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[PATCH 1/2] ASoC: ak5558: Add support for ak5552
by Shengjiu Wang 31 Mar '21

31 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> AK5552 is a 32-bit 2ch ADC and has the same register map as AK5558. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak5558.c | 93 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 88 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 85bdd0534180..af12afa83b4f 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -9,6 +9,7 @@ #include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> +#include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -23,6 +24,11 @@ #include "ak5558.h" +enum ak555x_type { + AK5558, + AK5552, +}; + #define AK5558_NUM_SUPPLIES 2 static const char *ak5558_supply_names[AK5558_NUM_SUPPLIES] = { "DVDD", @@ -59,6 +65,15 @@ static const struct soc_enum ak5558_mono_enum[] = { ARRAY_SIZE(mono_texts), mono_texts), }; +static const char * const mono_5552_texts[] = { + "2 Slot", "1 Slot (Fixed)", "2 Slot", "1 Slot (Optimal)", +}; + +static const struct soc_enum ak5552_mono_enum[] = { + SOC_ENUM_SINGLE(AK5558_01_POWER_MANAGEMENT2, 1, + ARRAY_SIZE(mono_5552_texts), mono_5552_texts), +}; + static const char * const digfil_texts[] = { "Sharp Roll-Off", "Show Roll-Off", "Short Delay Sharp Roll-Off", "Short Delay Show Roll-Off", @@ -74,6 +89,11 @@ static const struct snd_kcontrol_new ak5558_snd_controls[] = { SOC_ENUM("AK5558 Digital Filter", ak5558_adcset_enum[0]), }; +static const struct snd_kcontrol_new ak5552_snd_controls[] = { + SOC_ENUM("AK5552 Monaural Mode", ak5552_mono_enum[0]), + SOC_ENUM("AK5552 Digital Filter", ak5558_adcset_enum[0]), +}; + static const struct snd_soc_dapm_widget ak5558_dapm_widgets[] = { /* Analog Input */ SND_SOC_DAPM_INPUT("AIN1"), @@ -97,6 +117,17 @@ static const struct snd_soc_dapm_widget ak5558_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("SDTO", "Capture", 0, SND_SOC_NOPM, 0, 0), }; +static const struct snd_soc_dapm_widget ak5552_dapm_widgets[] = { + /* Analog Input */ + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + + SND_SOC_DAPM_ADC("ADC Ch1", NULL, AK5558_00_POWER_MANAGEMENT1, 0, 0), + SND_SOC_DAPM_ADC("ADC Ch2", NULL, AK5558_00_POWER_MANAGEMENT1, 1, 0), + + SND_SOC_DAPM_AIF_OUT("SDTO", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + static const struct snd_soc_dapm_route ak5558_intercon[] = { {"ADC Ch1", NULL, "AIN1"}, {"SDTO", NULL, "ADC Ch1"}, @@ -123,6 +154,14 @@ static const struct snd_soc_dapm_route ak5558_intercon[] = { {"SDTO", NULL, "ADC Ch8"}, }; +static const struct snd_soc_dapm_route ak5552_intercon[] = { + {"ADC Ch1", NULL, "AIN1"}, + {"SDTO", NULL, "ADC Ch1"}, + + {"ADC Ch2", NULL, "AIN2"}, + {"SDTO", NULL, "ADC Ch2"}, +}; + static int ak5558_set_mcki(struct snd_soc_component *component) { return snd_soc_component_update_bits(component, AK5558_02_CONTROL1, AK5558_CKS, @@ -267,6 +306,18 @@ static struct snd_soc_dai_driver ak5558_dai = { .ops = &ak5558_dai_ops, }; +static struct snd_soc_dai_driver ak5552_dai = { + .name = "ak5558-aif", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = AK5558_FORMATS, + }, + .ops = &ak5558_dai_ops, +}; + static void ak5558_power_off(struct ak5558_priv *ak5558) { if (!ak5558->reset_gpiod) @@ -354,6 +405,21 @@ static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_codec_dev_ak5552 = { + .probe = ak5558_probe, + .remove = ak5558_remove, + .controls = ak5552_snd_controls, + .num_controls = ARRAY_SIZE(ak5552_snd_controls), + .dapm_widgets = ak5552_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5552_dapm_widgets), + .dapm_routes = ak5552_intercon, + .num_dapm_routes = ARRAY_SIZE(ak5552_intercon), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + static const struct regmap_config ak5558_regmap = { .reg_bits = 8, .val_bits = 8, @@ -368,6 +434,7 @@ static int ak5558_i2c_probe(struct i2c_client *i2c) { struct ak5558_priv *ak5558; int ret = 0; + int dev_id; int i; ak5558 = devm_kzalloc(&i2c->dev, sizeof(*ak5558), GFP_KERNEL); @@ -396,11 +463,26 @@ static int ak5558_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_snd_soc_register_component(&i2c->dev, - &soc_codec_dev_ak5558, - &ak5558_dai, 1); - if (ret) + dev_id = (uintptr_t)of_device_get_match_data(&i2c->dev); + switch (dev_id) { + case AK5552: + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_ak5552, + &ak5552_dai, 1); + break; + case AK5558: + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_ak5558, + &ak5558_dai, 1); + break; + default: + dev_err(&i2c->dev, "unexpected device type\n"); + return -EINVAL; + } + if (ret < 0) { + dev_err(&i2c->dev, "failed to register component: %d\n", ret); return ret; + } pm_runtime_enable(&i2c->dev); regcache_cache_only(ak5558->regmap, true); @@ -416,7 +498,8 @@ static int ak5558_i2c_remove(struct i2c_client *i2c) } static const struct of_device_id ak5558_i2c_dt_ids[] __maybe_unused = { - { .compatible = "asahi-kasei,ak5558"}, + { .compatible = "asahi-kasei,ak5558", .data = (void *) AK5558 }, + { .compatible = "asahi-kasei,ak5552", .data = (void *) AK5552 }, { } }; MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids); -- 2.27.0

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[PATCH 0/2] clean up dt_binding_check and dtbs_check warnings of renesas, rsnd.yaml
by Zhen Lei 31 Mar '21

31 Mar '21

When I do dt_binding_check for all YAML files, below warning is reported: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.example.dt.yaml: sound@ec500000: 'dais' is a required property From schema: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml make dt_binding_check -j24 2>err.txt Before: cat err.txt | wc -l 2 After: cat err.txt | wc -l 0 make dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -j24 2>err.txt Before: cat err.txt | wc -l 130 After: cat err.txt | wc -l 0 Zhen Lei (2): ASoC: dt-bindings: renesas, rsnd: Clear warning 'dais' is a required property ASoC: dt-bindings: renesas, rsnd: Clear warning 'ports' does not match any of the regexes Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) -- 1.8.3

3 4

[PATCH v2] soundwire: intel: move to auxiliary bus
by Bard Liao 31 Mar '21

31 Mar '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Now that the auxiliary_bus exists, there's no reason to use platform devices as children of a PCI device any longer. This patch refactors the code by extending a basic auxiliary device with Intel link-specific structures that need to be passed between controller and link levels. This refactoring is much cleaner with no need for cross-pointers between device and link structures. Note that the auxiliary bus API has separate init and add steps, which requires more attention in the error unwinding paths. The main loop needs to deal with kfree() and auxiliary_device_uninit() for the current iteration before jumping to the common label which releases everything allocated in prior iterations. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - add link_dev_register for all kzalloc, device_init, and device_add. --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 56 ++++--- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 228 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 200 insertions(+), 105 deletions(-) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 016e74230bb7..2b7795233282 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -25,6 +25,7 @@ config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE select SOUNDWIRE_GENERIC_ALLOCATION + select AUXILIARY_BUS depends on ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fd95f94630b1..c11e3d8cd308 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -11,7 +11,7 @@ #include <linux/module.h> #include <linux/interrupt.h> #include <linux/io.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <sound/pcm_params.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -1327,11 +1327,14 @@ static int intel_init(struct sdw_intel *sdw) } /* - * probe and init + * probe and init (aux_dev_id argument is required by function prototype but not used) */ -static int intel_master_probe(struct platform_device *pdev) +static int intel_link_probe(struct auxiliary_device *auxdev, + const struct auxiliary_device_id *aux_dev_id) + { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); struct sdw_intel *sdw; struct sdw_cdns *cdns; struct sdw_bus *bus; @@ -1344,14 +1347,14 @@ static int intel_master_probe(struct platform_device *pdev) cdns = &sdw->cdns; bus = &cdns->bus; - sdw->instance = pdev->id; - sdw->link_res = dev_get_platdata(dev); + sdw->instance = auxdev->id; + sdw->link_res = &ldev->link_res; cdns->dev = dev; cdns->registers = sdw->link_res->registers; cdns->instance = sdw->instance; cdns->msg_count = 0; - bus->link_id = pdev->id; + bus->link_id = auxdev->id; sdw_cdns_probe(cdns); @@ -1384,10 +1387,10 @@ static int intel_master_probe(struct platform_device *pdev) return 0; } -int intel_master_startup(struct platform_device *pdev) +int intel_link_startup(struct auxiliary_device *auxdev) { struct sdw_cdns_stream_config config; - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1524,9 +1527,9 @@ int intel_master_startup(struct platform_device *pdev) return ret; } -static int intel_master_remove(struct platform_device *pdev) +static void intel_link_remove(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1542,19 +1545,17 @@ static int intel_master_remove(struct platform_device *pdev) snd_soc_unregister_component(dev); } sdw_bus_master_delete(bus); - - return 0; } -int intel_master_process_wakeen_event(struct platform_device *pdev) +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_intel *sdw; struct sdw_bus *bus; void __iomem *shim; u16 wake_sts; - sdw = platform_get_drvdata(pdev); + sdw = dev_get_drvdata(dev); bus = &sdw->cdns.bus; if (bus->prop.hw_disabled) { @@ -1976,17 +1977,22 @@ static const struct dev_pm_ops intel_pm = { SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) }; -static struct platform_driver sdw_intel_drv = { - .probe = intel_master_probe, - .remove = intel_master_remove, +static const struct auxiliary_device_id intel_link_id_table[] = { + { .name = "soundwire_intel.link" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, intel_link_id_table); + +static struct auxiliary_driver sdw_intel_drv = { + .probe = intel_link_probe, + .remove = intel_link_remove, .driver = { - .name = "intel-sdw", + /* auxiliary_driver_register() sets .name to be the modname */ .pm = &intel_pm, - } + }, + .id_table = intel_link_id_table }; - -module_platform_driver(sdw_intel_drv); +module_auxiliary_driver(sdw_intel_drv); MODULE_LICENSE("Dual BSD/GPL"); -MODULE_ALIAS("platform:intel-sdw"); -MODULE_DESCRIPTION("Intel Soundwire Master Driver"); +MODULE_DESCRIPTION("Intel Soundwire Link Driver"); diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 06bac8ba14e9..0b47b148da3f 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -7,7 +7,6 @@ /** * struct sdw_intel_link_res - Soundwire Intel link resource structure, * typically populated by the controller driver. - * @pdev: platform_device * @mmio_base: mmio base of SoundWire registers * @registers: Link IO registers base * @shim: Audio shim pointer @@ -23,7 +22,6 @@ * @list: used to walk-through all masters exposed by the same controller */ struct sdw_intel_link_res { - struct platform_device *pdev; void __iomem *mmio_base; /* not strictly needed, useful for debug */ void __iomem *registers; void __iomem *shim; @@ -48,7 +46,15 @@ struct sdw_intel { #endif }; -int intel_master_startup(struct platform_device *pdev); -int intel_master_process_wakeen_event(struct platform_device *pdev); +int intel_link_startup(struct auxiliary_device *auxdev); +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev); + +struct sdw_intel_link_dev { + struct auxiliary_device auxdev; + struct sdw_intel_link_res link_res; +}; + +#define auxiliary_dev_to_sdw_intel_link_dev(auxiliary_dev) \ + container_of(auxiliary_dev, struct sdw_intel_link_dev, auxdev) #endif /* __SDW_INTEL_LOCAL_H */ diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 05b726cdfebc..5b32a2ffd376 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -12,7 +12,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/module.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <linux/pm_runtime.h> #include <linux/soundwire/sdw_intel.h> #include "cadence_master.h" @@ -24,28 +24,108 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 +static void intel_link_dev_release(struct device *dev) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); + + kfree(ldev); +} + +/* alloc, init and add link devices */ +static struct sdw_intel_link_dev *intel_link_dev_register(struct sdw_intel_res *res, + struct sdw_intel_ctx *ctx, + struct fwnode_handle *fwnode, + const char *name, + int link_id) +{ + struct sdw_intel_link_dev *ldev; + struct sdw_intel_link_res *link; + struct auxiliary_device *auxdev; + int ret; + + ldev = kzalloc(sizeof(*ldev), GFP_KERNEL); + if (!ldev) + return ERR_PTR(-ENOMEM); + + auxdev = &ldev->auxdev; + auxdev->name = name; + auxdev->dev.parent = res->parent; + auxdev->dev.fwnode = fwnode; + auxdev->dev.release = intel_link_dev_release; + + /* we don't use an IDA since we already have a link ID */ + auxdev->id = link_id; + + /* + * keep a handle on the allocated memory, to be used in all other functions. + * Since the same pattern is used to skip links that are not enabled, there is + * no need to check if ctx->ldev[i] is NULL later on. + */ + ctx->ldev[link_id] = ldev; + + /* Add link information used in the driver probe */ + link = &ldev->link_res; + link->mmio_base = res->mmio_base; + link->registers = res->mmio_base + SDW_LINK_BASE + + (SDW_LINK_SIZE * link_id); + link->shim = res->mmio_base + SDW_SHIM_BASE; + link->alh = res->mmio_base + SDW_ALH_BASE; + + link->ops = res->ops; + link->dev = res->dev; + + link->clock_stop_quirks = res->clock_stop_quirks; + link->shim_lock = &ctx->shim_lock; + link->shim_mask = &ctx->shim_mask; + link->link_mask = ctx->link_mask; + + /* now follow the two-step init/add sequence */ + ret = auxiliary_device_init(auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to initialize link dev %s link_id %d\n", + name, link_id); + kfree(ldev); + return ERR_PTR(ret); + } + + ret = auxiliary_device_add(&ldev->auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to add link dev %s link_id %d\n", + ldev->auxdev.name, link_id); + /* ldev will be freed with the put_device() and .release sequence */ + auxiliary_device_uninit(&ldev->auxdev); + return ERR_PTR(ret); + } + + return ldev; +} + +static void intel_link_dev_unregister(struct sdw_intel_link_dev *ldev) +{ + auxiliary_device_delete(&ldev->auxdev); + auxiliary_device_uninit(&ldev->auxdev); +} + static int sdw_intel_cleanup(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link = ctx->links; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!link) - return 0; - link_mask = ctx->link_mask; - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - if (link->pdev) { - pm_runtime_disable(&link->pdev->dev); - platform_device_unregister(link->pdev); - } + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) - pm_runtime_put_noidle(link->dev); + pm_runtime_disable(&ldev->auxdev.dev); + intel_link_dev_unregister(ldev); + + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); } return 0; @@ -91,9 +171,8 @@ EXPORT_SYMBOL_NS(sdw_intel_thread, SOUNDWIRE_INTEL_INIT); static struct sdw_intel_ctx *sdw_intel_probe_controller(struct sdw_intel_res *res) { - struct platform_device_info pdevinfo; - struct platform_device *pdev; struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; struct sdw_intel_ctx *ctx; struct acpi_device *adev; struct sdw_slave *slave; @@ -116,67 +195,60 @@ static struct sdw_intel_ctx count = res->count; dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - ctx = devm_kzalloc(&adev->dev, sizeof(*ctx), GFP_KERNEL); + /* + * we need to alloc/free memory manually and can't use devm: + * this routine may be called from a workqueue, and not from + * the parent .probe. + * If devm_ was used, the memory might never be freed on errors. + */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return NULL; ctx->count = count; - ctx->links = devm_kcalloc(&adev->dev, ctx->count, - sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) + + /* + * allocate the array of pointers. The link-specific data is allocated + * as part of the first loop below and released with the auxiliary_device_uninit(). + * If some links are disabled, the link pointer will remain NULL. Given that the + * number of links is small, this is simpler than using a list to keep track of links. + */ + ctx->ldev = kcalloc(ctx->count, sizeof(*ctx->ldev), GFP_KERNEL); + if (!ctx->ldev) { + kfree(ctx); return NULL; + } - ctx->count = count; ctx->mmio_base = res->mmio_base; ctx->link_mask = res->link_mask; ctx->handle = res->handle; mutex_init(&ctx->shim_lock); - link = ctx->links; link_mask = ctx->link_mask; INIT_LIST_HEAD(&ctx->link_list); - /* Create SDW Master devices */ - for (i = 0; i < count; i++, link++) { - if (!(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); + for (i = 0; i < count; i++) { + if (!(link_mask & BIT(i))) continue; - } - link->mmio_base = res->mmio_base; - link->registers = res->mmio_base + SDW_LINK_BASE - + (SDW_LINK_SIZE * i); - link->shim = res->mmio_base + SDW_SHIM_BASE; - link->alh = res->mmio_base + SDW_ALH_BASE; - - link->ops = res->ops; - link->dev = res->dev; - - link->clock_stop_quirks = res->clock_stop_quirks; - link->shim_lock = &ctx->shim_lock; - link->shim_mask = &ctx->shim_mask; - link->link_mask = link_mask; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "intel-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - pdevinfo.data = link; - pdevinfo.size_data = sizeof(*link); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); + /* + * init and add a device for each link + * + * The name of the device will be soundwire_intel.link.[i], + * with the "soundwire_intel" module prefix automatically added + * by the auxiliary bus core. + */ + ldev = intel_link_dev_register(res, + ctx, + acpi_fwnode_handle(adev), + "link", + i); + if (IS_ERR(ldev)) goto err; - } - link->pdev = pdev; - link->cdns = platform_get_drvdata(pdev); + + link = &ldev->link_res; + link->cdns = dev_get_drvdata(&ldev->auxdev.dev); list_add_tail(&link->list, &ctx->link_list); bus = &link->cdns->bus; @@ -185,8 +257,7 @@ static struct sdw_intel_ctx num_slaves++; } - ctx->ids = devm_kcalloc(&adev->dev, num_slaves, - sizeof(*ctx->ids), GFP_KERNEL); + ctx->ids = kcalloc(num_slaves, sizeof(*ctx->ids), GFP_KERNEL); if (!ctx->ids) goto err; @@ -204,8 +275,14 @@ static struct sdw_intel_ctx return ctx; err: - ctx->count = i; - sdw_intel_cleanup(ctx); + while (i--) { + if (!(link_mask & BIT(i))) + continue; + ldev = ctx->ldev[i]; + intel_link_dev_unregister(ldev); + } + kfree(ctx->ldev); + kfree(ctx); return NULL; } @@ -213,7 +290,7 @@ static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { struct acpi_device *adev; - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; @@ -232,27 +309,28 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) return -EINVAL; } - if (!ctx->links) + if (!ctx->ldev) return -EINVAL; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_startup(link->pdev); + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) { + intel_link_startup(&ldev->auxdev); + + if (!ldev->link_res.clock_stop_quirks) { /* * we need to prevent the parent PCI device * from entering pm_runtime suspend, so that * power rails to the SoundWire IP are not * turned off. */ - pm_runtime_get_noresume(link->dev); + pm_runtime_get_noresume(ldev->link_res.dev); } } @@ -297,27 +375,31 @@ EXPORT_SYMBOL_NS(sdw_intel_startup, SOUNDWIRE_INTEL_INIT); void sdw_intel_exit(struct sdw_intel_ctx *ctx) { sdw_intel_cleanup(ctx); + kfree(ctx->ids); + kfree(ctx->ldev); + kfree(ctx); } EXPORT_SYMBOL_NS(sdw_intel_exit, SOUNDWIRE_INTEL_INIT); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!ctx->links) + if (!ctx->ldev) return; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_process_wakeen_event(link->pdev); + ldev = ctx->ldev[i]; + + intel_link_process_wakeen_event(&ldev->auxdev); } } EXPORT_SYMBOL_NS(sdw_intel_process_wakeen_event, SOUNDWIRE_INTEL_INIT); diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3a5446ac014a..1ebea7764011 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -58,7 +58,7 @@ struct sdw_intel_acpi_info { u32 link_mask; }; -struct sdw_intel_link_res; +struct sdw_intel_link_dev; /* Intel clock-stop/pm_runtime quirk definitions */ @@ -109,7 +109,7 @@ struct sdw_intel_slave_id { * Controller * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle - * @links: information for each link (controller-specific and kept + * @ldev: information for each link (controller-specific and kept * opaque here) * @ids: array of slave_id, representing Slaves exposed across all enabled * links @@ -123,7 +123,7 @@ struct sdw_intel_ctx { u32 link_mask; int num_slaves; acpi_handle handle; - struct sdw_intel_link_res *links; + struct sdw_intel_link_dev **ldev; struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ -- 2.17.1

