@ti.com> @@ -60,6 +60,7 @@ struct tas2781_hda_i2c_priv { int (*save_calibration)(struct tas2781_hda *h); int hda_chip_id; + bool skip_calibration; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -491,7 +492,8 @@ static void tasdevice_dspfw_init(void *context) /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - hda_priv->save_calibration(tas_hda); + if (!hda_priv->skip_calibration) + hda_priv->save_calibration(tas_hda); } static void tasdev_fw_ready(const struct firmware *fmw, void *context) @@ -548,6 +550,7 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, void *master_data) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; struct hda_component_parent *parent = master_data; struct hda_component *comp; struct hda_codec *codec; @@ -568,11 +571,22 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, case 0x1028: tas_hda->catlog_id = DELL; break; + case 0x103C: + tas_hda->catlog_id = HP; + break; default: tas_hda->catlog_id = LENOVO; break; } + /* + * Using ASUS ROG Xbox Ally X (RC73XA) UEFI calibration data + * causes audio dropouts during playback, use fallback data + * from DSP firmware as a workaround. + */ + if (codec->core.subsystem_id == 0x10431384) + hda_priv->skip_calibration = true; + pm_runtime_get_sync(dev); comp->dev = dev; diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index bf4d9d336561..0294177acc66 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -420,6 +420,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M6500RE"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), DMI_MATCH(DMI_PRODUCT_NAME, "M6501RM"), } }, diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 443cf59cb71a..fdf4a9add852 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -23,7 +23,6 @@ #include "tlv320adcx140.h" struct adcx140_priv { - struct snd_soc_component *component; struct regulator *supply_areg; struct gpio_desc *gpio_reset; struct regmap *regmap; @@ -338,7 +337,7 @@ static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0); static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch = - SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 1); /* Output Mixer */ static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { @@ -699,7 +698,6 @@ static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) { int pwr_ctrl = 0; int ret = 0; - struct snd_soc_component *component = adcx140->component; if (power_state) pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ; @@ -711,7 +709,7 @@ static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) ret = regmap_write(adcx140->regmap, ADCX140_PHASE_CALIB, adcx140->phase_calib_on ? 0x00 : 0x40); if (ret) - dev_err(component->dev, "%s: register write error %d\n", + dev_err(adcx140->dev, "%s: register write error %d\n", __func__, ret); } @@ -727,7 +725,7 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream, struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); u8 data = 0; - switch (params_width(params)) { + switch (params_physical_width(params)) { case 16: data = ADCX140_16_BIT_WORD; break; @@ -742,7 +740,7 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream, break; default: dev_err(component->dev, "%s: Unsupported width %d\n", - __func__, params_width(params)); + __func__, params_physical_width(params)); return -EINVAL; } @@ -1156,6 +1154,9 @@ static int adcx140_i2c_probe(struct i2c_client *i2c) adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) + return dev_err_probe(&i2c->dev, PTR_ERR(adcx140->gpio_reset), + "Failed to get Reset GPIO\n"); + if (!adcx140->gpio_reset) dev_info(&i2c->dev, "Reset GPIO not defined\n"); adcx140->supply_areg = devm_regulator_get_optional(adcx140->dev, diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index d7aca6567c2d..2fc234adca5f 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -678,6 +678,7 @@ struct wsa881x_priv { */ unsigned int sd_n_val; int active_ports; + bool hw_init; bool port_prepared[WSA881X_MAX_SWR_PORTS]; bool port_enable[WSA881X_MAX_SWR_PORTS]; }; @@ -687,6 +688,9 @@ static void wsa881x_init(struct wsa881x_priv *wsa881x) struct regmap *rm = wsa881x->regmap; unsigned int val = 0; + if (wsa881x->hw_init) + return; + regmap_register_patch(wsa881x->regmap, wsa881x_rev_2_0, ARRAY_SIZE(wsa881x_rev_2_0)); @@ -724,6 +728,8 @@ static void wsa881x_init(struct wsa881x_priv *wsa881x) regmap_update_bits(rm, WSA881X_OTP_REG_28, 0x3F, 0x3A); regmap_update_bits(rm, WSA881X_BONGO_RESRV_REG1, 0xFF, 0xB2); regmap_update_bits(rm, WSA881X_BONGO_RESRV_REG2, 0xFF, 0x05); + + wsa881x->hw_init = true; } static int wsa881x_component_probe(struct snd_soc_component *comp) @@ -1067,6 +1073,9 @@ static int wsa881x_update_status(struct sdw_slave *slave, { struct wsa881x_priv *wsa881x = dev_get_drvdata(&slave->dev); + if (status == SDW_SLAVE_UNATTACHED) + wsa881x->hw_init = false; + if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) wsa881x_init(wsa881x); diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index c3046e260cb9..468d2b38a22a 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -475,6 +475,7 @@ struct wsa883x_priv { int active_ports; int dev_mode; int comp_offset; + bool hw_init; /* * Protects temperature reading code (related to speaker protection) and * fields: temperature and pa_on. @@ -1043,6 +1044,9 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x) struct regmap *regmap = wsa883x->regmap; int variant, version, ret; + if (wsa883x->hw_init) + return 0; + ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant); if (ret) return ret; @@ -1054,22 +1058,23 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x) switch (variant) { case WSA8830: - dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8830\n", - version); + dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8830\n", + version); break; case WSA8835: - dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835\n", - version); + dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835\n", + version); break; case WSA8832: - dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8832\n", - version); + dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8832\n", + version); break; case WSA8835_V2: - dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835_V2\n", - version); + dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835_V2\n", + version); break; default: + dev_warn(wsa883x->dev, "unknown variant: %d\n", variant); break; } @@ -1085,6 +1090,8 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x) wsa883x->comp_offset); } + wsa883x->hw_init = true; + return 0; } @@ -1093,6 +1100,9 @@ static int wsa883x_update_status(struct sdw_slave *slave, { struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); + if (status == SDW_SLAVE_UNATTACHED) + wsa883x->hw_init = false; + if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) return wsa883x_init(wsa883x); diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 887edd2be705..6c6b497657d0 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -1534,7 +1534,7 @@ static void wsa884x_init(struct wsa884x_priv *wsa884x) wsa884x_set_gain_parameters(wsa884x); - wsa884x->hw_init = false; + wsa884x->hw_init = true; } static int wsa884x_update_status(struct sdw_slave *slave, @@ -2109,7 +2109,6 @@ static int wsa884x_probe(struct sdw_slave *pdev, /* Start in cache-only until device is enumerated */ regcache_cache_only(wsa884x->regmap, true); - wsa884x->hw_init = true; if (IS_REACHABLE(CONFIG_HWMON)) { struct device *hwmon; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 355f7ec8943c..bdc02e85b089 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1179,9 +1179,9 @@ void graph_util_parse_link_direction(struct device_node *np, bool is_playback_only = of_property_read_bool(np, "playback-only"); bool is_capture_only = of_property_read_bool(np, "capture-only"); - if (playback_only) + if (np && playback_only) *playback_only = is_playback_only; - if (capture_only) + if (np && capture_only) *capture_only = is_capture_only; } EXPORT_SYMBOL_GPL(graph_util_parse_link_direction); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 2c1001148d54..8721a098d53f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -767,6 +767,14 @@ 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, "0DD6") + }, + .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_ptlrvp"), }, .driver_data = (void *)(SOC_SDW_PCH_DMIC), diff --git a/sound/soc/sdw_utils/soc_sdw_cs42l43.c b/sound/soc/sdw_utils/soc_sdw_cs42l43.c index 4c954501e500..2685ff4f0932 100644 --- a/sound/soc/sdw_utils/soc_sdw_cs42l43.c +++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c @@ -44,7 +44,7 @@ static const struct snd_soc_dapm_route cs42l43_dmic_map[] = { static struct snd_soc_jack_pin soc_jack_pins[] = { { .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, + .mask = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, }, { .pin = "Headset Mic", diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index bf382aa07e92..ccf149f949e8 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -841,6 +841,19 @@ struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_part(const u64 adr) } EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_part, "SND_SOC_SDW_UTILS"); +static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_sdw_id(const struct sdw_slave_id *id) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + if (id->part_id == codec_info_list[i].part_id && + (!codec_info_list[i].version_id || + id->sdw_version == codec_info_list[i].version_id)) + return &codec_info_list[i]; + + return NULL; +} + struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id) { int i; @@ -873,22 +886,46 @@ struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, i } EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_dai, "SND_SOC_SDW_UTILS"); +static int asoc_sdw_find_codec_info_dai_index(const struct asoc_sdw_codec_info *codec_info, + const char *dai_name) +{ + int i; + + for (i = 0; i < codec_info->dai_num; i++) { + if (!strcmp(codec_info->dais[i].dai_name, dai_name)) + return i; + } + + return -ENOENT; +} + int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card); struct asoc_sdw_codec_info *codec_info; struct