1 0

[PATCH v6 0/9] soundwire: qcom: various improvements
by Srinivas Kandagatla 30 Mar '21

30 Mar '21

Thanks for reviewing v5 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Patch to add Auto Enumeration supported by the controller is also part of this series. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v5: - Use BIT macro in interrupt routine - add comment for fifo write Srinivas Kandagatla (9): dt-bindings: soundwire: qcom: clarify data port bus parameters soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts soundwire: export sdw_compare_devid, sdw_extract_slave_id and sdw_slave_add soundwire: qcom: add auto enumeration support soundwire: qcom: wait for enumeration to be complete in probe .../bindings/soundwire/qcom,sdw.txt | 20 + drivers/soundwire/bus.c | 4 +- drivers/soundwire/qcom.c | 529 ++++++++++++++---- drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 2 + 5 files changed, 443 insertions(+), 113 deletions(-) -- 2.21.0

2 10

[PATCH][next] ASoC: mediatek: mt6359: Fix spelling mistake "reate" -> "create"
by Colin King 30 Mar '21

30 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err error message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/mt6359-accdet.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index 0bef6eaad5ad..4222aed013f1 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -1019,7 +1019,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) priv->jd_workqueue = create_singlethread_workqueue("mt6359_accdet_jd"); INIT_WORK(&priv->jd_work, mt6359_accdet_jd_work); if (!priv->jd_workqueue) { - dev_err(&pdev->dev, "Failed to reate jack detect workqueue\n"); + dev_err(&pdev->dev, "Failed to create jack detect workqueue\n"); ret = -1; goto err_eint_wq; } -- 2.30.2

2 1

[PATCH] ASoC: q6afe-clocks: fix reprobing of the driver
by Dmitry Baryshkov 30 Mar '21

30 Mar '21

Q6afe-clocks driver can get reprobed. For example if the APR services are restarted after the firmware crash. However currently Q6afe-clocks driver will oops because hw.init will get cleared during first _probe call. Rewrite the driver to fill the clock data at runtime rather than using big static array of clocks. Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-clocks.c | 209 ++++++++++++++-------------- sound/soc/qcom/qdsp6/q6afe.c | 2 +- sound/soc/qcom/qdsp6/q6afe.h | 2 +- 3 files changed, 108 insertions(+), 105 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-clocks.c b/sound/soc/qcom/qdsp6/q6afe-clocks.c index f0362f061652..9431656283cd 100644 --- a/sound/soc/qcom/qdsp6/q6afe-clocks.c +++ b/sound/soc/qcom/qdsp6/q6afe-clocks.c @@ -11,33 +11,29 @@ #include <linux/slab.h> #include "q6afe.h" -#define Q6AFE_CLK(id) &(struct q6afe_clk) { \ +#define Q6AFE_CLK(id) { \ .clk_id = id, \ .afe_clk_id = Q6AFE_##id, \ .name = #id, \ - .attributes = LPASS_CLK_ATTRIBUTE_COUPLE_NO, \ .rate = 19200000, \ - .hw.init = &(struct clk_init_data) { \ - .ops = &clk_q6afe_ops, \ - .name = #id, \ - }, \ } -#define Q6AFE_VOTE_CLK(id, blkid, n) &(struct q6afe_clk) { \ +#define Q6AFE_VOTE_CLK(id, blkid, n) { \ .clk_id = id, \ .afe_clk_id = blkid, \ - .name = #n, \ - .hw.init = &(struct clk_init_data) { \ - .ops = &clk_vote_q6afe_ops, \ - .name = #id, \ - }, \ + .name = n, \ } -struct q6afe_clk { - struct device *dev; +struct q6afe_clk_init { int clk_id; int afe_clk_id; char *name; + int rate; +}; + +struct q6afe_clk { + struct device *dev; + int afe_clk_id; int attributes; int rate; uint32_t handle; @@ -48,8 +44,7 @@ struct q6afe_clk { struct q6afe_cc { struct device *dev; - struct q6afe_clk **clks; - int num_clks; + struct q6afe_clk *clks[Q6AFE_MAX_CLK_ID]; }; static int clk_q6afe_prepare(struct clk_hw *hw) @@ -105,7 +100,7 @@ static int clk_vote_q6afe_block(struct clk_hw *hw) struct q6afe_clk *clk = to_q6afe_clk(hw); return q6afe_vote_lpass_core_hw(clk->dev, clk->afe_clk_id, - clk->name, &clk->handle); + clk_hw_get_name(&clk->hw), &clk->handle); } static void clk_unvote_q6afe_block(struct clk_hw *hw) @@ -120,84 +115,76 @@ static const struct clk_ops clk_vote_q6afe_ops = { .unprepare = clk_unvote_q6afe_block, }; -static struct q6afe_clk *q6afe_clks[Q6AFE_MAX_CLK_ID] = { - [LPASS_CLK_ID_PRI_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_IBIT), - [LPASS_CLK_ID_PRI_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_EBIT), - [LPASS_CLK_ID_SEC_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_MI2S_IBIT), - [LPASS_CLK_ID_SEC_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_MI2S_EBIT), - [LPASS_CLK_ID_TER_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_TER_MI2S_IBIT), - [LPASS_CLK_ID_TER_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_TER_MI2S_EBIT), - [LPASS_CLK_ID_QUAD_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUAD_MI2S_IBIT), - [LPASS_CLK_ID_QUAD_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUAD_MI2S_EBIT), - [LPASS_CLK_ID_SPEAKER_I2S_IBIT] = - Q6AFE_CLK(LPASS_CLK_ID_SPEAKER_I2S_IBIT), - [LPASS_CLK_ID_SPEAKER_I2S_EBIT] = - Q6AFE_CLK(LPASS_CLK_ID_SPEAKER_I2S_EBIT), - [LPASS_CLK_ID_SPEAKER_I2S_OSR] = - Q6AFE_CLK(LPASS_CLK_ID_SPEAKER_I2S_OSR), - [LPASS_CLK_ID_QUI_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUI_MI2S_IBIT), - [LPASS_CLK_ID_QUI_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUI_MI2S_EBIT), - [LPASS_CLK_ID_SEN_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEN_MI2S_IBIT), - [LPASS_CLK_ID_SEN_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEN_MI2S_EBIT), - [LPASS_CLK_ID_INT0_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT0_MI2S_IBIT), - [LPASS_CLK_ID_INT1_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT1_MI2S_IBIT), - [LPASS_CLK_ID_INT2_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT2_MI2S_IBIT), - [LPASS_CLK_ID_INT3_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT3_MI2S_IBIT), - [LPASS_CLK_ID_INT4_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT4_MI2S_IBIT), - [LPASS_CLK_ID_INT5_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT5_MI2S_IBIT), - [LPASS_CLK_ID_INT6_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_INT6_MI2S_IBIT), - [LPASS_CLK_ID_QUI_MI2S_OSR] = Q6AFE_CLK(LPASS_CLK_ID_QUI_MI2S_OSR), - [LPASS_CLK_ID_PRI_PCM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_PCM_IBIT), - [LPASS_CLK_ID_PRI_PCM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_PCM_EBIT), - [LPASS_CLK_ID_SEC_PCM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_PCM_IBIT), - [LPASS_CLK_ID_SEC_PCM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_PCM_EBIT), - [LPASS_CLK_ID_TER_PCM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_TER_PCM_IBIT), - [LPASS_CLK_ID_TER_PCM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_TER_PCM_EBIT), - [LPASS_CLK_ID_QUAD_PCM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUAD_PCM_IBIT), - [LPASS_CLK_ID_QUAD_PCM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUAD_PCM_EBIT), - [LPASS_CLK_ID_QUIN_PCM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUIN_PCM_IBIT), - [LPASS_CLK_ID_QUIN_PCM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUIN_PCM_EBIT), - [LPASS_CLK_ID_QUI_PCM_OSR] = Q6AFE_CLK(LPASS_CLK_ID_QUI_PCM_OSR), - [LPASS_CLK_ID_PRI_TDM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_TDM_IBIT), - [LPASS_CLK_ID_PRI_TDM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_TDM_EBIT), - [LPASS_CLK_ID_SEC_TDM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_TDM_IBIT), - [LPASS_CLK_ID_SEC_TDM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_TDM_EBIT), - [LPASS_CLK_ID_TER_TDM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_TER_TDM_IBIT), - [LPASS_CLK_ID_TER_TDM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_TER_TDM_EBIT), - [LPASS_CLK_ID_QUAD_TDM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUAD_TDM_IBIT), - [LPASS_CLK_ID_QUAD_TDM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUAD_TDM_EBIT), - [LPASS_CLK_ID_QUIN_TDM_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUIN_TDM_IBIT), - [LPASS_CLK_ID_QUIN_TDM_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_QUIN_TDM_EBIT), - [LPASS_CLK_ID_QUIN_TDM_OSR] = Q6AFE_CLK(LPASS_CLK_ID_QUIN_TDM_OSR), - [LPASS_CLK_ID_MCLK_1] = Q6AFE_CLK(LPASS_CLK_ID_MCLK_1), - [LPASS_CLK_ID_MCLK_2] = Q6AFE_CLK(LPASS_CLK_ID_MCLK_2), - [LPASS_CLK_ID_MCLK_3] = Q6AFE_CLK(LPASS_CLK_ID_MCLK_3), - [LPASS_CLK_ID_MCLK_4] = Q6AFE_CLK(LPASS_CLK_ID_MCLK_4), - [LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE] = - Q6AFE_CLK(LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE), - [LPASS_CLK_ID_INT_MCLK_0] = Q6AFE_CLK(LPASS_CLK_ID_INT_MCLK_0), - [LPASS_CLK_ID_INT_MCLK_1] = Q6AFE_CLK(LPASS_CLK_ID_INT_MCLK_1), - [LPASS_CLK_ID_WSA_CORE_MCLK] = Q6AFE_CLK(LPASS_CLK_ID_WSA_CORE_MCLK), - [LPASS_CLK_ID_WSA_CORE_NPL_MCLK] = - Q6AFE_CLK(LPASS_CLK_ID_WSA_CORE_NPL_MCLK), - [LPASS_CLK_ID_VA_CORE_MCLK] = Q6AFE_CLK(LPASS_CLK_ID_VA_CORE_MCLK), - [LPASS_CLK_ID_TX_CORE_MCLK] = Q6AFE_CLK(LPASS_CLK_ID_TX_CORE_MCLK), - [LPASS_CLK_ID_TX_CORE_NPL_MCLK] = - Q6AFE_CLK(LPASS_CLK_ID_TX_CORE_NPL_MCLK), - [LPASS_CLK_ID_RX_CORE_MCLK] = Q6AFE_CLK(LPASS_CLK_ID_RX_CORE_MCLK), - [LPASS_CLK_ID_RX_CORE_NPL_MCLK] = - Q6AFE_CLK(LPASS_CLK_ID_RX_CORE_NPL_MCLK), - [LPASS_CLK_ID_VA_CORE_2X_MCLK] = - Q6AFE_CLK(LPASS_CLK_ID_VA_CORE_2X_MCLK), - [LPASS_HW_AVTIMER_VOTE] = Q6AFE_VOTE_CLK(LPASS_HW_AVTIMER_VOTE, - Q6AFE_LPASS_CORE_AVTIMER_BLOCK, - "LPASS_AVTIMER_MACRO"), - [LPASS_HW_MACRO_VOTE] = Q6AFE_VOTE_CLK(LPASS_HW_MACRO_VOTE, - Q6AFE_LPASS_CORE_HW_MACRO_BLOCK, - "LPASS_HW_MACRO"), - [LPASS_HW_DCODEC_VOTE] = Q6AFE_VOTE_CLK(LPASS_HW_DCODEC_VOTE, - Q6AFE_LPASS_CORE_HW_DCODEC_BLOCK, - "LPASS_HW_DCODEC"), +static const struct q6afe_clk_init q6afe_clks[] = { + Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEC_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEC_MI2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_TER_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_TER_MI2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUAD_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUAD_MI2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_SPEAKER_I2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_SPEAKER_I2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_SPEAKER_I2S_OSR), + Q6AFE_CLK(LPASS_CLK_ID_QUI_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUI_MI2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEN_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEN_MI2S_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT0_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT1_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT2_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT3_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT4_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT5_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_INT6_MI2S_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUI_MI2S_OSR), + Q6AFE_CLK(LPASS_CLK_ID_PRI_PCM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_PRI_PCM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEC_PCM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEC_PCM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_TER_PCM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_TER_PCM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUAD_PCM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUAD_PCM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUIN_PCM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUIN_PCM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUI_PCM_OSR), + Q6AFE_CLK(LPASS_CLK_ID_PRI_TDM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_PRI_TDM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEC_TDM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_SEC_TDM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_TER_TDM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_TER_TDM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUAD_TDM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUAD_TDM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUIN_TDM_IBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUIN_TDM_EBIT), + Q6AFE_CLK(LPASS_CLK_ID_QUIN_TDM_OSR), + Q6AFE_CLK(LPASS_CLK_ID_MCLK_1), + Q6AFE_CLK(LPASS_CLK_ID_MCLK_2), + Q6AFE_CLK(LPASS_CLK_ID_MCLK_3), + Q6AFE_CLK(LPASS_CLK_ID_MCLK_4), + Q6AFE_CLK(LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE), + Q6AFE_CLK(LPASS_CLK_ID_INT_MCLK_0), + Q6AFE_CLK(LPASS_CLK_ID_INT_MCLK_1), + Q6AFE_CLK(LPASS_CLK_ID_WSA_CORE_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_WSA_CORE_NPL_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_VA_CORE_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_TX_CORE_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_TX_CORE_NPL_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_RX_CORE_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_RX_CORE_NPL_MCLK), + Q6AFE_CLK(LPASS_CLK_ID_VA_CORE_2X_MCLK), + Q6AFE_VOTE_CLK(LPASS_HW_AVTIMER_VOTE, + Q6AFE_LPASS_CORE_AVTIMER_BLOCK, + "LPASS_AVTIMER_MACRO"), + Q6AFE_VOTE_CLK(LPASS_HW_MACRO_VOTE, + Q6AFE_LPASS_CORE_HW_MACRO_BLOCK, + "LPASS_HW_MACRO"), + Q6AFE_VOTE_CLK(LPASS_HW_DCODEC_VOTE, + Q6AFE_LPASS_CORE_HW_DCODEC_BLOCK, + "LPASS_HW_DCODEC"), }; static struct clk_hw *q6afe_of_clk_hw_get(struct of_phandle_args *clkspec, @@ -207,7 +194,7 @@ static struct clk_hw *q6afe_of_clk_hw_get(struct of_phandle_args *clkspec, unsigned int idx = clkspec->args[0]; unsigned int attr = clkspec->args[1]; - if (idx >= cc->num_clks || attr > LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR) { + if (idx >= Q6AFE_MAX_CLK_ID || attr > LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR) { dev_err(cc->dev, "Invalid clk specifier (%d, %d)\n", idx, attr); return ERR_PTR(-EINVAL); } @@ -230,20 +217,36 @@ static int q6afe_clock_dev_probe(struct platform_device *pdev) if (!cc) return -ENOMEM; - cc->clks = &q6afe_clks[0]; - cc->num_clks = ARRAY_SIZE(q6afe_clks); + cc->dev = dev; for (i = 0; i < ARRAY_SIZE(q6afe_clks); i++) { - if (!q6afe_clks[i]) - continue; + unsigned int id = q6afe_clks[i].clk_id; + struct clk_init_data init = { + .name = q6afe_clks[i].name, + }; + struct q6afe_clk *clk; + + clk = devm_kzalloc(dev, sizeof(*clk), GFP_KERNEL); + if (!clk) + return -ENOMEM; + + clk->dev = dev; + clk->afe_clk_id = q6afe_clks[i].afe_clk_id; + clk->rate = q6afe_clks[i].rate; + clk->hw.init = &init; + + if (clk->rate) + init.ops = &clk_q6afe_ops; + else + init.ops = &clk_vote_q6afe_ops; - q6afe_clks[i]->dev = dev; + cc->clks[id] = clk; - ret = devm_clk_hw_register(dev, &q6afe_clks[i]->hw); + ret = devm_clk_hw_register(dev, &clk->hw); if (ret) return ret; } - ret = of_clk_add_hw_provider(dev->of_node, q6afe_of_clk_hw_get, cc); + ret = devm_of_clk_add_hw_provider(dev, q6afe_of_clk_hw_get, cc); if (ret) return ret; diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index cad1cd1bfdf0..4327b72162ec 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1681,7 +1681,7 @@ int q6afe_unvote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, EXPORT_SYMBOL(q6afe_unvote_lpass_core_hw); int q6afe_vote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, - char *client_name, uint32_t *client_handle) + const char *client_name, uint32_t *client_handle) { struct q6afe *afe = dev_get_drvdata(dev->parent); struct afe_cmd_remote_lpass_core_hw_vote_request *vote_cfg; diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 22e10269aa10..3845b56c0ed3 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -236,7 +236,7 @@ int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, int q6afe_set_lpass_clock(struct device *dev, int clk_id, int clk_src, int clk_root, unsigned int freq); int q6afe_vote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, - char *client_name, uint32_t *client_handle); + const char *client_name, uint32_t *client_handle); int q6afe_unvote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, uint32_t client_handle); #endif /* __Q6AFE_H__ */ -- 2.30.2

4 4

[PATCH v2] ASoC: dt-bindings: nvidia, tegra210-ahub: Add missing child nodes
by Rob Herring 30 Mar '21

30 Mar '21

The nvidia,tegra210-ahub binding is missing schema for child nodes. This results in warnings if 'additionalProperties: false' is set (or when the tools implement 'unevaluatedProperties' support). Add the child nodes and reference their schema if one exists. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Jonathan Hunter <jonathanh(a)nvidia.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-tegra(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Also add 'dspk' child node This patch ideally should be applied before this series[1]. [1] https://lore.kernel.org/r/20210323163634.877511-1-robh@kernel.org/ --- .../bindings/sound/nvidia,tegra210-ahub.yaml | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml index e568d6c7dddd..1118a9488345 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml @@ -69,6 +69,22 @@ properties: $ref: audio-graph-port.yaml# unevaluatedProperties: false +patternProperties: + '^i2s@[0-9a-f]+$': + type: object + + '^dmic@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-dmic.yaml# + + '^admaif@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-admaif.yaml# + + '^dspk@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-dspk.yaml# + required: - compatible - reg -- 2.27.0

4 4

[PATCH] ASoC: Intel: add max98390 echo reference support
by mac.chiang＠intel.com 30 Mar '21

30 Mar '21

From: Mac Chiang <mac.chiang(a)intel.com> load new topology file with speaker capture pipeline. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 2161b3b..7f6ef82 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -79,7 +79,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98390_spk_codecs, .sof_fw_filename = "sof-cml.ri", - .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + .sof_tplg_filename = "sof-cml-da7219-max98390.tplg", }, {}, }; -- 2.7.4