snd_soc_dai *dai; + struct sdw_slave *sdw_peripheral; const char *spk_components=""; int dai_index; int ret; int i; for_each_rtd_codec_dais(rtd, i, dai) { - codec_info = asoc_sdw_find_codec_info_dai(dai->name, &dai_index); + if (is_sdw_slave(dai->component->dev)) + sdw_peripheral = dev_to_sdw_dev(dai->component->dev); + else if (dai->component->dev->parent && is_sdw_slave(dai->component->dev->parent)) + sdw_peripheral = dev_to_sdw_dev(dai->component->dev->parent); + else + continue; + + codec_info = asoc_sdw_find_codec_info_sdw_id(&sdw_peripheral->id); if (!codec_info) return -EINVAL; + dai_index = asoc_sdw_find_codec_info_dai_index(codec_info, dai->name); + WARN_ON(dai_index < 0); + /* * A codec dai can be connected to different dai links for capture and playback, * but we only need to call the rtd_init function once. @@ -898,6 +935,10 @@ int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) if (codec_info->dais[dai_index].rtd_init_done) continue; + dev_dbg(card->dev, "%#x/%s initializing for %s/%s\n", + codec_info->part_id, codec_info->dais[dai_index].dai_name, + dai->component->name, dai->name); + /* * Add card controls and dapm widgets for the first codec dai. * The controls and widgets will be used for all codec dais. diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 624e9269fc25..ba42939d5f01 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -543,11 +543,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, ucontrol->value.bytes.data[0] &= ~params->mask; break; case 2: - ((u16 *)(&ucontrol->value.bytes.data))[0] + ((__be16 *)(&ucontrol->value.bytes.data))[0] &= cpu_to_be16(~params->mask); break; case 4: - ((u32 *)(&ucontrol->value.bytes.data))[0] + ((__be32 *)(&ucontrol->value.bytes.data))[0] &= cpu_to_be32(~params->mask); break; default: diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index 261d9067d27b..e795907a3963 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -2077,7 +2077,7 @@ static const struct regmap_config tegra210_ahub_regmap_config = { .val_bits = 32, .reg_stride = 4, .max_register = TEGRA210_MAX_REGISTER_ADDR, - .cache_type = REGCACHE_FLAT_S, + .cache_type = REGCACHE_FLAT, }; static const struct regmap_config tegra186_ahub_regmap_config = { @@ -2085,7 +2085,7 @@ static const struct regmap_config tegra186_ahub_regmap_config = { .val_bits = 32, .reg_stride = 4, .max_register = TEGRA186_MAX_REGISTER_ADDR, - .cache_type = REGCACHE_FLAT_S, + .cache_type = REGCACHE_FLAT, }; static const struct regmap_config tegra264_ahub_regmap_config = { @@ -2094,7 +2094,7 @@ static const struct regmap_config tegra264_ahub_regmap_config = { .reg_stride = 4, .writeable_reg = tegra264_ahub_wr_reg, .max_register = TEGRA264_MAX_REGISTER_ADDR, - .cache_type = REGCACHE_FLAT_S, + .cache_type = REGCACHE_FLAT, }; static const struct tegra_ahub_soc_data soc_data_tegra210 = { diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 3848766d96c3..ad514c2e5a25 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -194,27 +194,32 @@ static int davinci_evm_probe(struct platform_device *pdev) return -EINVAL; dai->cpus->of_node = of_parse_phandle(np, "ti,mcasp-controller", 0); - if (!dai->cpus->of_node) - return -EINVAL; + if (!dai->cpus->of_node) { + ret = -EINVAL; + goto err_put; + } dai->platforms->of_node = dai->cpus->of_node; evm_soc_card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); if (ret) - return ret; + goto err_put; mclk = devm_clk_get(&pdev->dev, "mclk"); if (PTR_ERR(mclk) == -EPROBE_DEFER) { - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto err_put; } else if (IS_ERR(mclk)) { dev_dbg(&pdev->dev, "mclk not found.\n"); mclk = NULL; } drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); - if (!drvdata) - return -ENOMEM; + if (!drvdata) { + ret = -ENOMEM; + goto err_put; + } drvdata->mclk = mclk; @@ -224,7 +229,8 @@ static int davinci_evm_probe(struct platform_device *pdev) if (!drvdata->mclk) { dev_err(&pdev->dev, "No clock or clock rate defined.\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put; } drvdata->sysclk = clk_get_rate(drvdata->mclk); } else if (drvdata->mclk) { @@ -240,8 +246,25 @@ static int davinci_evm_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&evm_soc_card, drvdata); ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card); - if (ret) + if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto err_put; + } + + return ret; + +err_put: + dai->platforms->of_node = NULL; + + if (dai->cpus->of_node) { + of_node_put(dai->cpus->of_node); + dai->cpus->of_node = NULL; + } + + if (dai->codecs->of_node) { + of_node_put(dai->codecs->of_node); + dai->codecs->of_node = NULL; + } return ret; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 54d01dfd820f..263abb36bb2d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1553,7 +1553,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { counts = snd_usb_endpoint_next_packet_size(ep, ctx, i, avail); - if (counts < 0) + if (counts < 0 || frames + counts >= ep->max_urb_frames) break; /* set up descriptor */ urb->iso_frame_desc[i].offset = frames * stride; |