3 2

[PATCH v4 0/6] ALSA: control - add generic LED API
by Jaroslav Kysela 30 Mar '21

30 Mar '21

This patchset tries to resolve the diversity in the audio LED control among the ALSA drivers. A new control layer registration is introduced which allows to run additional operations on top of the elementary ALSA sound controls. A new control access group (three bits in the access flags) was introduced to carry the LED group information for the sound controls. The low-level sound drivers can just mark those controls using this access group. This information is not exported to the user space, but user space can manage the LED sound control associations through sysfs (last patch) per Mark's request. It makes things fully configurable in the kernel and user space (UCM). The actual state ('route') evaluation is really easy (the minimal value check for all channels / controls / cards). If there's more complicated logic for a given hardware, the card driver may eventually export a new read-only sound control for the LED group and do the logic itself. The new LED trigger control code is completely separated and possibly optional (there's no symbol dependency). The full code separation allows eventually to move this LED trigger control to the user space in future. Actually it replaces the already present functionality in the kernel space (HDA drivers) and allows a quick adoption for the recent hardware (ASoC codecs including SoundWire). # lsmod | grep snd_ctl_led snd_ctl_led 24576 0 The sound driver implementation is really easy: 1) call snd_ctl_led_request() when control LED layer should be automatically activated / it calls module_request("snd-ctl-led") on demand / 2) mark all related kcontrols with SNDRV_CTL_ELEM_ACCESS_SPK_LED or SNDRV_CTL_ELEM_ACCESS_MIC_LED v4 changes: - the LED access flags are private to kernel now (no user space API change) - fixes (kctl management, sysfs cleanup) - add the sysfs LED marking kcontrol management - https://lore.kernel.org/alsa-devel/28fffebd-1ce9-7480-0f2f-ed8369abddf1@per… v3 changes: - reorder the controls_rwsem use to fix the remaining mutex issue card->controls_rwsem <-> snd_ctl_layer_rwsem v2 changes: - fix the locking - remove the controls_rwsem read lock in the element get (the consistency is already protected with the global snd_ctl_led_mutex and possible partial value writes are catched with the next value change notification callback) - rename state to brightness and show the brightness unsigned integer value instead the text on/off string (sync with the LED core routines) - remove snd_ctl_led_hello() function (CI warning) - make snd_ctl_led_get_by_access() function static (CI warning) - move snd_ctl_layer_rwsem lock before the registraction callback call in snd_ctl_register_layer() - optimization v1: - https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ Original RFC: - https://lore.kernel.org/alsa-devel/20210207201157.869972-1-perex@perex.cz/ Cc: Hans de Goede <hdegoede(a)redhat.com> Jaroslav Kysela (6): ALSA: control - introduce snd_ctl_notify_one() helper ALSA: control - add layer registration routines ALSA: control - add generic LED trigger module as the new control layer ALSA: HDA - remove the custom implementation for the audio LED trigger ALSA: control - add sysfs support to the LED trigger module ALSA: led control - add sysfs kcontrol LED marking layer include/sound/control.h | 35 +- sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/control.c | 182 ++++++-- sound/core/control_led.c | 770 ++++++++++++++++++++++++++++++++ sound/pci/hda/Kconfig | 4 +- sound/pci/hda/hda_codec.c | 69 +-- sound/pci/hda/hda_generic.c | 162 ++----- sound/pci/hda/hda_generic.h | 15 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/patch_ca0132.c | 4 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +- sound/pci/hda/thinkpad_helper.c | 2 +- 14 files changed, 1011 insertions(+), 264 deletions(-) create mode 100644 sound/core/control_led.c -- 2.29.2

5 29

Re: [PATCH v2] ASoC: rt1019: add rt1019 amplifier driver
by Rob Herring 30 Mar '21

30 Mar '21

On Mon, Mar 29, 2021 at 9:54 PM Jack Yu <jack.yu(a)realtek.com> wrote: > > > > -----Original Message----- > > From: Mark Brown <broonie(a)kernel.org> > > Sent: Monday, March 29, 2021 11:37 PM > > To: Jack Yu <jack.yu(a)realtek.com> > > Cc: Rob Herring <robh(a)kernel.org>; lgirdwood(a)gmail.com; Oder Chiou > > <oder_chiou(a)realtek.com>; alsa-devel(a)alsa-project.org; lars(a)metafoo.de; > > kent_chen(a)realtek.com [陳建宏] <kent_chen(a)realtek.com>; 陳昶志 > > <kenny_chen(a)realtek.com>; Derek [方德義] <derek.fang(a)realtek.com>; > > Shuming [范書銘] <shumingf(a)realtek.com>; Flove(HsinFu) > > <flove(a)realtek.com> > > Subject: Re: [PATCH v2] ASoC: rt1019: add rt1019 amplifier driver > > > > On Mon, Mar 29, 2021 at 08:09:26AM +0000, Jack Yu wrote: > > > > > > Bindings should also be sent to the DT list. get_maintainer.pl will tell you > > this. > > > > > 1. Does it mean I should add my name in MAINTAINERS? > > > > It means you should have sent the patch with the binding changes to the DT list > > and maintainers as well as everyone else. > > > > > 2. Is below warning message caused by #1 question? > > > > No, it's the content of the change that's introduced problems in the binding. > > Hi Mark, > > I'll add below "address-cells" and "size-cells " in to rt1019.yaml like below > > examples: > - | > i2c { > + #address-cells = <1>; > + #size-cells = <0>; > rt1019: codec@28 { > compatible = "realtek,rt1019"; > reg = <0x28>; > }; > }; > > but I'm not sure how to fix some of below warnings, like 'reg_format' and 'i2c_bus_bridge' and 'avoid_default_addr_size', > are these caused by lack of "address-cells" and "size-cells "? Yes, but run 'make dt_binding_check' to make sure. Rob

1 0

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP 640 G8
by Jeremy Szu 30 Mar '21

30 Mar '21

The HP EliteBook 640 G8 Notebook PC is using ALC236 codec which is using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 316b9b4ccb32..9d08b452e9ae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8057,6 +8057,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), -- 2.30.1

2 1

[PATCH] ALSA: core: remove redundant spin_lock pair in snd_card_disconnect
by root 30 Mar '21

30 Mar '21

RnJvbTogSmlhIFpob3UgPHpob3UuamlhMkB6dGUuY29tLmNuPg0KDQptb2RpZmljYXRpb24gaW4g Y29tbWl0IDJhM2Y3MjIxYWNkZCAoIkFMU0E6IGNvcmU6IEZpeCBjYXJkIHJhY2VzIGJldHdlZW4N CnJlZ2lzdGVyIGFuZCBkaXNjb25uZWN0IikgcmVzdWx0aW5nIGluIHRoaXMgcHJvYmxlbS4NCg0K U2lnbmVkLW9mZi1ieTogSmlhIFpob3UgPHpob3UuamlhMkB6dGUuY29tLmNuPg0KU2lnbmVkLW9m Zi1ieTogWWkgV2FuZyA8d2FuZy55aTU5QHp0ZS5jb20uY24+DQotLS0NCiBzb3VuZC9jb3JlL2lu aXQuYyB8IDIgLS0NCiAxIGZpbGUgY2hhbmdlZCwgMiBkZWxldGlvbnMoLSkNCg0KZGlmZiAtLWdp dCBhL3NvdW5kL2NvcmUvaW5pdC5jIGIvc291bmQvY29yZS9pbml0LmMNCmluZGV4IDAxOGNlNGVm MS4uOWY1MjcwYzkwIDEwMDY0NA0KLS0tIGEvc291bmQvY29yZS9pbml0LmMNCisrKyBiL3NvdW5k L2NvcmUvaW5pdC5jDQpAQCAtMzkwLDEwICszOTAsOCBAQCBpbnQgc25kX2NhcmRfZGlzY29ubmVj dChzdHJ1Y3Qgc25kX2NhcmQgKmNhcmQpDQogCQlyZXR1cm4gMDsNCiAJfQ0KIAljYXJkLT5zaHV0 ZG93biA9IDE7DQotCXNwaW5fdW5sb2NrKCZjYXJkLT5maWxlc19sb2NrKTsNCiANCiAJLyogcmVw bGFjZSBmaWxlLT5mX29wIHdpdGggc3BlY2lhbCBkdW1teSBvcGVyYXRpb25zICovDQotCXNwaW5f bG9jaygmY2FyZC0+ZmlsZXNfbG9jayk7DQogCWxpc3RfZm9yX2VhY2hfZW50cnkobWZpbGUsICZj YXJkLT5maWxlc19saXN0LCBsaXN0KSB7DQogCQkvKiBpdCdzIGNyaXRpY2FsIHBhcnQsIHVzZSBl bmRsZXNzIGxvb3AgKi8NCiAJCS8qIHdlIGhhdmUgbm8gcm9vbSB0byBmYWlsICovDQotLSANCjIu MTguNA==

3 3

No active event for loopback
by Jonas Holmberg 30 Mar '21

30 Mar '21

I don't get an active-event when a loopback device becomes active if format, rate and channels has changed. If I use aplay and alasctl monitor it looks like this: aplay -D hw:0,0,0 -f S32_LE -r 48000 -c 1 -s 1 /dev/zero => node hw:0, #0 (3,1,0,PCM Slave Active,0) VALUE node hw:0, #0 (3,1,0,PCM Slave Active,0) VALUE I.e. active value is true when playback starts and false when playback stops. aplay -D hw:0,0,0 -f S16_LE -r 48000 -c 1 -s 1 /dev/zero => node hw:0, #0 (3,1,0,PCM Slave Format,0) VALUE node hw:0, #0 (3,1,0,PCM Slave Active,0) VALUE I.e. format is changed to S16_LE when playback starts and active is false when playback stops. But no active-event was sent when playback started. The same happens if rate and channels changes: aplay -D hw:0,0,0 -f S16_LE -r 32000 -c 1 -s 1 /dev/zero => node hw:0, #0 (3,1,0,PCM Slave Rate,0) VALUE node hw:0, #0 (3,1,0,PCM Slave Active,0) VALUE aplay -D hw:0,0,0 -f S16_LE -r 32000 -c 2 -s 1 /dev/zero => node hw:0, #0 (3,1,0,PCM Slave Channels,0) VALUE node hw:0, #0 (3,1,0,PCM Slave Active,0) VALUE If all params are changed back at the same time only a format event is sent: aplay -D hw:0,0,0 -f S32_LE -r 48000 -c 1 -s 1 /dev/zero => node hw:0, #0 (3,1,0,PCM Slave Format,0) VALUE node hw:0, #0 (3,1,0,PCM Slave Active,0) VALUE Do format, rate and channels events always imply that the device becomes active or is it a bug that no active-event is sent when the device becomes active if a param has changed? /Jonas

1 0

[PATCH 0/2] soundwire: configure block packing mode with new
by Bard Liao 30 Mar '21

30 Mar '21

The existing code multiply a channel number by zero (SDW_BLK_GRP_CNT_1), and the result is used to configure the block packing mode. Sampling grouping and channel packing are two separate concepts in SoundWire. In addition, the bandwidth allocation allocates a vertical slice for each stream, which makes the use of the PER_CHANNEL packing mode irrelevant. Let's use the proper definition for block packing mode (PER_PORT). This change has no functional impact though since the net result is the same configuration of the DPN_BlockCtrl3 register, when implemented. Pierre-Louis Bossart (2): soundwire: add definition for DPn BlockPackingMode soundwire: generic_allocation: fix confusion between group and packing drivers/soundwire/generic_bandwidth_allocation.c | 7 +++---- include/linux/soundwire/sdw.h | 6 ++++++ 2 files changed, 9 insertions(+), 4 deletions(-) -- 2.17.1

2 3

[PATCH] soundwire: cadence: only prepare attached devices on clock stop
by Bard Liao 30 Mar '21

30 Mar '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We sometimes see COMMAND_IGNORED responses during the clock stop sequence. It turns out we already have information if devices are present on a link, so we should only prepare those when they are attached. In addition, even when COMMAND_IGNORED are received, we should still proceed with the clock stop. The device will not be prepared but that's not a problem. The only case where the clock stop will fail is if the Cadence IP reports an error (including a timeout), or if the devices throw a COMMAND_FAILED response. BugLink: https://github.com/thesofproject/linux/issues/2621 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index d05442e646a3..57c59a33ce61 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1450,10 +1450,12 @@ int sdw_cdns_clock_stop(struct sdw_cdns *cdns, bool block_wake) } /* Prepare slaves for clock stop */ - ret = sdw_bus_prep_clk_stop(&cdns->bus); - if (ret < 0) { - dev_err(cdns->dev, "prepare clock stop failed %d", ret); - return ret; + if (slave_present) { + ret = sdw_bus_prep_clk_stop(&cdns->bus); + if (ret < 0 && ret != -ENODATA) { + dev_err(cdns->dev, "prepare clock stop failed %d\n", ret); + return ret; + } } /* -- 2.17.1

2 1

[PATCH v5 0/9] soundwire: qcom: various improvements
by Srinivas Kandagatla 30 Mar '21

30 Mar '21

Thanks for reviewing v4 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Patch to add Auto Enumeration supported by the controller is also part of this series. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v4: - exported sdw_slave_add as kernel test robot reported error Srinivas Kandagatla (9): dt-bindings: soundwire: qcom: clarify data port bus parameters soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts soundwire: export sdw_compare_devid, sdw_extract_slave_id and sdw_slave_add soundwire: qcom: add auto enumeration support soundwire: qcom: wait for enumeration to be complete in probe .../bindings/soundwire/qcom,sdw.txt | 20 + drivers/soundwire/bus.c | 4 +- drivers/soundwire/qcom.c | 529 ++++++++++++++---- drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 2 + 5 files changed, 443 insertions(+), 113 deletions(-) -- 2.21.0

2 11

[PATCH 00/11] soundwire: some cleanup patches
by Bard Liao 30 Mar '21

30 Mar '21

To make soundwire driver more decent and less Cppcheck complaint. Pierre-Louis Bossart (11): soundwire: bus: use correct driver name in error messages soundwire: bus: test read status soundwire: bus: use consistent tests for return values soundwire: bus: demote clock stop prepare log to dev_dbg() soundwire: bus: uniquify dev_err() for SCP_INT access soundwire: bus: remove useless initialization soundwire: generic_bandwidth_allocation: remove useless init soundwire: intel: remove useless readl soundwire: qcom: check of_property_read status soundwire: stream: remove useless initialization soundwire: stream: remove useless bus initializations drivers/soundwire/bus.c | 54 +++++++++++-------- drivers/soundwire/bus_type.c | 15 ++++-- .../soundwire/generic_bandwidth_allocation.c | 4 +- drivers/soundwire/intel.c | 2 - drivers/soundwire/qcom.c | 3 ++ drivers/soundwire/stream.c | 8 +-- 6 files changed, 52 insertions(+), 34 deletions(-) -- 2.17.1

2 12

[PATCH 1/3] ASoC: SOF: parse multiple SSP DAI and hw configs
by Kai Vehmanen 29 Mar '21

29 Mar '21

From: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> ASoC parses multiple hw_configs defined in topology. However currently in SOF only the first config is used and others are discarded. First change SOF driver to parse and save possible multiple configs in ssp case. Also save the default config value provided by ASoC. Functionality with only one defined config stays the same. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/pcm.c | 8 +- sound/soc/sof/sof-audio.h | 2 + sound/soc/sof/topology.c | 190 ++++++++++++++++++++++---------------- 3 files changed, 116 insertions(+), 84 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 61c3fe17342d..c3dc533f2627 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -673,10 +673,10 @@ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_pa /* read rate and channels from topology */ switch (dai->dai_config->type) { case SOF_DAI_INTEL_SSP: - rate->min = dai->dai_config->ssp.fsync_rate; - rate->max = dai->dai_config->ssp.fsync_rate; - channels->min = dai->dai_config->ssp.tdm_slots; - channels->max = dai->dai_config->ssp.tdm_slots; + rate->min = dai->dai_config[dai->current_config].ssp.fsync_rate; + rate->max = dai->dai_config[dai->current_config].ssp.fsync_rate; + channels->min = dai->dai_config[dai->current_config].ssp.tdm_slots; + channels->max = dai->dai_config[dai->current_config].ssp.tdm_slots; dev_dbg(component->dev, "rate_min: %d rate_max: %d\n", rate->min, rate->max); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index dc930fc2f4b5..dc274e63ed9a 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -112,6 +112,8 @@ struct snd_sof_dai { const char *cpu_dai_name; struct sof_ipc_comp_dai comp_dai; + int number_configs; + int current_config; struct sof_ipc_dai_config *dai_config; struct list_head list; /* list in sdev dai list */ }; diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 10f99620eb31..036202768b75 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2811,12 +2811,14 @@ static void sof_dai_set_format(struct snd_soc_tplg_hw_config *hw_config, * name. Note that the function can only be used for the case that all DAIs * have a common DAI config for now. */ -static int sof_set_dai_config(struct snd_sof_dev *sdev, u32 size, - struct snd_soc_dai_link *link, - struct sof_ipc_dai_config *config) +static int sof_set_dai_config_multi(struct snd_sof_dev *sdev, u32 size, + struct snd_soc_dai_link *link, + struct sof_ipc_dai_config *config, + int num_conf, int curr_conf) { struct snd_sof_dai *dai; int found = 0; + int i; list_for_each_entry(dai, &sdev->dai_list, list) { if (!dai->name) @@ -2832,19 +2834,27 @@ static int sof_set_dai_config(struct snd_sof_dev *sdev, u32 size, * dai config's dai_index match to the component's * dai_index. */ - config->dai_index = dai->comp_dai.dai_index; + for (i = 0; i < num_conf; i++) + config[i].dai_index = dai->comp_dai.dai_index; + dev_dbg(sdev->dev, "set DAI config for %s index %d\n", + dai->name, config[curr_conf].dai_index); /* send message to DSP */ ret = sof_ipc_tx_message(sdev->ipc, - config->hdr.cmd, config, size, + config[curr_conf].hdr.cmd, + &config[curr_conf], size, &reply, sizeof(reply)); if (ret < 0) { - dev_err(sdev->dev, "error: failed to set DAI config for %s index %d\n", - dai->name, config->dai_index); + dev_err(sdev->dev, + "error: failed to set DAI config for %s index %d\n", + dai->name, config[curr_conf].dai_index); return ret; } - dai->dai_config = kmemdup(config, size, GFP_KERNEL); + + dai->number_configs = num_conf; + dai->current_config = curr_conf; + dai->dai_config = kmemdup(config, size * num_conf, GFP_KERNEL); if (!dai->dai_config) return -ENOMEM; @@ -2868,64 +2878,81 @@ static int sof_set_dai_config(struct snd_sof_dev *sdev, u32 size, return 0; } +static int sof_set_dai_config(struct snd_sof_dev *sdev, u32 size, + struct snd_soc_dai_link *link, + struct sof_ipc_dai_config *config) +{ + return sof_set_dai_config_multi(sdev, size, link, config, 1, 0); +} + static int sof_link_ssp_load(struct snd_soc_component *scomp, int index, struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg, struct snd_soc_tplg_hw_config *hw_config, - struct sof_ipc_dai_config *config) + struct sof_ipc_dai_config *config, int curr_conf) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct snd_soc_tplg_private *private = &cfg->priv; + int num_conf = le32_to_cpu(cfg->num_hw_configs); u32 size = sizeof(*config); int ret; + int i; - /* handle master/slave and inverted clocks */ - sof_dai_set_format(hw_config, config); - - /* init IPC */ - memset(&config->ssp, 0, sizeof(struct sof_ipc_dai_ssp_params)); - config->hdr.size = size; + /* + * Parse common data, we should have 1 common data per hw_config. + */ + ret = sof_parse_token_sets(scomp, &config->ssp, ssp_tokens, + ARRAY_SIZE(ssp_tokens), private->array, + le32_to_cpu(private->size), + num_conf, size); - ret = sof_parse_tokens(scomp, &config->ssp, ssp_tokens, - ARRAY_SIZE(ssp_tokens), private->array, - le32_to_cpu(private->size)); if (ret != 0) { dev_err(scomp->dev, "error: parse ssp tokens failed %d\n", le32_to_cpu(private->size)); return ret; } - config->ssp.mclk_rate = le32_to_cpu(hw_config->mclk_rate); - config->ssp.bclk_rate = le32_to_cpu(hw_config->bclk_rate); - config->ssp.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->ssp.tdm_slots = le32_to_cpu(hw_config->tdm_slots); - config->ssp.tdm_slot_width = le32_to_cpu(hw_config->tdm_slot_width); - config->ssp.mclk_direction = hw_config->mclk_direction; - config->ssp.rx_slots = le32_to_cpu(hw_config->rx_slots); - config->ssp.tx_slots = le32_to_cpu(hw_config->tx_slots); + /* process all possible hw configs */ + for (i = 0; i < num_conf; i++) { - dev_dbg(scomp->dev, "tplg: config SSP%d fmt 0x%x mclk %d bclk %d fclk %d width (%d)%d slots %d mclk id %d quirks %d\n", - config->dai_index, config->format, - config->ssp.mclk_rate, config->ssp.bclk_rate, - config->ssp.fsync_rate, config->ssp.sample_valid_bits, - config->ssp.tdm_slot_width, config->ssp.tdm_slots, - config->ssp.mclk_id, config->ssp.quirks); - - /* validate SSP fsync rate and channel count */ - if (config->ssp.fsync_rate < 8000 || config->ssp.fsync_rate > 192000) { - dev_err(scomp->dev, "error: invalid fsync rate for SSP%d\n", - config->dai_index); - return -EINVAL; - } + /* handle master/slave and inverted clocks */ + sof_dai_set_format(&hw_config[i], &config[i]); - if (config->ssp.tdm_slots < 1 || config->ssp.tdm_slots > 8) { - dev_err(scomp->dev, "error: invalid channel count for SSP%d\n", - config->dai_index); - return -EINVAL; + config[i].hdr.size = size; + + /* copy differentiating hw configs to ipc structs */ + config[i].ssp.mclk_rate = le32_to_cpu(hw_config[i].mclk_rate); + config[i].ssp.bclk_rate = le32_to_cpu(hw_config[i].bclk_rate); + config[i].ssp.fsync_rate = le32_to_cpu(hw_config[i].fsync_rate); + config[i].ssp.tdm_slots = le32_to_cpu(hw_config[i].tdm_slots); + config[i].ssp.tdm_slot_width = le32_to_cpu(hw_config[i].tdm_slot_width); + config[i].ssp.mclk_direction = hw_config[i].mclk_direction; + config[i].ssp.rx_slots = le32_to_cpu(hw_config[i].rx_slots); + config[i].ssp.tx_slots = le32_to_cpu(hw_config[i].tx_slots); + + dev_dbg(scomp->dev, "tplg: config SSP%d fmt 0x%x mclk %d bclk %d fclk %d width (%d)%d slots %d mclk id %d quirks %d\n", + config[i].dai_index, config[i].format, + config[i].ssp.mclk_rate, config[i].ssp.bclk_rate, + config[i].ssp.fsync_rate, config[i].ssp.sample_valid_bits, + config[i].ssp.tdm_slot_width, config[i].ssp.tdm_slots, + config[i].ssp.mclk_id, config[i].ssp.quirks); + + /* validate SSP fsync rate and channel count */ + if (config[i].ssp.fsync_rate < 8000 || config[i].ssp.fsync_rate > 192000) { + dev_err(scomp->dev, "error: invalid fsync rate for SSP%d\n", + config[i].dai_index); + return -EINVAL; + } + + if (config[i].ssp.tdm_slots < 1 || config[i].ssp.tdm_slots > 8) { + dev_err(scomp->dev, "error: invalid channel count for SSP%d\n", + config[i].dai_index); + return -EINVAL; + } } /* set config for all DAI's with name matching the link name */ - ret = sof_set_dai_config(sdev, size, link, config); + ret = sof_set_dai_config_multi(sdev, size, link, config, num_conf, curr_conf); if (ret < 0) dev_err(scomp->dev, "error: failed to save DAI config for SSP%d\n", config->dai_index); @@ -3216,11 +3243,13 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_link_config *cfg) { struct snd_soc_tplg_private *private = &cfg->priv; - struct sof_ipc_dai_config config; struct snd_soc_tplg_hw_config *hw_config; - int num_hw_configs; + struct sof_ipc_dai_config common_config; + struct sof_ipc_dai_config *config; + int curr_conf; + int num_conf; int ret; - int i = 0; + int i; if (!link->platforms) { dev_err(scomp->dev, "error: no platforms\n"); @@ -3257,13 +3286,11 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, return -EINVAL; } - /* Send BE DAI link configurations to DSP */ - memset(&config, 0, sizeof(config)); + memset(&common_config, 0, sizeof(common_config)); /* get any common DAI tokens */ - ret = sof_parse_tokens(scomp, &config, dai_link_tokens, - ARRAY_SIZE(dai_link_tokens), private->array, - le32_to_cpu(private->size)); + ret = sof_parse_tokens(scomp, &common_config, dai_link_tokens, ARRAY_SIZE(dai_link_tokens), + private->array, le32_to_cpu(private->size)); if (ret != 0) { dev_err(scomp->dev, "error: parse link tokens failed %d\n", le32_to_cpu(private->size)); @@ -3274,71 +3301,74 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, * DAI links are expected to have at least 1 hw_config. * But some older topologies might have no hw_config for HDA dai links. */ - num_hw_configs = le32_to_cpu(cfg->num_hw_configs); - if (!num_hw_configs) { - if (config.type != SOF_DAI_INTEL_HDA) { + hw_config = cfg->hw_config; + num_conf = le32_to_cpu(cfg->num_hw_configs); + if (!num_conf) { + if (common_config.type != SOF_DAI_INTEL_HDA) { dev_err(scomp->dev, "error: unexpected DAI config count %d!\n", le32_to_cpu(cfg->num_hw_configs)); return -EINVAL; } + num_conf = 1; + curr_conf = 0; } else { dev_dbg(scomp->dev, "tplg: %d hw_configs found, default id: %d!\n", cfg->num_hw_configs, le32_to_cpu(cfg->default_hw_config_id)); - for (i = 0; i < num_hw_configs; i++) { - if (cfg->hw_config[i].id == cfg->default_hw_config_id) + for (curr_conf = 0; curr_conf < num_conf; curr_conf++) { + if (hw_config[curr_conf].id == cfg->default_hw_config_id) break; } - if (i == num_hw_configs) { + if (curr_conf == num_conf) { dev_err(scomp->dev, "error: default hw_config id: %d not found!\n", le32_to_cpu(cfg->default_hw_config_id)); return -EINVAL; } } - /* configure dai IPC message */ - hw_config = &cfg->hw_config[i]; + /* Reserve memory for all hw configs, eventually freed by widget */ + config = kcalloc(num_conf, sizeof(*config), GFP_KERNEL); + if (!config) + return -ENOMEM; - config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; - config.format = le32_to_cpu(hw_config->fmt); + /* Copy common data to all config ipc structs */ + for (i = 0; i < num_conf; i++) { + config[i].hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config[i].format = hw_config[i].fmt; + config[i].type = common_config.type; + config[i].dai_index = common_config.dai_index; + } /* now load DAI specific data and send IPC - type comes from token */ - switch (config.type) { + switch (common_config.type) { case SOF_DAI_INTEL_SSP: - ret = sof_link_ssp_load(scomp, index, link, cfg, hw_config, - &config); + ret = sof_link_ssp_load(scomp, index, link, cfg, hw_config, config, curr_conf); break; case SOF_DAI_INTEL_DMIC: - ret = sof_link_dmic_load(scomp, index, link, cfg, hw_config, - &config); + ret = sof_link_dmic_load(scomp, index, link, cfg, hw_config + curr_conf, config); break; case SOF_DAI_INTEL_HDA: - ret = sof_link_hda_load(scomp, index, link, cfg, hw_config, - &config); + ret = sof_link_hda_load(scomp, index, link, cfg, hw_config + curr_conf, config); break; case SOF_DAI_INTEL_ALH: - ret = sof_link_alh_load(scomp, index, link, cfg, hw_config, - &config); + ret = sof_link_alh_load(scomp, index, link, cfg, hw_config + curr_conf, config); break; case SOF_DAI_IMX_SAI: - ret = sof_link_sai_load(scomp, index, link, cfg, hw_config, - &config); + ret = sof_link_sai_load(scomp, index, link, cfg, hw_config + curr_conf, config); break; case SOF_DAI_IMX_ESAI: - ret = sof_link_esai_load(scomp, index, link, cfg, hw_config, - &config); + ret = sof_link_esai_load(scomp, index, link, cfg, hw_config + curr_conf, config); break; default: - dev_err(scomp->dev, "error: invalid DAI type %d\n", - config.type); + dev_err(scomp->dev, "error: invalid DAI type %d\n", common_config.type); ret = -EINVAL; break; } - if (ret < 0) - return ret; - return 0; + kfree(config); + + return ret; } static int sof_link_hda_unload(struct snd_sof_dev *sdev, base-commit: e49bcf4f5579828d4d77e5fcc9cb799db69d1f85 -- 2.31.0

2 3

[PATCH RESEND 1/2] ASoC: amd: update spdx license for acp machine driver
by Vijendar Mukunda 29 Mar '21

29 Mar '21

update SPDX license for acp machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 29 +++++------------------------ 1 file changed, 5 insertions(+), 24 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index e65e007..84e3906 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -1,27 +1,8 @@ -/* - * Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec - * - * Copyright 2017 Advanced Micro Devices, Inc. - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR - * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, - * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR - * OTHER DEALINGS IN THE SOFTWARE. - * - */ +// SPDX-License-Identifier: MIT +// +// Machine driver for AMD ACP Audio engine using DA7219, RT5682 & MAX98357 codec +// +//Copyright 2017-2021 Advanced Micro Devices, Inc. #include <sound/core.h> #include <sound/soc.h> -- 2.7.4

4 4

[PATCH] ASoC: fsl_rpmsg: initialise pointers to NULL
by Shengjiu Wang 29 Mar '21

29 Mar '21

This fixes the following sparse warnings: sound/soc/fsl/fsl_rpmsg.c:45:45: sparse: sparse: Using plain integer as NULL pointer sound/soc/fsl/fsl_rpmsg.c:45:56: sparse: sparse: Using plain integer as NULL pointer Fixes: b73d9e6225e8 ("ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/fsl/fsl_rpmsg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 2d09d8850e2c..ea5c973e2e84 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -42,7 +42,7 @@ static int fsl_rpmsg_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_rpmsg *rpmsg = snd_soc_dai_get_drvdata(dai); - struct clk *p = rpmsg->mclk, *pll = 0, *npll = 0; + struct clk *p = rpmsg->mclk, *pll = NULL, *npll = NULL; u64 rate = params_rate(params); int ret = 0; -- 2.17.1

2 1

[PATCH -next] ASoC: dwc: Fix -Wmissing-prototypes warnings
by YueHaibing 29 Mar '21

29 Mar '21

while CONFIG_SND_DESIGNWARE_PCM is not set, building with W=1 shows this: sound/soc/dwc/local.h:127:6: warning: no previous prototype for ‘dw_pcm_push_tx’ [-Wmissing-prototypes] void dw_pcm_push_tx(struct dw_i2s_dev *dev) { } ^~~~~~~~~~~~~~ sound/soc/dwc/local.h:128:6: warning: no previous prototype for ‘dw_pcm_pop_rx’ [-Wmissing-prototypes] void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { } ^~~~~~~~~~~~~ sound/soc/dwc/local.h:129:5: warning: no previous prototype for ‘dw_pcm_register’ [-Wmissing-prototypes] int dw_pcm_register(struct platform_device *pdev) ^~~~~~~~~~~~~~~ Change these to inline functions to fix the warnings. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/dwc/local.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 91dc70a826f8..1c361eb6127e 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -124,9 +124,9 @@ void dw_pcm_push_tx(struct dw_i2s_dev *dev); void dw_pcm_pop_rx(struct dw_i2s_dev *dev); int dw_pcm_register(struct platform_device *pdev); #else -void dw_pcm_push_tx(struct dw_i2s_dev *dev) { } -void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { } -int dw_pcm_register(struct platform_device *pdev) +static inline void dw_pcm_push_tx(struct dw_i2s_dev *dev) { } +static inline void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { } +static inline int dw_pcm_register(struct platform_device *pdev) { return -EINVAL; } -- 2.17.1

2 1

[PATCH -next] ASoC: amd: acp-da7219-max98357a: Fix -Wunused-variable warning
by YueHaibing 29 Mar '21

29 Mar '21

While ACPI is not set, make W=1 warns: sound/soc/amd/acp-da7219-max98357a.c:684:28: warning: ‘cz_rt5682_card’ defined but not used [-Wunused-variable] static struct snd_soc_card cz_rt5682_card = { ^~~~~~~~~~~~~~ sound/soc/amd/acp-da7219-max98357a.c:671:28: warning: ‘cz_card’ defined but not used [-Wunused-variable] static struct snd_soc_card cz_card = { Use #ifdef block to guard this. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/acp-da7219-max98357a.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index e65e007fc604..1bf0458e22a8 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,13 +47,15 @@ #define DUAL_CHANNEL 2 #define RT5682_PLL_FREQ (48000 * 512) +extern bool bt_uart_enable; +void *acp_soc_is_rltk_max(struct device *dev); + +#ifdef CONFIG_ACPI static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; -extern bool bt_uart_enable; -void *acp_soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -692,6 +694,7 @@ static struct snd_soc_card cz_rt5682_card = { .controls = cz_mc_controls, .num_controls = ARRAY_SIZE(cz_mc_controls), }; +#endif void *acp_soc_is_rltk_max(struct device *dev) { -- 2.17.1

2 1

Re: [PATCH v2] ASoC: rt1019: add rt1019 amplifier driver
by Mark Brown 29 Mar '21

29 Mar '21

On Mon, Mar 29, 2021 at 08:09:26AM +0000, Jack Yu wrote: > > Bindings should also be sent to the DT list. get_maintainer.pl will tell you this. > 1. Does it mean I should add my name in MAINTAINERS? It means you should have sent the patch with the binding changes to the DT list and maintainers as well as everyone else. > 2. Is below warning message caused by #1 question? No, it's the content of the change that's introduced problems in the binding.

1 0

Adding movable PCIe BARs support in snd_hda_intel
by Andrey Grodzovsky 29 Mar '21

29 Mar '21

Hi, I am working on adding AMD related drivers support for PCIe BARs move feature developed by Sergei (https://linuxplumbersconf.org/event/7/contributions/847/attachments/584/103…). His feature is still not upstream, all his code and mine on top can be seen here - https://cgit.freedesktop.org/~agrodzov/linux/log/?h=yadro/pcie_hotplug/mova… I did basic implementation fro amdgpu driver and now I am doing the same for snd_hda_intel to support our on GPU Azalia audio chips. This relevant commit is here - https://cgit.freedesktop.org/~agrodzov/linux/commit/?h=yadro/pcie_hotplug/m… Any driver that wants to support movable BARs needs to implement rescan_prepare, rescan_done and bar_fixed callbacks where rescan_prepare needs to stop HW/SW and unamp all MMIO mappings and rescan_done needs to ioremap the BAR from it's new MMIO location and restart HW/SW. I am able currently to trigger BARs move by sysfs using "/sys/bus/pci/rescan" and the driver will go through the sequence I described above without hangs. Problem is that after this when i try to use mplayer I am getting following errors: andrey@andrey-test:~$ sudo mplayer -ao alsa:device=hw=0.3 Downloads/file_example_MP3_5MG.mp3 MPlayer 1.3.0 (Debian), built with gcc-9 (C) 2000-2016 MPlayer Team do_connect: could not connect to socket connect: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing Downloads/file_example_MP3_5MG.mp3. libavformat version 58.29.100 (external) Audio only file format detected. Load subtitles in Downloads/ ========================================================================== Opening audio decoder: [mpg123] MPEG 1.0/2.0/2.5 layers I, II, III AUDIO: 44100 Hz, 2 ch, s16le, 320.0 kbit/22.68% (ratio: 40000->176400) Selected audio codec: [mpg123] afm: mpg123 (MPEG 1.0/2.0/2.5 layers I, II, III) ========================================================================== AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 0.1 (00.0) of 132.0 (02:12.0) ??,?% Audio device got stuck! and in dmesg I see [ 365.355518] snd_hda_codec_hdmi hdaudioC0D0: Unable to sync register 0x2f0d00. -5 Also I see 296.619014] snd_hda_intel 0000:0a:00.1: CORB reset timeout#2, CORBRP = 65535 error during the rescan_done callback execution I know it has to do with the move of BAR's MMIO address because when i disallow BAR migration by returning true from bar_fixed callback I have no such errors and mplayer works fine. I enabled MMIO trace and didn't see that post BAR move there is a wrong MMIO access - all of them are from the new MMIO base offset - 0xfcd80000 (trace attached including mmio trace and dmesg) I would be happy for any idea on this and any comment on the correctness of my sequence in the code Andrey

3 25

[PATCH 1/2] ASoC: tlv320aic32x4: Increase maximum register in regmap
by Annaliese McDermond 29 Mar '21

29 Mar '21

AIC32X4_REFPOWERUP was added as a register, but the maximum register value in the regmap and regmap range was not correspondingly increased. This caused an error when this register was attempted to be written. Fixes: ec96690 ("ASoC: tlv320aic32x4: Enable fast charge") Cc: stable(a)vger.kernel.org Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f04f88c8d425..1ac3b3b12dc6 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -577,12 +577,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { .window_start = 0, .window_len = 128, .range_min = 0, - .range_max = AIC32X4_RMICPGAVOL, + .range_max = AIC32X4_REFPOWERUP, }, }; const struct regmap_config aic32x4_regmap_config = { - .max_register = AIC32X4_RMICPGAVOL, + .max_register = AIC32X4_REFPOWERUP, .ranges = aic32x4_regmap_pages, .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), }; -- 2.27.0

1 0

[PATCH 2/2] ASoC: tlv320aic32x4: Register clocks before registering component
by Annaliese McDermond 29 Mar '21

29 Mar '21

Clock registration must be performed before the component is registered. aic32x4_component_probe attempts to get all the clocks right off the bat. If the component is registered before the clocks there is a race condition where the clocks may not be registered by the time aic32x4_componet_probe actually runs. Fixes: d1c859d ("ASoC: codec: tlv3204: Increased maximum supported channels") Cc: stable(a)vger.kernel.org Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1ac3b3b12dc6..b689f26fc4be 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1243,6 +1243,10 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) if (ret) goto err_disable_regulators; + ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); + if (ret) + goto err_disable_regulators; + ret = devm_snd_soc_register_component(dev, &soc_component_dev_aic32x4, &aic32x4_dai, 1); if (ret) { @@ -1250,10 +1254,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) goto err_disable_regulators; } - ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); - if (ret) - goto err_disable_regulators; - return 0; err_disable_regulators: -- 2.27.0

1 0

[PATCH 0/2] ASoC: tlv320aic32x4: Bugfixes for tlv320aic32x4 driver
by Annaliese McDermond 29 Mar '21

29 Mar '21

These are a couple of bugfixes for the tlv320aic32x4 driver that we discovered after the Raspberry Pi Foundation's move to the 5.10 kernel. The first increases the maximum value in the regmap configuration so that the powerup command actually works. The second changes the ordering of the clock registration in the module probe function so that the clocks are properly registered before the component probe function runs. Annaliese McDermond (2): ASoC: tlv320aic32x4: Increase maximum register in regmap ASoC: tlv320aic32x4: Register clocks before registering component sound/soc/codecs/tlv320aic32x4.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) -- 2.27.0

1 0

[PATCH 1/2] ALSA: hda: Re-add dropped snd_poewr_change_state() calls
by Takashi Iwai 29 Mar '21

29 Mar '21

The card power state change via snd_power_change_state() at the system suspend/resume seems dropped mistakenly during the PM code rewrite. The card power state doesn't play much role nowadays but it's still referred in a few places such as the HDMI codec driver. This patch restores them, but in a more appropriate place now in the prepare and complete callbacks. Fixes: f5dac54d9d93 ("ALSA: hda: Separate runtime and system suspend") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5b492c3f816c..12e51ac973c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1025,6 +1025,7 @@ static int azx_prepare(struct device *dev) chip = card->private_data; chip->pm_prepared = 1; + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); /* HDA controller always requires different WAKEEN for runtime suspend * and system suspend, so don't use direct-complete here. @@ -1038,6 +1039,7 @@ static void azx_complete(struct device *dev) struct azx *chip; chip = card->private_data; + snd_power_change_state(card, SNDRV_CTL_POWER_D0); chip->pm_prepared = 0; } -- 2.26.2

1 1

[PATCH] ALSA: usb-audio: Apply sample rate quirk to Logitech Connect
by Ikjoon Jang 29 Mar '21

29 Mar '21

Logitech ConferenceCam Connect is a compound USB device with UVC and UAC. Not 100% reproducible but sometimes it keeps responding STALL to every control transfer once it receives get_freq request. This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419 Signed-off-by: Ikjoon Jang <ikjn(a)chromium.org> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d3001fb18141..176437a441e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1521,6 +1521,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */ + case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */ return true; } -- 2.31.0.291.g576ba9dcdaf-goog

4 7

[PATCH] ASoC: audio-graph-card: Add audio mixer for Motorola mdm6600
by Pavel Machek 29 Mar '21

29 Mar '21

motmdm.c handles audio configuration on "gsmtty2"; it needs to know whether we are in call or not; that part is in motmdm-state.c and listens on "gsmtty1". To configure Alsamixer for voice calls do for example: Speaker Right -> Voice Call Noise Cancellation -> Unmute Call Output -> Speakerphone Call -> 100 Mic2 -> 40 Left -> Mic 2 Voice -> 55 Mic2 -> 40 Left -> Mic 2 Tony wrote original version using custom interface to n_gsm, Pavel switched it to plain serdev and split it to two drivers to be easier to debug and understand. Credit is Tony's, bugs are probably Pavel's. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> Co-authored-by: Tony Lindgren <tony(a)atomide.com> --- v2: Fix compile error. --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/motmdm-state.c | 177 ++++++++ sound/soc/codecs/motmdm.c | 688 ++++++++++++++++++++++++++++++++ 4 files changed, 877 insertions(+) create mode 100644 sound/soc/codecs/motmdm-state.c create mode 100644 sound/soc/codecs/motmdm.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ba4eb54aafcb..94000ab3cf3e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -932,6 +932,14 @@ config SND_SOC_MAX9860 depends on I2C select REGMAP_I2C +config SND_SOC_MOTMDM + tristate "Motorola Modem TS 27.010 Voice Call Codec" + depends on SERIAL_DEV_BUS + help + Enable support for Motorola TS 27.010 serdev voice + call codec driver for Motorola Mapphone series of + devices such as Droid 4. + config SND_SOC_MSM8916_WCD_ANALOG tristate "Qualcomm MSM8916 WCD Analog Codec" depends on SPMI || COMPILE_TEST diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d277f0366e09..f63022e0a292 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -128,6 +128,8 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-motmdm-objs := motmdm.o +snd-soc-motmdm-state-objs := motmdm-state.o snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o @@ -443,6 +445,8 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_MOTMDM) += snd-soc-motmdm.o +obj-$(CONFIG_SND_SOC_MOTMDM) += snd-soc-motmdm-state.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o diff --git a/sound/soc/codecs/motmdm-state.c b/sound/soc/codecs/motmdm-state.c new file mode 100644 index 000000000000..a3b2870c5167 --- /dev/null +++ b/sound/soc/codecs/motmdm-state.c @@ -0,0 +1,177 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Motorola Mapphone MDM6600 voice call audio support + * Copyright 2018 - 2020 Tony Lindgren <tony(a)atomide.com> + * Copyright 2020 - 2021 Pavel Machek <pavel(a)ucw.cz> + * + * Designed to provide notifications about voice call state to the + * motmdm.c driver. This one listens on "gsmtty1". + */ + +#include <linux/init.h> +#include <linux/kfifo.h> +#include <linux/module.h> +#include <linux/of_graph.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/serdev.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#define MOTMDM_HEADER_LEN 5 /* U1234 */ +#define MOTMDM_AUDIO_MAX_LEN 128 +#define MOTMDM_VOICE_RESP_LEN 7 /* U1234~+CIEV= */ + +struct motmdm_driver_data { + struct serdev_device *serdev; + unsigned char *buf; + size_t len; + spinlock_t lock; /* enable/disabled lock */ +}; + +static BLOCKING_NOTIFIER_HEAD(modem_state_chain_head); + +int register_modem_state_notifier(struct notifier_block *nb) +{ + return blocking_notifier_chain_register(&modem_state_chain_head, nb); +} +EXPORT_SYMBOL_GPL(register_modem_state_notifier); + +int unregister_modem_state_notifier(struct notifier_block *nb) +{ + return blocking_notifier_chain_unregister(&modem_state_chain_head, nb); +} +EXPORT_SYMBOL_GPL(unregister_modem_state_notifier); + +static int modem_state_notifier_call_chain(unsigned long val) +{ + int ret; + ret = __blocking_notifier_call_chain(&modem_state_chain_head, val, NULL, + -1, NULL); + return notifier_to_errno(ret); +} + +/* Parses the voice call state from unsolicited notifications on dlci1 */ +static int motmdm_voice_get_state(struct motmdm_driver_data *ddata, + const unsigned char *buf, + size_t len) +{ + struct device *dev = &ddata->serdev->dev; + bool enable; + const unsigned char *state; + + if (len < MOTMDM_HEADER_LEN + MOTMDM_VOICE_RESP_LEN + 5) + return 0; + + /* We only care about the unsolicted messages */ + if (buf[MOTMDM_HEADER_LEN] != '~') + return 0; + + if (strncmp(buf + MOTMDM_HEADER_LEN + 1, "+CIEV=", 6)) + return len; + + state = buf + MOTMDM_HEADER_LEN + MOTMDM_VOICE_RESP_LEN; + dev_info(dev, "%s: ciev=%5s\n", __func__, state); + + if (!strncmp(state, "1,1,0", 5) || /* connecting */ + !strncmp(state, "1,4,0", 5) || /* incoming call */ + !strncmp(state, "1,2,0", 5)) /* connected */ + enable = true; + else if (!strncmp(state, "1,0,0", 5) || /* disconnected */ + !strncmp(state, "1,0,2", 5)) /* call failed */ + enable = false; + else + return len; + + modem_state_notifier_call_chain(enable); + return len; +} + +static int voice_receive_data(struct serdev_device *serdev, + const unsigned char *buf, + size_t len) +{ + struct motmdm_driver_data *ddata = serdev_device_get_drvdata(serdev); + + if (len > MOTMDM_AUDIO_MAX_LEN) + len = MOTMDM_AUDIO_MAX_LEN; + + if (len <= MOTMDM_HEADER_LEN) + return 0; + + if (buf[MOTMDM_HEADER_LEN] == '~') + motmdm_voice_get_state(ddata, buf, len); + + return len; +} + +static const struct serdev_device_ops voice_serdev_ops = { + .receive_buf = voice_receive_data, + .write_wakeup = serdev_device_write_wakeup, +}; + +static void motmdm_free_voice_serdev(struct motmdm_driver_data *ddata) +{ + serdev_device_close(ddata->serdev); +} + +static int motmdm_soc_probe(struct serdev_device *serdev) +{ + struct device *dev = &serdev->dev; + struct motmdm_driver_data *ddata; + int error; + + ddata = devm_kzalloc(dev, sizeof(*ddata), GFP_KERNEL); + if (!ddata) + return -ENOMEM; + + ddata->serdev = serdev; + spin_lock_init(&ddata->lock); + ddata->len = MOTMDM_AUDIO_MAX_LEN; + + ddata->buf = devm_kzalloc(dev, ddata->len, GFP_KERNEL); + if (!ddata->buf) + return -ENOMEM; + + serdev_device_set_drvdata(ddata->serdev, ddata); + serdev_device_set_client_ops(ddata->serdev, &voice_serdev_ops); + + error = serdev_device_open(ddata->serdev); + return error; +} + +static void motmdm_state_remove(struct serdev_device *serdev) +{ + struct motmdm_driver_data *ddata = serdev_device_get_drvdata(serdev); + + motmdm_free_voice_serdev(ddata); +} + +static int motmdm_state_probe(struct serdev_device *serdev) +{ + return motmdm_soc_probe(serdev); +} + +#ifdef CONFIG_OF +static const struct of_device_id motmdm_of_match[] = { + { .compatible = "motorola,mapphone-mdm6600-modem" }, + {}, +}; +MODULE_DEVICE_TABLE(of, motmdm_of_match); +#endif + +static struct serdev_device_driver motmdm_state_driver = { + .driver = { + .name = "mot-mdm6600-modem", + .of_match_table = of_match_ptr(motmdm_of_match), + }, + .probe = motmdm_state_probe, + .remove = motmdm_state_remove, +}; +module_serdev_device_driver(motmdm_state_driver); + +MODULE_ALIAS("platform:motmdm-state"); +MODULE_DESCRIPTION("Motorola Mapphone MDM6600 modem state driver"); +MODULE_AUTHOR("Pavel Machek <pavel(a)ucw.cz>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/motmdm.c b/sound/soc/codecs/motmdm.c new file mode 100644 index 000000000000..93615a01ff1b --- /dev/null +++ b/sound/soc/codecs/motmdm.c @@ -0,0 +1,688 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Motorola Mapphone MDM6600 voice call audio support + * Copyright 2018 - 2020 Tony Lindgren <tony(a)atomide.com> + * Copyright 2020 - 2021 Pavel Machek <pavel(a)ucw.cz> + * + * This one handles audio configuration on "gsmtty2"; it needs to know + * whether we are in call or not, and that part is in motmdm-state.c + * and listens on "gsmtty1". + */ + +#include <linux/init.h> +#include <linux/kfifo.h> +#include <linux/module.h> +#include <linux/of_graph.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/serdev.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "motmdm-state.h" + +#define MOTMDM_HEADER_LEN 5 /* U1234 */ + +#define MOTMDM_AUDIO_RESP_LEN 6 /* U1234+XXXX= */ +#define MOTMDM_AUDIO_MAX_LEN 128 + +#define MOTMDM_VOICE_RESP_LEN 7 /* U1234~+CIEV= */ + +struct motmdm_driver_data { + struct notifier_block notifier; + struct snd_soc_component *component; + struct snd_soc_dai *master_dai; + struct device *modem; + struct serdev_device *serdev; + struct regmap *regmap; + unsigned char *buf; + size_t len; + unsigned int parsed:1; + unsigned int enabled:1; + spinlock_t lock; /* enable/disabled lock */ + struct mutex mutex; /* for sending commands */ + wait_queue_head_t read_queue; + + unsigned int dtmf_val; + unsigned int dtmf_en; +}; + +enum motmdm_cmd { + CMD_AT_EACC, + CMD_AT_CLVL, + CMD_AT_NREC, +}; + +const char * const motmd_read_fmt[] = { + [CMD_AT_EACC] = "AT+EACC?", + [CMD_AT_CLVL] = "AT+CLVL?", + [CMD_AT_NREC] = "AT+NREC?", +}; + +const char * const motmd_write_fmt[] = { + [CMD_AT_EACC] = "AT+EACC=%u,0", + [CMD_AT_CLVL] = "AT+CLVL=%u", + [CMD_AT_NREC] = "AT+NREC=%u", +}; + +/* + * Currently unconfigured additional inactive (error producing) options + * seem to be: + * "TTY Headset", "HCQ Headset", "VCQ Headset", "No-Mic Headset", + * "Handset Fluence Med", "Handset Fluence Low", "Car Dock", "Lapdock" + */ +static const char * const motmdm_out_mux_texts[] = { + "Handset", "Headset", "Speakerphone", "Bluetooth", +}; + +static SOC_ENUM_SINGLE_EXT_DECL(motmdm_out_enum, motmdm_out_mux_texts); + +static const DECLARE_TLV_DB_SCALE(motmdm_gain_tlv, 0, 100, 0); + +int motmdm_send_command(struct motmdm_driver_data *ddata, + const u8 *buf, int len) +{ + struct device *dev = ddata->component->dev; + const int timeout_ms = 1000; + unsigned char cmd[MOTMDM_AUDIO_MAX_LEN]; + int ret, cmdlen; + + cmdlen = len + 5 + 1; + if (cmdlen > MOTMDM_AUDIO_MAX_LEN) + return -EINVAL; + + mutex_lock(&ddata->mutex); + memset(ddata->buf, 0, ddata->len); + ddata->parsed = false; + snprintf(cmd, cmdlen, "U%04li%s", jiffies % 10000, buf); + + ret = serdev_device_write(ddata->serdev, cmd, cmdlen, MAX_SCHEDULE_TIMEOUT); + if (ret < 0) + goto out_unlock; + + serdev_device_wait_until_sent(ddata->serdev, 0); + + ret = wait_event_timeout(ddata->read_queue, ddata->parsed, + msecs_to_jiffies(timeout_ms)); + if (ret == 0) { + ret = -ETIMEDOUT; + goto out_unlock; + } else if (ret < 0) { + goto out_unlock; + } + + if (strstr(ddata->buf, "ERROR")) { + dev_err(dev, "command %s error %s\n", cmd, ddata->buf); + ret = -EPIPE; + } + + ret = len; + +out_unlock: + mutex_unlock(&ddata->mutex); + printk("send_command -- ret %d\n", ret); + + return ret; +} + +static int motmdm_read_reg(void *context, unsigned int reg, + unsigned int *value) +{ + struct snd_soc_component *component = context; + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + const unsigned char *cmd; + unsigned int val; + int error; + + cmd = motmd_read_fmt[reg]; + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) { + dev_err(component->dev, "%s: %s failed with %i\n", + __func__, cmd, error); + + return error; + } + + error = kstrtouint(ddata->buf + MOTMDM_AUDIO_RESP_LEN, 0, &val); + if (error) + return -ENODEV; + + *value = val; + + return error; +} + +static int motmdm_write_reg(void *context, unsigned int reg, + unsigned int value) +{ + struct snd_soc_component *component = context; + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + const unsigned char *fmt, *cmd; + int error; + + fmt = motmd_write_fmt[reg]; + cmd = kasprintf(GFP_KERNEL, fmt, value); + if (!cmd) { + error = -ENOMEM; + goto free; + } + + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) + dev_err(component->dev, "%s: %s failed with %i\n", + __func__, cmd, error); + +free: + kfree(cmd); + + return error; +} + +static const struct reg_default motmdm_reg_defaults[] = { + { CMD_AT_EACC, 0x0 }, + { CMD_AT_CLVL, 0x0 }, +}; + +static const struct regmap_config motmdm_regmap = { + .reg_bits = 32, + .reg_stride = 1, + .val_bits = 32, + .max_register = CMD_AT_NREC, + .reg_defaults = motmdm_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(motmdm_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .reg_read = motmdm_read_reg, + .reg_write = motmdm_write_reg, +}; + +static int motmdm_value_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + enum motmdm_cmd reg, + int cmd_base) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + unsigned int val; + int error; + + error = regmap_read(ddata->regmap, reg, &val); + if (error) + return error; + + if (val >= cmd_base) + val -= cmd_base; + + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int motmdm_value_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + enum motmdm_cmd reg, + int cmd_base) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + int error; + + error = regmap_write(ddata->regmap, reg, + ucontrol->value.enumerated.item[0] + cmd_base); + if (error) + return error; + + regcache_mark_dirty(ddata->regmap); + + return error; +} + +static int motmdm_audio_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_get(kcontrol, ucontrol, CMD_AT_EACC, 1); +} + +static int motmdm_audio_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_put(kcontrol, ucontrol, CMD_AT_EACC, 1); +} + +static int motmdm_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_get(kcontrol, ucontrol, CMD_AT_CLVL, 0); +} + +static int motmdm_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_put(kcontrol, ucontrol, CMD_AT_CLVL, 0); +} + +static int motmdm_noise_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_get(kcontrol, ucontrol, CMD_AT_NREC, 0); +} + +static int motmdm_noise_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_put(kcontrol, ucontrol, CMD_AT_NREC, 0); +} + +static const char * const motmdm_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static SOC_ENUM_SINGLE_EXT_DECL(motmd_tonegen_dtmf_enum, + motmdm_tonegen_dtmf_key_txt); + +static int motmdm_dtmf_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = ddata->dtmf_val; + + return 0; +} + +static int motmdm_dtmf_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + + ddata->dtmf_val = ucontrol->value.enumerated.item[0]; + + return 0; +} + +static int motmdm_tonegen_dtmf_send_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = ddata->dtmf_en; + + return 0; +} + +static int motmdm_tonegen_dtmf_send_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + const unsigned char *cmd, *fmt = "AT+DTSE=%s,%i"; + const char *tone = ""; + int error; + + if (!ddata->enabled) + return 0; + + ddata->dtmf_en = ucontrol->value.enumerated.item[0]; + if (ddata->dtmf_en) + tone = motmdm_tonegen_dtmf_key_txt[ddata->dtmf_val]; + + /* Value 0 enables tone generator, 1 disables it */ + cmd = kasprintf(GFP_KERNEL, fmt, tone, !ddata->dtmf_en); + + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) { + dev_err(component->dev, "%s: %s failed with %i\n", + __func__, cmd, error); + goto free; + } + +free: + kfree(cmd); + + return error; +} + +static int +motmdm_enable_primary_dai(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + int error; + + if (!ddata->master_dai) + return -ENODEV; + + error = snd_soc_dai_set_sysclk(ddata->master_dai, 1, 19200000, + SND_SOC_CLOCK_OUT); + if (error) + return error; + + error = snd_soc_dai_set_fmt(ddata->master_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (error) + return error; + + error = snd_soc_dai_set_tdm_slot(ddata->master_dai, 0, 1, 1, 8); + if (error) + return error; + + return error; +} + +static int +motmdm_disable_primary_dai(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + int error; + + if (!ddata->master_dai) { + return -ENODEV; + } + + error = snd_soc_dai_set_sysclk(ddata->master_dai, 0, 26000000, + SND_SOC_CLOCK_OUT); + if (error) { + return error; + } + + error = snd_soc_dai_set_fmt(ddata->master_dai, + SND_SOC_DAIFMT_CBM_CFM); + if (error) { + return error; + } + + error = snd_soc_dai_set_tdm_slot(ddata->master_dai, 0, 0, 0, 48); + if (error) { + return error; + } + + return error; +} + +static int motmdm_find_primary_dai(struct snd_soc_component *component, + const char *name) +{ + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + struct device_node *bitclkmaster = NULL, *framemaster = NULL; + struct device_node *ep, *master_ep, *master = NULL; + struct snd_soc_dai_link_component dlc = { 0 }; + unsigned int daifmt; + + ep = of_graph_get_next_endpoint(component->dev->of_node, NULL); + if (!ep) + return -ENODEV; + + master_ep = of_graph_get_remote_endpoint(ep); + of_node_put(ep); + if (!master_ep) + return -ENODEV; + + daifmt = snd_soc_of_parse_daifmt(master_ep, NULL, + &bitclkmaster, &framemaster); + of_node_put(master_ep); + if (bitclkmaster && framemaster) + master = of_graph_get_port_parent(bitclkmaster); + of_node_put(bitclkmaster); + of_node_put(framemaster); + if (!master) + return -ENODEV; + + dlc.of_node = master; + dlc.dai_name = name; + ddata->master_dai = snd_soc_find_dai(&dlc); + of_node_put(master); + if (!ddata->master_dai) + return -EPROBE_DEFER; + + dev_info(component->dev, "Master DAI is %s\n", + dev_name(ddata->master_dai->dev)); + + return 0; +} + +static int motmdm_parse_tdm(struct snd_soc_component *component) +{ + return motmdm_find_primary_dai(component, "cpcap-voice"); +} + +static const struct snd_kcontrol_new motmdm_snd_controls[] = { + SOC_ENUM_EXT("Call Output", motmdm_out_enum, + motmdm_audio_out_get, + motmdm_audio_out_put), + SOC_SINGLE_EXT_TLV("Call Volume", + 0, 0, 7, 0, + motmdm_gain_get, + motmdm_gain_put, + motmdm_gain_tlv), + SOC_SINGLE_BOOL_EXT("Call Noise Cancellation", 0, + motmdm_noise_get, + motmdm_noise_put), + SOC_ENUM_EXT("Call DTMF", motmd_tonegen_dtmf_enum, + motmdm_dtmf_get, + motmdm_dtmf_put), + SOC_SINGLE_BOOL_EXT("Call DTMF Send", 0, + motmdm_tonegen_dtmf_send_get, + motmdm_tonegen_dtmf_send_put), +}; + +static struct snd_soc_dai_driver motmdm_dai[] = { + { + .name = "mdm-call", + .playback = { + .stream_name = "Voice Call Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Voice Call Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +static int +motmdm_notifier_call(struct notifier_block *bl, unsigned long state, + void *unused) +{ + struct motmdm_driver_data *ddata = + container_of(bl, struct motmdm_driver_data, notifier); + bool enable, notify = false; + unsigned long flags; + + enable = !!state; + + spin_lock_irqsave(&ddata->lock, flags); + if (ddata->enabled != enable) { + ddata->enabled = enable; + notify = true; + } + spin_unlock_irqrestore(&ddata->lock, flags); + + if (!notify) + return NOTIFY_DONE; + + if (enable) + motmdm_enable_primary_dai(ddata->component); + else + motmdm_disable_primary_dai(ddata->component); + + return NOTIFY_DONE; +} + +static int voice_receive_data(struct serdev_device *serdev, + const unsigned char *buf, + size_t len) +{ + struct motmdm_driver_data *ddata = serdev_device_get_drvdata(serdev); + struct device *dev = ddata->component->dev; + + printk("voice_receive_data: have %s %d\n", buf, len); + + if (len > MOTMDM_AUDIO_MAX_LEN) + len = MOTMDM_AUDIO_MAX_LEN; + + if (len <= MOTMDM_HEADER_LEN) + return 0; + + printk("voice_receive_data: command reply? -- %s %d\n", buf, len); + + snprintf(ddata->buf, len - MOTMDM_HEADER_LEN, buf + MOTMDM_HEADER_LEN); + dev_info(dev, "%s: received: %s\n", __func__, ddata->buf); + ddata->parsed = true; + wake_up(&ddata->read_queue); + + return len; +} + +static const struct serdev_device_ops voice_serdev_ops = { + .receive_buf = voice_receive_data, + .write_wakeup = serdev_device_write_wakeup, +}; + +static void motmdm_free_voice_serdev(struct motmdm_driver_data *ddata) +{ + serdev_device_close(ddata->serdev); +} + +static int motmdm_soc_probe(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata; + const unsigned char *cmd = "AT+CMUT=0"; + int error; + u32 line; + + ddata = devm_kzalloc(component->dev, sizeof(*ddata), GFP_KERNEL); + if (!ddata) + return -ENOMEM; + + error = of_property_read_u32(component->dev->of_node, "reg", &line); + if (error) + return error; + + ddata->serdev = (struct serdev_device *) component->dev; + ddata->component = component; + ddata->modem = component->dev->parent; + mutex_init(&ddata->mutex); + init_waitqueue_head(&ddata->read_queue); + ddata->len = PAGE_SIZE; + spin_lock_init(&ddata->lock); + snd_soc_component_set_drvdata(component, ddata); + ddata->len = MOTMDM_AUDIO_MAX_LEN; + + ddata->buf = devm_kzalloc(component->dev, ddata->len, GFP_KERNEL); + if (!ddata->buf) + return -ENOMEM; + + ddata->regmap = devm_regmap_init(component->dev, NULL, component, + &motmdm_regmap); + if (IS_ERR(ddata->regmap)) { + error = PTR_ERR(ddata->regmap); + dev_err(component->dev, "%s: Failed to allocate regmap: %d\n", + __func__, error); + + return error; + } + + serdev_device_set_drvdata(ddata->serdev, ddata); + serdev_device_set_client_ops(ddata->serdev, &voice_serdev_ops); + + error = serdev_device_open(ddata->serdev); + if (error) + return error; + + error = motmdm_parse_tdm(component); + if (error) + goto unregister_serdev; + + regcache_sync(ddata->regmap); + + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) + goto unregister_serdev; + + error = motmdm_disable_primary_dai(ddata->component); + if (error) + goto unregister_serdev; + + ddata->notifier.notifier_call = motmdm_notifier_call; + register_modem_state_notifier(&ddata->notifier); + + return 0; + +unregister_serdev: + motmdm_free_voice_serdev(ddata); + serdev_device_close(ddata->serdev); + + return error; +} + +static void motmdm_soc_remove(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + + unregister_modem_state_notifier(&ddata->notifier); + + motmdm_free_voice_serdev(ddata); +} + +static struct snd_soc_component_driver soc_codec_dev_motmdm = { + .probe = motmdm_soc_probe, + .remove = motmdm_soc_remove, + .controls = motmdm_snd_controls, + .num_controls = ARRAY_SIZE(motmdm_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int motmdm_codec_probe(struct serdev_device *serdev) +{ + return devm_snd_soc_register_component(&serdev->dev, + &soc_codec_dev_motmdm, + motmdm_dai, + ARRAY_SIZE(motmdm_dai)); +} + +#ifdef CONFIG_OF +static const struct of_device_id motmdm_of_match[] = { + { .compatible = "motorola,mapphone-mdm6600-codec" }, + {}, +}; +MODULE_DEVICE_TABLE(of, motmdm_of_match); +#endif + +static struct serdev_device_driver motmdm_driver = { + .probe = motmdm_codec_probe, + .driver = { + .name = "mot-mdm6600-codec", + .of_match_table = of_match_ptr(motmdm_of_match), + }, +}; +module_serdev_device_driver(motmdm_driver); + +MODULE_ALIAS("platform:motmdm-codec"); +MODULE_DESCRIPTION("ASoC Motorola Mapphone MDM6600 codec driver"); +MODULE_AUTHOR("Tony Lindgren <tony(a)atomide.com>"); +MODULE_LICENSE("GPL v2"); -- 2.20.1 -- http://www.livejournal.com/~pavelmachek

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[PATCH v5 1/2] platform/x86: dell-privacy: Add support for Dell hardware privacy
by Perry Yuan 28 Mar '21

28 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> add support for Dell privacy driver for the Dell units equipped hardware privacy design, which protect users privacy of audio and camera from hardware level. Once the audio or camera privacy mode activated, any applications will not get any audio or video stream when user pressed ctrl+F4 hotkey, audio privacy mode will be enabled, micmute led will be also changed accordingly The micmute led is fully controlled by hardware & EC(embedded controller) and camera mute hotkey is Ctrl+F9. Currently design only emmits SW_CAMERA_LENS_COVER event while the camera lens shutter will be changed by EC & hw(hadware) control *The flow is like this: 1) User presses key. HW does stuff with this key (timeout timer is started) 2) WMI event is emitted from BIOS to kernel 3) WMI event is received by dell-privacy 4) KEY_MICMUTE emitted from dell-privacy 5) Userland picks up key and modifies kcontrol for SW mute 6) Codec kernel driver catches and calls ledtrig_audio_set, like this: ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led ? LED_ON :LED_OFF); 7) If "LED" is set to on dell-privacy notifies EC, and timeout is cancelled, HW mic mute activated. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- v4 -> v5: * addressed feedback from Randy Dunlap * addressed feedback from Pierre-Louis Bossart * rebase to latest 5.12 rc4 upstream kernel * fix some space alignment problem v3 -> v4: * fix format for Kconfig * add sysfs document * add flow comments to the privacy wmi/acpi driver * addressed feedback from Barnabás Pőcze[Thanks very much] * export privacy_valid to make the global state simpler to query * fix one issue which will block the dell-laptop driver to load when privacy driver invalid * addressed feedback from Pierre-Louis Bossart,remove the EC ID match v2 -> v3: * add sysfs attributes doc v1 -> v2: * query EC handle from EC driver directly. * fix some code style. * add KEY_END to keymap array. * clean platform device when cleanup called * use hexadecimal format for log print in dev_dbg * remove __set_bit for the report keys from probe. * fix keymap leak * add err_free_keymap in dell_privacy_wmi_probe * wmi driver will be unregistered if privacy_acpi_init() fails * add sysfs attribute files for user space query. * add leds micmute driver to privacy acpi * add more design info the commit info --- --- .../testing/sysfs-platform-dell-privacy-wmi | 32 ++ drivers/platform/x86/dell/Kconfig | 16 + drivers/platform/x86/dell/Makefile | 3 + drivers/platform/x86/dell/dell-laptop.c | 23 +- drivers/platform/x86/dell/dell-privacy-acpi.c | 164 +++++++++ drivers/platform/x86/dell/dell-privacy-wmi.c | 340 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 35 ++ drivers/platform/x86/dell/dell-wmi.c | 27 +- 8 files changed, 627 insertions(+), 13 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-acpi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h diff --git a/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi new file mode 100644 index 000000000000..20c15a51ec38 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi @@ -0,0 +1,32 @@ +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/devices_supported +Date: Feb 2021 +KernelVersion: 5.12 +Contact: "perry.yuan(a)dell.com>" +Description: + Display which dell hardware level privacy devices are supported + “Dell Privacy” is a set of HW, FW, and SW features to enhance + Dell’s commitment to platform privacy for MIC, Camera, and + ePrivacy screens. + The supported hardware privacy devices are: + - 0 = Not Supported + - 1 = Supported + - Bit0 -> Microphone + - Bit1 -> Camera + - Bit2 -> ePrivacy Screen + +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/current_state +Date: Feb 2021 +KernelVersion: 5.12 +Contact: "perry.yuan(a)dell.com>" +Description: + Allow user space to check current dell privacy device state. + Describes the Device State class exposed by BIOS which can be + consumed by various applications interested in knowing the Privacy + feature capabilities + There are three Bits for available states: + - 0 = Enabled + - 1 = Disabled + - Bit0 -> Microphone + - Bit1 -> Camera + - Bit2 -> ePrivacyScreen + diff --git a/drivers/platform/x86/dell/Kconfig b/drivers/platform/x86/dell/Kconfig index e0a55337f51a..0e0f1eb35bd6 100644 --- a/drivers/platform/x86/dell/Kconfig +++ b/drivers/platform/x86/dell/Kconfig @@ -204,4 +204,20 @@ config DELL_WMI_SYSMAN To compile this driver as a module, choose M here: the module will be called dell-wmi-sysman. +config DELL_PRIVACY + tristate "Dell Hardware Privacy Support" + depends on ACPI + depends on ACPI_WMI + depends on INPUT + depends on DELL_LAPTOP + depends on LEDS_TRIGGER_AUDIO + select DELL_WMI + help + This driver provides support for the "Dell Hardware Privacy" feature + of Dell laptops. + Support for a micmute and camera mute privacy will be provided as + hardware button Ctrl+F4 and Ctrl+F9 hotkey. + + To compile this driver as a module, choose M here: the module will + be called dell_privacy. endif # X86_PLATFORM_DRIVERS_DELL diff --git a/drivers/platform/x86/dell/Makefile b/drivers/platform/x86/dell/Makefile index d720a3e42ae3..8b0883ee9412 100644 --- a/drivers/platform/x86/dell/Makefile +++ b/drivers/platform/x86/dell/Makefile @@ -19,3 +19,6 @@ obj-$(CONFIG_DELL_WMI_AIO) += dell-wmi-aio.o obj-$(CONFIG_DELL_WMI_DESCRIPTOR) += dell-wmi-descriptor.o obj-$(CONFIG_DELL_WMI_LED) += dell-wmi-led.o obj-$(CONFIG_DELL_WMI_SYSMAN) += dell-wmi-sysman/ +obj-$(CONFIG_DELL_PRIVACY) += dell-privacy.o +dell-privacy-objs := dell-privacy-wmi.o \ + dell-privacy-acpi.o diff --git a/drivers/platform/x86/dell/dell-laptop.c b/drivers/platform/x86/dell/dell-laptop.c index 70edc5bb3a14..4c840ab6cb59 100644 --- a/drivers/platform/x86/dell/dell-laptop.c +++ b/drivers/platform/x86/dell/dell-laptop.c @@ -31,6 +31,8 @@ #include "dell-rbtn.h" #include "dell-smbios.h" +#include "dell-privacy-wmi.h" + struct quirk_entry { bool touchpad_led; bool kbd_led_not_present; @@ -90,6 +92,7 @@ static struct rfkill *wifi_rfkill; static struct rfkill *bluetooth_rfkill; static struct rfkill *wwan_rfkill; static bool force_rfkill; +static bool has_privacy; module_param(force_rfkill, bool, 0444); MODULE_PARM_DESC(force_rfkill, "enable rfkill on non whitelisted models"); @@ -2206,10 +2209,16 @@ static int __init dell_init(void) if (dell_smbios_find_token(GLOBAL_MIC_MUTE_DISABLE) && dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); - if (ret < 0) - goto fail_led; + if (!privacy_valid) + has_privacy = true; + else + has_privacy = false; + if (!has_privacy) { + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); + if (ret < 0) + goto fail_led; + } } if (acpi_video_get_backlight_type() != acpi_backlight_vendor) @@ -2257,7 +2266,8 @@ static int __init dell_init(void) fail_get_brightness: backlight_device_unregister(dell_backlight_device); fail_backlight: - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); fail_led: dell_cleanup_rfkill(); fail_rfkill: @@ -2278,7 +2288,8 @@ static void __exit dell_exit(void) touchpad_led_exit(); kbd_led_exit(); backlight_device_unregister(dell_backlight_device); - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); dell_cleanup_rfkill(); if (platform_device) { platform_device_unregister(platform_device); diff --git a/drivers/platform/x86/dell/dell-privacy-acpi.c b/drivers/platform/x86/dell/dell-privacy-acpi.c new file mode 100644 index 000000000000..4c785b62c7c4 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-acpi.c @@ -0,0 +1,164 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/bits.h> +#include <linux/device.h> +#include <linux/fs.h> +#include <linux/kernel.h> +#include <linux/leds.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/platform_device.h> +#include <linux/string.h> +#include <linux/sysfs.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define PRIVACY_PLATFORM_NAME "dell-privacy-acpi" +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" + +struct privacy_acpi_priv { + struct device *dev; + struct platform_device *platform_device; + struct led_classdev cdev; +}; +static struct privacy_acpi_priv *privacy_acpi; + +static int dell_privacy_micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct privacy_acpi_priv *priv = privacy_acpi; + acpi_status status; + acpi_handle handle; + static char *acpi_method = (char *)"ECAK"; + + handle = ec_get_handle(); + if (!handle) + return -EIO; + if (!acpi_has_method(handle, acpi_method)) + return -EIO; + status = acpi_evaluate_object(handle, acpi_method, NULL, NULL); + if (ACPI_FAILURE(status)) { + dev_err(priv->dev, "Error setting privacy EC ack value: %s\n", + acpi_format_exception(status)); + return -EIO; + } + pr_debug("set dell privacy micmute ec ack event done\n"); + return 0; +} + +static int dell_privacy_acpi_remove(struct platform_device *pdev) +{ + struct privacy_acpi_priv *priv = dev_get_drvdata(privacy_acpi->dev); + + led_classdev_unregister(&priv->cdev); + + return 0; +} +/* + * Pressing the mute key activates a time delayed circuit to physically cut + * off the mute. The LED is in the same circuit, so it reflects the true + * state of the HW mute. The reason for the EC "ack" is so that software + * can first invoke a SW mute before the HW circuit is cut off. Without SW + * cutting this off first does not affect the time delayed muting or status + * of the LED but there is a possibility of a "popping" noise. + * + * If the EC receives the SW ack, the circuit will be activated before the + * delay completed. + * + * Exposing as an LED device allows the codec drivers notification path to + * EC ACK to work + */ +static int dell_privacy_leds_setup(struct device *dev) +{ + struct privacy_acpi_priv *priv = dev_get_drvdata(dev); + int ret = 0; + + priv->cdev.name = "dell-privacy::micmute"; + priv->cdev.max_brightness = 1; + priv->cdev.brightness_set_blocking = dell_privacy_micmute_led_set; + priv->cdev.default_trigger = "audio-micmute"; + priv->cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = devm_led_classdev_register(dev, &priv->cdev); + if (ret) + return ret; + return 0; +} + +static int dell_privacy_acpi_probe(struct platform_device *pdev) +{ + int ret; + + platform_set_drvdata(pdev, privacy_acpi); + privacy_acpi->dev = &pdev->dev; + privacy_acpi->platform_device = pdev; + + ret = dell_privacy_leds_setup(&pdev->dev); + if (ret) + return -EIO; + + return 0; +} + +static struct platform_driver dell_privacy_platform_drv = { + .driver = { + .name = PRIVACY_PLATFORM_NAME, + }, + .probe = dell_privacy_acpi_probe, + .remove = dell_privacy_acpi_remove, +}; + +int __init dell_privacy_acpi_init(void) +{ + int err; + struct platform_device *pdev; + + if (!wmi_has_guid(DELL_PRIVACY_GUID)) + return -ENODEV; + + privacy_acpi = kzalloc(sizeof(*privacy_acpi), GFP_KERNEL); + if (!privacy_acpi) + return -ENOMEM; + + err = platform_driver_register(&dell_privacy_platform_drv); + if (err) + goto pdrv_err; + + pdev = platform_device_register_simple( + PRIVACY_PLATFORM_NAME, PLATFORM_DEVID_NONE, NULL, 0); + if (IS_ERR(pdev)) { + err = PTR_ERR(pdev); + goto pdev_err; + } + + return 0; + +pdev_err: + platform_device_unregister(pdev); +pdrv_err: + kfree(privacy_acpi); + return err; +} + +void __exit dell_privacy_acpi_exit(void) +{ + struct platform_device *pdev = to_platform_device(privacy_acpi->dev); + + platform_device_unregister(pdev); + platform_driver_unregister(&dell_privacy_platform_drv); + kfree(privacy_acpi); +} + +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("DELL Privacy ACPI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.c b/drivers/platform/x86/dell/dell-privacy-wmi.c new file mode 100644 index 000000000000..8bde97613477 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.c @@ -0,0 +1,340 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/input.h> +#include <linux/input/sparse-keymap.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" +#define MICROPHONE_STATUS BIT(0) +#define CAMERA_STATUS BIT(1) +#define PRIVACY_SCREEN_STATUS BIT(2) + +/* + * The wmi_list is used to store the privacy_priv struct with mutex protecting + */ +static LIST_HEAD(wmi_list); +static DEFINE_MUTEX(list_mutex); +/* + * global privacy state for other modules to query if the privacy driver loaded successfully + * or no privacy WMI device was detected at initial probe phase + */ +int privacy_valid = -EPROBE_DEFER; +EXPORT_SYMBOL_GPL(privacy_valid); + +struct privacy_wmi_data { + struct input_dev *input_dev; + struct wmi_device *wdev; + struct list_head list; + u32 features_present; + u32 last_status; +}; + +/* + * Keymap for WMI privacy events of type 0x0012 + */ +static const struct key_entry dell_wmi_keymap_type_0012[] = { + /* privacy mic mute */ + { KE_KEY, 0x0001, { KEY_MICMUTE } }, + /* privacy camera mute */ + { KE_SW, 0x0002, { SW_CAMERA_LENS_COVER } }, + { KE_END, 0}, +}; + +int dell_privacy_state(void) +{ + return privacy_valid; +} +EXPORT_SYMBOL_GPL(dell_privacy_state); +/* + * The flow of privacy event: + * 1) User presses key. HW does stuff with this key (timeout is started) + * 2) WMI event is emitted from BIOS + * 3) WMI event is received by dell-privacy + * 4) KEY_MICMUTE emitted from dell-privacy + * 5) Userland picks up key and modifies kcontrol for SW mute + * 6) Codec kernel driver catches and calls ledtrig_audio_set defined by + * dell-privacy-acpi driver. + * codec driver will call like this to switch micmute led state. + * ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led ? LED_ON :LED_OFF); + * 7) If "LED" is set to on dell-privacy notifies EC,and timeout is cancelled, + * HW mic mute activated. + */ +void dell_privacy_process_event(int type, int code, int status) +{ + struct privacy_wmi_data *priv; + const struct key_entry *key; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + goto error; + } + key = sparse_keymap_entry_from_scancode(priv->input_dev, (type << 16) | code); + if (!key) { + dev_dbg(&priv->wdev->dev, "Unknown key with type 0x%04x and code 0x%04x pressed\n", + type, code); + goto error; + } + switch (code) { + case DELL_PRIVACY_TYPE_AUDIO: /* Mic mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + break; + case DELL_PRIVACY_TYPE_CAMERA: /* Camera mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + break; + default: + dev_dbg(&priv->wdev->dev, "unknown event type 0x%04x 0x%04x", + type, code); + } + +error: + mutex_unlock(&list_mutex); +} +EXPORT_SYMBOL_GPL(dell_privacy_process_event); + +static ssize_t devices_supported_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + return sysfs_emit(buf, "%x\n", priv->features_present); +} + +static ssize_t current_state_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + return sysfs_emit(buf, "%x\n", priv->last_status); +} + +static DEVICE_ATTR_RO(devices_supported); +static DEVICE_ATTR_RO(current_state); + +static struct attribute *privacy_attributes[] = { + &dev_attr_devices_supported.attr, + &dev_attr_current_state.attr, + NULL, +}; + +static const struct attribute_group privacy_attribute_group = { + .attrs = privacy_attributes +}; + +/* + * Describes the Device State class exposed by BIOS which can be consumed by + * various applications interested in knowing the Privacy feature capabilities. + * class DeviceState + * { + * [key, read] string InstanceName; + * [read] boolean ReadOnly; + * [WmiDataId(1), read] uint32 DevicesSupported; + * 0 – None, 0x1 – Microphone, 0x2 – Camera, 0x4 -ePrivacy Screen + * [WmiDataId(2), read] uint32 CurrentState; + * 0:Off; 1:On. Bit0 – Microphone, Bit1 – Camera, Bit2 - ePrivacyScreen + * }; + */ + +static int get_current_status(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + union acpi_object *obj_present; + u32 *buffer; + int ret = 0; + + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + return -EINVAL; + } + /* check privacy support features and device states */ + obj_present = wmidev_block_query(wdev, 0); + if (!obj_present) { + dev_err(&wdev->dev, "failed to read Binary MOF\n"); + ret = -EIO; + privacy_valid = ret; + return ret; + } + + if (obj_present->type != ACPI_TYPE_BUFFER) { + dev_err(&wdev->dev, "Binary MOF is not a buffer!\n"); + ret = -EIO; + privacy_valid = ret; + goto obj_free; + } + /* Although it's not technically a failure, this would lead to + * unexpected behavior + */ + if (obj_present->buffer.length != 8) { + dev_err(&wdev->dev, "Dell privacy buffer has unexpected length (%d)!\n", + obj_present->buffer.length); + ret = -EINVAL; + privacy_valid = ret; + goto obj_free; + } + buffer = (u32 *)obj_present->buffer.pointer; + priv->features_present = buffer[0]; + priv->last_status = buffer[1]; + privacy_valid = 0; + +obj_free: + kfree(obj_present); + return ret; +} + +static int dell_privacy_wmi_probe(struct wmi_device *wdev, const void *context) +{ + struct privacy_wmi_data *priv; + struct key_entry *keymap; + int ret, i; + + priv = devm_kzalloc(&wdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&wdev->dev, priv); + priv->wdev = wdev; + /* create evdev passing interface */ + priv->input_dev = devm_input_allocate_device(&wdev->dev); + if (!priv->input_dev) + return -ENOMEM; + /* remap the wmi keymap event to new keymap */ + keymap = kcalloc(ARRAY_SIZE(dell_wmi_keymap_type_0012), + sizeof(struct key_entry), GFP_KERNEL); + if (!keymap) { + ret = -ENOMEM; + goto err_free_dev; + } + /* remap the keymap code with Dell privacy key type 0x12 as prefix + * KEY_MICMUTE scancode will be reported as 0x120001 + */ + for (i = 0; i < ARRAY_SIZE(dell_wmi_keymap_type_0012); i++) { + keymap[i] = dell_wmi_keymap_type_0012[i]; + keymap[i].code |= (0x0012 << 16); + } + ret = sparse_keymap_setup(priv->input_dev, keymap, NULL); + if (ret) + return ret; + priv->input_dev->dev.parent = &wdev->dev; + priv->input_dev->name = "Dell Privacy Driver"; + priv->input_dev->id.bustype = BUS_HOST; + if (input_register_device(priv->input_dev)) { + pr_debug("input_register_device failed to register!\n"); + goto err_free_keymap; + } + mutex_lock(&list_mutex); + list_add_tail(&priv->list, &wmi_list); + mutex_unlock(&list_mutex); + + if (get_current_status(priv->wdev)) + goto err_free_input; + + ret = devm_device_add_group(&wdev->dev, &privacy_attribute_group); + if (ret) + goto err_free_input; + + kfree(keymap); + return 0; + +err_free_input: + input_unregister_device(priv->input_dev); +err_free_keymap: + privacy_valid = -ENODEV; + kfree(keymap); +err_free_dev: + return ret; +} + +static int dell_privacy_wmi_remove(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + + mutex_lock(&list_mutex); + list_del(&priv->list); + mutex_unlock(&list_mutex); + privacy_valid = -ENODEV; + input_unregister_device(priv->input_dev); + + return 0; +} + +static const struct wmi_device_id dell_wmi_privacy_wmi_id_table[] = { + { .guid_string = DELL_PRIVACY_GUID }, + { }, +}; + +static struct wmi_driver dell_privacy_wmi_driver = { + .driver = { + .name = "dell-privacy", + }, + .probe = dell_privacy_wmi_probe, + .remove = dell_privacy_wmi_remove, + .id_table = dell_wmi_privacy_wmi_id_table, +}; + +static int __init init_dell_privacy(void) +{ + int ret; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) { + privacy_valid = -ENODEV; + pr_debug("Unable to detect available Dell privacy devices: %d\n", ret); + return privacy_valid; + } + + ret = wmi_driver_register(&dell_privacy_wmi_driver); + if (ret) { + pr_err("failed to initialize privacy wmi driver: %d\n", ret); + return ret; + } + + ret = dell_privacy_acpi_init(); + if (ret) { + pr_err("failed to initialize privacy acpi driver: %d\n", ret); + goto err_init; + } + + return 0; + +err_init: + wmi_driver_unregister(&dell_privacy_wmi_driver); + return ret; +} + +static void dell_privacy_wmi_exit(void) +{ + wmi_driver_unregister(&dell_privacy_wmi_driver); +} + +static void __exit exit_dell_privacy(void) +{ + dell_privacy_wmi_exit(); + dell_privacy_acpi_exit(); +} + +module_init(init_dell_privacy); +module_exit(exit_dell_privacy); + +MODULE_DEVICE_TABLE(wmi, dell_wmi_privacy_wmi_id_table); +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("Dell Privacy WMI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.h b/drivers/platform/x86/dell/dell-privacy-wmi.h new file mode 100644 index 000000000000..6dda2990df5d --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.h @@ -0,0 +1,35 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#ifndef _DELL_PRIVACY_WMI_H_ +#define _DELL_PRIVACY_WMI_H_ + +#if IS_ENABLED(CONFIG_DELL_PRIVACY) +extern int privacy_valid; +int dell_privacy_state(void); +void dell_privacy_process_event(int type, int code, int status); +#else /* CONFIG_DELL_PRIVACY */ +int privacy_valid = -ENODEV; +static inline int dell_privacy_state(void) +{ + return privacy_valid; +} + +static inline void dell_privacy_process_event(int type, int code, int status) +{} +#endif /* CONFIG_DELL_PRIVACY */ + +int dell_privacy_acpi_init(void); +void dell_privacy_acpi_exit(void); + +/* DELL Privacy Type */ +enum { + DELL_PRIVACY_TYPE_UNKNOWN = 0x0, + DELL_PRIVACY_TYPE_AUDIO, + DELL_PRIVACY_TYPE_CAMERA, +}; +#endif diff --git a/drivers/platform/x86/dell/dell-wmi.c b/drivers/platform/x86/dell/dell-wmi.c index bbdb3e860892..389e86de24b0 100644 --- a/drivers/platform/x86/dell/dell-wmi.c +++ b/drivers/platform/x86/dell/dell-wmi.c @@ -13,20 +13,22 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt -#include <linux/kernel.h> -#include <linux/module.h> +#include <linux/acpi.h> +#include <linux/dmi.h> #include <linux/init.h> -#include <linux/slab.h> -#include <linux/types.h> #include <linux/input.h> #include <linux/input/sparse-keymap.h> -#include <linux/acpi.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> #include <linux/string.h> -#include <linux/dmi.h> +#include <linux/types.h> #include <linux/wmi.h> #include <acpi/video.h> + #include "dell-smbios.h" #include "dell-wmi-descriptor.h" +#include "dell-privacy-wmi.h" MODULE_AUTHOR("Matthew Garrett <mjg(a)redhat.com>"); MODULE_AUTHOR("Pali Rohár <pali(a)kernel.org>"); @@ -381,6 +383,7 @@ static void dell_wmi_notify(struct wmi_device *wdev, u16 *buffer_entry, *buffer_end; acpi_size buffer_size; int len, i; + int err; if (obj->type != ACPI_TYPE_BUFFER) { pr_warn("bad response type %x\n", obj->type); @@ -427,7 +430,6 @@ static void dell_wmi_notify(struct wmi_device *wdev, switch (buffer_entry[1]) { case 0x0000: /* One key pressed or event occurred */ - case 0x0012: /* Event with extended data occurred */ if (len > 2) dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[2]); @@ -439,6 +441,17 @@ static void dell_wmi_notify(struct wmi_device *wdev, dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[i]); break; + case 0x0012: + err = dell_privacy_state(); + if (err == 0) { + dell_privacy_process_event(buffer_entry[1], + buffer_entry[3], buffer_entry[4]); + } else { + if (len > 2) + dell_wmi_process_key(wdev, buffer_entry[1], + buffer_entry[2]); + } + break; default: /* Unknown event */ pr_info("Unknown WMI event type 0x%x\n", (int)buffer_entry[1]); -- 2.25.1

6 11

[PATCH 1/1] Adding some Roland devices' USB digital audio
by Lucas 28 Mar '21

28 Mar '21

A few days ago, I found this excellent informational post, and that Mike Oliphant's patch is in mainline now: http://blog.nostatic.org/2020/01/getting-boss-gt-1-effects-processor-to.html Since I was fairly certain it applies to the Roland devices I own, I added my USB IDs. Wonderfully, now USB audio capture seems to be working perfectly from the three devices I can test: the INTEGRA-7, Boutique D-05, & VG-99! I've also tested all three for playback capability, since that's implied when a device is using implicit feedback, which I hadn't realized the first time around. Anyway, they all seem to be working as expected for playback too. There are two caveats for playback, though: 1. The INTEGRA-7 won't output playback in any mode other than its "96[kHz]" setting, though capture works with all of its Sample Rate choices. 2. The D-05's analog outputs aren't as clean as the other two devices, with little quiet occasional clicks sounding something like dust on an LP record's play. Possibly that can be chalked up to it having only 1/8" minijack connections, but I suspect something more driver-related as I've not noticed it during use as a synthesizer. Its firmware version is "1.02(1033)," so not up-to-date which may mean it's just the hardware itself, but I've decided not to risk any changes with it yet. If I knew I could revert it, that would be another story. Also USB captures from it, play perfectly cleanly on my usual sound device, the Edirol UA-4FX. You probably already know this, as someone coded them, but here are the commands I've had to use for format compatibility with these devices: D-05: arecord -D hw:Boutique -f S32_LE -c 2 -r 96000 ./file.wav aplay -D hw:Boutique -f S32_LE -c 2 -r 96000 ./file.wav INTEGRA-7: arecord -D hw:INTEGRA7 -f S32_LE -c 2 -r 96000 ./file.wav (then change MENU->SYSTEM->SOUND->Sampling Rate to 44.1 [kHz] (otherwise its 96[kHz]) and change MENU->SYSTEM->SOUND->Ext Part Source Select to USB AUDIO) arecord -D hw:INTEGRA7 -f S32_LE -c 2 -r 44100 ./file-44100.wav aplay -D hw:INTEGRA7 -f S32_LE -c 2 -r 44100 ./file-44100.wav VG-99: arecord -D hw:VG99 -f S24_3LE -c 2 -r 44100 ./file.wav aplay -D hw:VG99 -f S24_3LE -c 2 -r 44100 ./file.wav I searched a bit for other Roland devices people have had USB digital audio issues with over the years, and took the liberty of adding them to the patch too. Likely there are still many more, especially in the Boutique line, but I couldn't find their USB IDs. I have another of the devices I added, with the patch, arriving soon (the R-26) so I'll have to get back to you with how that works too. I hope this can be added to the mainline kernel too, as this support should help many that have struggled. I, certainly, have been hoping somebody would figure this out for quite a while, and even bought a USB through/monitoring device to try to develop it myself (sadly, the software supporting that has its own bugs, so I probably should've just used wireshark). Suffice it to say, I'm quite happy to have this figured out, and to not have to delve into stuff I know nothing about! :) Thanks to anyone looking into adding this, Lucas The patch follows: diff -Nurp linux-5.11.9.orig/sound/usb/implicit.c linux-5.11.9.roland/sound/usb/implicit.c --- linux-5.11.9.orig/sound/usb/implicit.c 2021-03-24 05:54:19.000000000 -0500 +++ linux-5.11.9.roland/sound/usb/implicit.c 2021-03-26 21:43:10.177824402 -0500 @@ -71,27 +71,55 @@ static const struct snd_usb_implicit_fb_ .ep_num = 0x84, .iface = 0 }, /* MOTU MicroBook II */ /* No quirk for playback but with capture quirk (see below) */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00a6), /* Roland JUNO-G */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00ad), /* Roland SH-201 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00b2), /* Roland VG-99 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00c7), /* Roland V-Synth GT */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0111), /* Roland GAIA SH-01 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0120), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x012b), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x012f), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0130), /* BOSS BR-80 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0138), /* BOSS RC-300 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x013e), /* Roland R-26 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x015b), /* Roland INTEGRA-7 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x017c), /* Roland TR-8 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0185), /* BOSS GP-10 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01b5), /* Roland JP-08 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01ff), /* Roland D-05 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ {} /* terminator */ }; /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a6, 0x0d, 0x01), /* Roland JUNO-G */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ad, 0x0d, 0x01), /* Roland SH-201 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00b2, 0x0d, 0x01), /* Roland VG-99 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c7, 0x0d, 0x01), /* Roland V-Synth GT */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0111, 0x0d, 0x01), /* Roland GAIA SH-01 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0120, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012b, 0x0d, 0x01), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012f, 0x0d, 0x01), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0138, 0x0d, 0x01), /* BOSS RC-300 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x013e, 0x0d, 0x01), /* Roland R-26 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x015b, 0x0d, 0x01), /* Roland INTEGRA-7 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x017c, 0x0d, 0x01), /* Roland TR-8 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01b5, 0x0d, 0x01), /* Roland JP-08 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01ff, 0x0d, 0x01), /* Roland D-05 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */

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Re: 003 Firewire Isoch Rate Matching
by Takashi Sakamoto 27 Mar '21

27 Mar '21

Hi, (C.C. to alsa-devel, for someone interested in the topic) On Mon, Mar 22, 2021 at 03:10:57AM -0400, paul(a)neyrinck.com wrote: > Hi Takashi. It is nice to hear back from you. > > > The specific point of payload in isochronous packet from the model is that > > audio data frame is encoded by specific way. We call it as > > 'double-oh-three' algorithm and your can see its detail in Robin Gareus's > > documentation[4], who worked for reverse-engineering. You can see current > > implementation in 'sound/firewire/digi00x/amdtp-dot.c'. > > Yes, I see the middle byte encryption. But I am ignoring it for now while > I try to trying to get streaming to operate. I assume that I should be > able to first implement audio streaming and then implement the encryption > later. > > > Another point is that the sequence of number of data blocks per packet does > > not follow to IEC 61883-1/6. In this point, it's a mutation. > > Can you point me to the code that manages this? This is not something I > understand yet and I think I need more information. > > > The way to start/keep/stop a pair of isochronous communication is not so > > complicated and you can see it in `sound/firewire/digi00x/digi00x-stream.c`. > > I have the "start/keep" part working. My code streams talker packets of 19 > channels (1 MIDI + 18 24-bit audio). The 003 1394 LED changes from > flashing to constantly ON, but the audio output of the 003 is very > bad...just audio glitches. I looked at the FireWire analyzer image on > Robin Gareus' web page. But it shows a small sample. > > Are there any FireWire analyzer data files available to look at to help > me understand the protocols? On Mon, Mar 22, 2021 at 20:07:50PM -0400, paul(a)neyrinck.com wrote: > I have a question about something I do not understand. Does the 003 > require configuration of PCR (Plug Control Registers)? I am ignoring > this (because I don't understand it yet). At first, I describe the way to management a pair of isochronous packet streaming for the target device. Any registers of PCR is not used by 002/003. Instead, quadlet write operation to registers of target device is used for the equivalent operations: * 0x'ffff'e000'0110: sampling transfer frequency * 0x'ffff'e000'0100: registration for a pair of isochronous channels * 0x'ffff'e000'0004: start/stop packet streaming (The LED turns on/off). * 0x'ffff'e000'0000: current status of packet streaming Then the target device starts to transfer isochronous packet to IEEE 1394 bus, and listen to isochronous packet from IEEE 1394 bus, according to registered isochronous channels. You can see current implementation in begin_session() helper function in 'sound/firewire/digi00x/digi00x-stream.c'[1]. Of course, you need to keep isochronous bandwidth and channels in advance of packet streaming. Lock operations are used according to specification of IEEE 1394. You need to communicate to 'isochronous resource manager' node at current bus generation (it can differ each generation): * 0x'ffff'f000'0220: bandwidth * 0x'ffff'f000'0224: channel 31-18 * 0x'ffff'f000'0228: channel 17-0 I guess firewire stack on OS X has API for the above operations. The target device has service for applications to get current sampling rate from external inputs; e.g. S/PDIF: * 0x'ffff'e000'0114: rate of external clock source * 0x'ffff'e000'012c: detection of external clock input The target device can also switch source of sampling clock: * 0x'ffff'e000'0118: source of sampling clock The target device can switch mode of optical interface between ADAT and S/PDIF: * 0x'ffff'e000'011c: optical interface mode Next, I describe the way to process audio data frame in the isochronous packet. One isochronous packet consists of Common Isochronous Packet (CIP) header and payload. For example: CIP | quadlet ------- | ---------- header | 0x00130032 | 0x90020000 payload | ... | ... For the fields in CIP header, please refer to IEC 61883-1/6 which came from specification in vendor association (1394TA). Fortunately, the updated version of ancestor specification is freely available, 'Audio and Music Data Transmission Protocol 2.3' (1394 Trade Association, April 24, 2012, Document 2009013)[2]. The remarkable point of 002/003 case is that SYT field is not used, thus media clock recovery is done just by the number of data blocks in sequence of packet for base cycle time (not cleared yet). CIP payload consists of several number of 'data blocks' according to current sampling transfer frequency (STF). The first quadlet of each data block is for MIDI conformant data channel, which has sequence multiplexing. Some multi bit linear audio data channels follows. In detail, please refer to the above specification. The point is that one data block has one audio data frame. In the case of 002/003, the size of data block is: STF | data channel count -------- | ------------------ 44.1 kHz | 1 + 8 + 8 + 2 (MIDI/Analog/ADAT/SPDIF) 48.0 kHz | 1 + 8 + 8 + 2 (MIDI/Analog/ADAT/SPDIF) 88.2 kHz | 1 + 8 + 2 (MIDI/Analog/SPDIF) 96.0 kHz | 1 + 8 + 2 (MIDI/Analog/SPDIF) Well, we need to adjust the number of data blocks in sequence of packet to keep effective sampling rate (='media clock' in IEEE 1722). For this purpose, IEC 61883-1/6 defines two ways; blocking and non-blocking. In blocking way, an packet has fixed number of data blocks or empty packet. An example at 44.1 STF: isoc | the number of data cycle | blocks per packet ----- | ------------------- 0 | 8 1 | 8 2 | 0 3 | 8 4 | 8 5 | 8 6 | 0 ... | ... In non-blocking way, the number data blocks per packet is variable according to sampling timing each isochronous cycle. An example at 44.1 STF: isoc | the number of data cycle | blocks per packet ----- | ------------------- 0 | 6 1 | 6 2 | 5 3 | 6 4 | 5 5 | 6 6 | 5 ... | ... The above is basics of adjustment for media clock. You can see current implementation of ALSA firewire stack in 'calculate_data_blocks()' helper function in 'sound/firewire/amdtp-stream.c'[3]. In the implementation, the sequence of packet to the target device is adjusted independent of the sequence of packet from the target device. As you issued, the lack of adjustment is the cause of pop-noise in generated sound. We need the way to recover media clock in the sequence of packet from the target device. But as long as I know, the way is different between device types and models. For example, the sequence of packet from Windows driver to 002 is different from the one to 003 in the point. It's the reason that the issue is long standing. Additionally, Digi 002 and 003 uses protocol mutated from the above mechanism in the point of media clock. For example, we can see the actual sequence of packet to the target device at 48.0 kHz STF in Robin Gareus' post[5]. By calculating the number of data blocks according to data block counter field in CIP header: number of | data data blocks | block | counter ----------- | ------- - | 0x32 7 | 0x39 7 | 0x40 5 | 0x45 5 | 0x4a 5 | 0x4f 5 | 0x54 5 | 0x59 5 | 0x5e 5 | 0x63 5 | 0x68 7 | 0x6f 7 | 0x76 7 | 0x7d 7 | 0x84 7 | 0x8b 7 | 0x92 7 | 0x99 7 | 0xa0 5 | 0xa5 5 | 0xaa 5 | 0xaf 5 | 0xb4 5 | 0xb9 5 | 0xbe 5 | 0xc3 5 | 0xc8 7 | 0xcf 7 | 0xd6 7 | 0xdd It's neither blocking nor non-blocking. The sequence is relevant to both of packet buffering and audio data frame buffering, thus the above sequence the size of buffering is not something I know. I use DAP Technology FireSpy 810[5] for packet analyzing (fortunately I found it at bargain price in used market). The libhinoko[6] can make your Linux box as packet analyzer and might be useful. Enough later, I'll send you log of packet sequence for long duration (1 min or so) for 003 console and rack. > I have a Pro Tools license on Mac and PC. I think I have an Mbox 2 Pro > available to me. Please let me know how I can help you solve Mbox 2. Ok, thanks. I'll post some questions about control application for M-Box 2 Pro enough later. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/tree/sound/… [2] http://1394ta.org/developers/publicspecs/2009013.pdf [3] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/tree/sound/… [4] https://github.com/takaswie/snd-firewire-improve/issues/26#issuecomment-803… [5] https://gareus.org/wiki/digi003 [6] https://www.daptechnology.com/products/bus-analyzer/single-bus/single-bus-g… [7] https://mailman.alsa-project.org/pipermail/alsa-devel/2019-April/147862.html Regards Takashi Sakamoto

1 1

[PATCH] Behringer UFX1604: get rid of pops and clicks while on 96000hz
by Geraldo 27 Mar '21

27 Mar '21

Hello everyone! This one has been bugging me for quite a while. I went deep hard in the guts of ALSA to try to solve it, and it turned out to be a minor thing apparently. The problem is old, and every UFX1604 Linux user can attest that it's impossible to use 96000hz in DUPLEX mode without annoying pops and clicks on the capture stream. The fix is simple: after we alter the CLOCK_SOURCE to match our sample rate, let's tell the CLOCK_SELECTOR we want CLOCK_SOURCE 212 (synced to USB SOF) on pin 1. Solves the problem for me, no more pops and clicks while on 96000hz. If you own an UFX1604 please give this patch a good testing. Let me know if it solves the pops and clicks on the input stream for you while on DUPLEX 96000hz. --- clock.c.git 2021-03-22 04:19:55.543485748 -0300 +++ clock.c 2021-03-25 19:23:38.597197159 -0300 @@ -610,6 +610,13 @@ int snd_usb_set_sample_rate_v2v3(struct if (err < 0) return err; + if (chip->usb_id == USB_ID(0x1397, 0x0001)) { /* Behringer UFX1604 */ + printk(KERN_WARNING "Setting clock selector for UFX1604"); + err = uac_clock_selector_set_val(chip, 211, 1); + if (err < 0) + return err; + } + return get_sample_rate_v2v3(chip, fmt->iface, fmt->altsetting, clock); }

1 2

RE: Patch for some Roland devices' USB digital audio
by Lucas 27 Mar '21

27 Mar '21

Sorry, I should've realized that embedded plaintext patches are preferred, as I've been on the linux-media list since forever. This should improve the ability to find the pertinent devices too, so here it is again: A couple days ago, I found this excellent informational post, and that Mike Oliphant's patch is in mainline now: http://blog.nostatic.org/2020/01/getting-boss-gt-1-effects-processor-to.html Since I was fairly certain it applies to the Roland devices I own, I added my USB IDs. Wonderfully, now USB audio capture seems to be working perfectly from the three devices I can test: the INTEGRA-7, Boutique D-05, & VG-99! I don't know if the D-05 and VG-99 have digital audio playback/output, as I only tried it with the INTEGRA-7. It does work, but only when the INTEGRA-7's "Sampling Rate" is set to "44.1 kHz" and "Ext Part Source Select" is set to "USB AUDIO". I searched a bit for other Roland devices people have had USB digital audio issues with over the years, and took the liberty of adding them to the patch too. Likely there are still many more, especially in the Boutique line, but I couldn't find their USB IDs. I hope this can be added to the mainline kernel too, as this support should help many that have struggled. I, certainly, have been hoping somebody would figure this out for quite a while, and even bought a USB through/monitoring device to try to develop it myself (sadly, the software supporting that has its own bugs, so I probably should've just used wireshark). Suffice it to say, I'm quite happy to have this figured out, and to not have to delve into stuff I know nothing about! :) Thanks, Lucas The patch follows: diff -Nur ./linux-5.11.9.orig/sound/usb/implicit.c ./linux-5.11.9.roland/sound/usb/implicit.c --- ./linux-5.11.9.orig/sound/usb/implicit.c 2021-03-24 05:54:19.000000000 -0500 +++ ./linux-5.11.9.roland/sound/usb/implicit.c 2021-03-25 03:05:33.414870404 -0500 @@ -71,13 +71,27 @@ .ep_num = 0x84, .iface = 0 }, /* MOTU MicroBook II */ /* No quirk for playback but with capture quirk (see below) */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00a6), /* Roland JUNO-G */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00ad), /* Roland SH-201 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00b2), /* Roland VG-99 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00c7), /* Roland V-Synth GT */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0111), /* Roland GAIA SH-01 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0120), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x012b), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x012f), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0130), /* BOSS BR-80 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0138), /* BOSS RC-300 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x013e), /* Roland R-26 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x015b), /* Roland INTEGRA-7 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x017c), /* Roland TR-8 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0185), /* BOSS GP-10 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01b5), /* Roland JP-08 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01ff), /* Roland D-05 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ {} /* terminator */ @@ -85,13 +99,27 @@ /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a6, 0x0d, 0x01), /* Roland JUNO-G */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ad, 0x0d, 0x01), /* Roland SH-201 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00b2, 0x0d, 0x01), /* Roland VG-99 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c7, 0x0d, 0x01), /* Roland V-Synth GT */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0111, 0x0d, 0x01), /* Roland GAIA SH-01 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0120, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012b, 0x0d, 0x01), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012f, 0x0d, 0x01), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0138, 0x0d, 0x01), /* BOSS RC-300 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x013e, 0x0d, 0x01), /* Roland R-26 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x015b, 0x0d, 0x01), /* Roland INTEGRA-7 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x017c, 0x0d, 0x01), /* Roland TR-8 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01b5, 0x0d, 0x01), /* Roland JP-08 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01ff, 0x0d, 0x01), /* Roland D-05 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */

1 1

[PATCH v2] ASoC: rt1019: add rt1019 amplifier driver
by jack.yu＠realtek.com 26 Mar '21

26 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> This is initial amplifier driver for rt1019. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1019.yaml | 33 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1019.c | 940 ++++++++++++++++++ sound/soc/codecs/rt1019.h | 320 ++++++ 5 files changed, 1302 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt1019.yaml create mode 100644 sound/soc/codecs/rt1019.c create mode 100644 sound/soc/codecs/rt1019.h diff --git a/Documentation/devicetree/bindings/sound/rt1019.yaml b/Documentation/devicetree/bindings/sound/rt1019.yaml new file mode 100644 index 000000000000..c24c29eafa54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1019.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rt1019.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1019 Mono Class-D Audio Amplifier + +maintainers: + - jack.yu(a)realtek.com + +properties: + compatible: + const: realtek,rt1019 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + rt1019: codec@28 { + compatible = "realtek,rt1019"; + reg = <0x28>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ce74c99a305..4ab34bca71aa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -161,6 +161,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1011 imply SND_SOC_RT1015 imply SND_SOC_RT1015P + imply SND_SOC_RT1019 imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 @@ -1079,6 +1080,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT1011=y default y if SND_SOC_RT1015=y default y if SND_SOC_RT1015P=y + default y if SND_SOC_RT1019=y default y if SND_SOC_RT1305=y default y if SND_SOC_RT1308=y default m if SND_SOC_RT5514=m @@ -1097,6 +1099,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT1011=m default m if SND_SOC_RT1015=m default m if SND_SOC_RT1015P=m + default m if SND_SOC_RT1019=m default m if SND_SOC_RT1305=m default m if SND_SOC_RT1308=m @@ -1133,6 +1136,10 @@ config SND_SOC_RT1015P tristate depends on GPIOLIB +config SND_SOC_RT1019 + tristate + depends on I2C + config SND_SOC_RT1305 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dcc2f757bb82..edff5c5b92d3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -170,6 +170,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o snd-soc-rt1015-objs := rt1015.o snd-soc-rt1015p-objs := rt1015p.o +snd-soc-rt1019-objs := rt1019.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o snd-soc-rt1308-sdw-objs := rt1308-sdw.o @@ -487,6 +488,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o obj-$(CONFIG_SND_SOC_RT1015P) += snd-soc-rt1015p.o +obj-$(CONFIG_SND_SOC_RT1019) += snd-soc-rt1019.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c new file mode 100644 index 000000000000..fb275686a00f --- /dev/null +++ b/sound/soc/codecs/rt1019.c @@ -0,0 +1,940 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1019.c -- RT1019 ALSA SoC audio amplifier driver +// Author: Jack Yu <jack.yu(a)realtek.com> +// +// Copyright(c) 2021 Realtek Semiconductor Corp. +// +// + +#include <linux/acpi.h> +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt1019.h" + +static const struct reg_default rt1019_reg[] = { + { 0x0000, 0x00 }, + { 0x0002, 0x55 }, + { 0x0003, 0x55 }, + { 0x0005, 0x54 }, + { 0x0006, 0x05 }, + { 0x0007, 0x01 }, + { 0x0008, 0x70 }, + { 0x0009, 0x28 }, + { 0x000a, 0x7f }, + { 0x0011, 0x04 }, + { 0x0013, 0x00 }, + { 0x0015, 0x00 }, + { 0x0017, 0x00 }, + { 0x0019, 0x30 }, + { 0x001b, 0x01 }, + { 0x001d, 0x18 }, + { 0x001f, 0x29 }, + { 0x0021, 0x09 }, + { 0x0023, 0x02 }, + { 0x0025, 0x00 }, + { 0x0026, 0x00 }, + { 0x0028, 0x03 }, + { 0x0053, 0x00 }, + { 0x0055, 0x00 }, + { 0x0056, 0x00 }, + { 0x0057, 0x00 }, + { 0x0058, 0x00 }, + { 0x005a, 0x00 }, + { 0x005c, 0x00 }, + { 0x005d, 0x00 }, + { 0x005e, 0x10 }, + { 0x005f, 0xec }, + { 0x0061, 0x10 }, + { 0x0062, 0x19 }, + { 0x0064, 0x00 }, + { 0x0066, 0x08 }, + { 0x0068, 0x00 }, + { 0x006a, 0x00 }, + { 0x006c, 0x00 }, + { 0x006e, 0x00 }, + { 0x0100, 0x80 }, + { 0x0100, 0x51 }, + { 0x0102, 0x23 }, + { 0x0102, 0x0f }, + { 0x0104, 0x6c }, + { 0x0105, 0xec }, + { 0x0106, 0x00 }, + { 0x0107, 0x00 }, + { 0x0108, 0x00 }, + { 0x0200, 0x40 }, + { 0x0201, 0x00 }, + { 0x0202, 0x00 }, + { 0x0203, 0x00 }, + { 0x0301, 0x02 }, + { 0x0302, 0xaa }, + { 0x0303, 0x2a }, + { 0x0304, 0x6a }, + { 0x0306, 0xb0 }, + { 0x0308, 0x48 }, + { 0x030a, 0x0a }, + { 0x030b, 0x4b }, + { 0x030d, 0x7d }, + { 0x030e, 0xef }, + { 0x030f, 0x5a }, + { 0x0311, 0x00 }, + { 0x0312, 0x3e }, + { 0x0313, 0x86 }, + { 0x0315, 0xa8 }, + { 0x0318, 0x1b }, + { 0x031a, 0x3d }, + { 0x031c, 0x40 }, + { 0x031d, 0x40 }, + { 0x031e, 0x30 }, + { 0x031f, 0xbb }, + { 0x0320, 0xa5 }, + { 0x0321, 0xa5 }, + { 0x0323, 0x5a }, + { 0x0324, 0xaa }, + { 0x0325, 0x80 }, + { 0x0326, 0xaa }, + { 0x0327, 0x66 }, + { 0x0328, 0x94 }, + { 0x0329, 0x00 }, + { 0x0330, 0x00 }, + { 0x0331, 0x30 }, + { 0x0332, 0x05 }, + { 0x0400, 0x03 }, + { 0x0401, 0x02 }, + { 0x0402, 0x01 }, + { 0x0403, 0x23 }, + { 0x0404, 0x45 }, + { 0x0405, 0x67 }, + { 0x0500, 0x80 }, + { 0x0501, 0x00 }, + { 0x0502, 0x00 }, + { 0x0503, 0x00 }, + { 0x0504, 0xff }, + { 0x0505, 0x24 }, + { 0x0600, 0x75 }, + { 0x0601, 0x41 }, + { 0x0602, 0x60 }, + { 0x0603, 0x7f }, + { 0x0604, 0x65 }, + { 0x0605, 0x43 }, + { 0x0606, 0x00 }, + { 0x0607, 0x00 }, + { 0x0608, 0x00 }, + { 0x0609, 0x00 }, + { 0x060a, 0x00 }, + { 0x060b, 0x00 }, + { 0x060c, 0x00 }, + { 0x060d, 0x00 }, + { 0x060e, 0x00 }, + { 0x060f, 0x00 }, + { 0x0700, 0x15 }, + { 0x0701, 0xc8 }, + { 0x0704, 0x02 }, + { 0x0705, 0x00 }, + { 0x0706, 0x00 }, + { 0x0707, 0x80 }, + { 0x0708, 0x08 }, + { 0x0709, 0x00 }, + { 0x0800, 0x00 }, + { 0x0801, 0x00 }, + { 0x0802, 0x09 }, + { 0x0803, 0x00 }, + { 0x0900, 0x87 }, + { 0x0a01, 0x99 }, + { 0x0a02, 0x40 }, + { 0x0a03, 0x10 }, + { 0x0b00, 0x50 }, + { 0x0b01, 0xc3 }, + { 0x0c00, 0x84 }, + { 0x0c01, 0x00 }, + { 0x0c02, 0xbb }, + { 0x0c03, 0x80 }, + { 0x0c04, 0x10 }, + { 0x0c05, 0x30 }, + { 0x0c06, 0x00 }, + { 0x0d00, 0x80 }, + { 0x0d01, 0xbb }, + { 0x0d02, 0x80 }, + { 0x0d03, 0x00 }, + { 0x0d04, 0x00 }, + { 0x0d05, 0x00 }, + { 0x0e00, 0x80 }, + { 0x0e01, 0xbb }, + { 0x0e02, 0x80 }, + { 0x0e03, 0x00 }, + { 0x0e04, 0x10 }, + { 0x0e05, 0x30 }, + { 0x0f00, 0x80 }, + { 0x0f01, 0xbb }, + { 0x0f02, 0x80 }, + { 0x0f03, 0x00 }, + { 0x0f04, 0x10 }, + { 0x0f05, 0x30 }, + { 0x0f06, 0x88 }, + { 0x0f07, 0x88 }, + { 0x0f08, 0x00 }, + { 0x0f09, 0x00 }, +}; + +static bool rt1019_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1019_PWR_STRP_1: + case RT1019_PWR_STRP_2: + case RT1019_SIL_DET_GAT: + case RT1019_PHASE_SYNC: + case RT1019_STAT_MACH_2: + case RT1019_FS_DET_1: + case RT1019_FS_DET_2: + case RT1019_FS_DET_3: + case RT1019_FS_DET_4: + case RT1019_FS_DET_5: + case RT1019_FS_DET_6: + case RT1019_FS_DET_7: + case RT1019_ANA_READ: + case RT1019_VER_ID: + case RT1019_CUSTOM_ID: + case RT1019_VEND_ID_1: + case RT1019_VEND_ID_2: + case RT1019_DEV_ID_1: + case RT1019_DEV_ID_2: + case RT1019_CAL_TOP_3: + case RT1019_CAL_TOP_7: + case RT1019_CAL_TOP_17: + case RT1019_CAL_TOP_18: + case RT1019_CAL_TOP_19: + case RT1019_CAL_TOP_20: + case RT1019_CAL_TOP_21: + case RT1019_CAL_TOP_22: + case RT1019_MDRE_CTRL_2: + case RT1019_MDRE_CTRL_3: + case RT1019_MDRE_CTRL_4: + case RT1019_SIL_DET_2: + case RT1019_PWM_DC_DET_1: + case RT1019_PMC_8: + case RT1019_PMC_9: + case RT1019_SPKDRC_7: + case RT1019_HALF_FREQ_7: + case RT1019_CUR_CTRL_11: + case RT1019_CUR_CTRL_12: + case RT1019_CUR_CTRL_13: + return true; + + default: + return false; + } +} + +static bool rt1019_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1019_RESET: + case RT1019_PAD_DRV_1: + case RT1019_PAD_DRV_2: + case RT1019_PAD_PULL_1: + case RT1019_PAD_PULL_2: + case RT1019_PAD_PULL_3: + case RT1019_I2C_CTRL_1: + case RT1019_I2C_CTRL_2: + case RT1019_I2C_CTRL_3: + case RT1019_IDS_CTRL: + case RT1019_ASEL_CTRL: + case RT1019_PLL_RESET: + case RT1019_PWR_STRP_1: + case RT1019_PWR_STRP_2: + case RT1019_BEEP_TONE: + case RT1019_SIL_DET_GAT: + case RT1019_CLASSD_TIME: + case RT1019_CLASSD_OCP: + case RT1019_PHASE_SYNC: + case RT1019_STAT_MACH_1: + case RT1019_STAT_MACH_2: + case RT1019_EFF_CTRL: + case RT1019_FS_DET_1: + case RT1019_FS_DET_2: + case RT1019_FS_DET_3: + case RT1019_FS_DET_4: + case RT1019_FS_DET_5: + case RT1019_FS_DET_6: + case RT1019_FS_DET_7: + case RT1019_ANA_CTRL: + case RT1019_DUMMY_A: + case RT1019_DUMMY_B: + case RT1019_DUMMY_C: + case RT1019_DUMMY_D: + case RT1019_ANA_READ: + case RT1019_VER_ID: + case RT1019_CUSTOM_ID: + case RT1019_VEND_ID_1: + case RT1019_VEND_ID_2: + case RT1019_DEV_ID_1: + case RT1019_DEV_ID_2: + case RT1019_TEST_PAD: + case RT1019_SDB_CTRL: + case RT1019_TEST_CTRL_1: + case RT1019_TEST_CTRL_2: + case RT1019_TEST_CTRL_3: + case RT1019_SCAN_MODE: + case RT1019_CLK_TREE_1: + case RT1019_CLK_TREE_2: + case RT1019_CLK_TREE_3: + case RT1019_CLK_TREE_4: + case RT1019_CLK_TREE_5: + case RT1019_CLK_TREE_6: + case RT1019_CLK_TREE_7: + case RT1019_CLK_TREE_8: + case RT1019_CLK_TREE_9: + case RT1019_ASRC_1: + case RT1019_ASRC_2: + case RT1019_ASRC_3: + case RT1019_ASRC_4: + case RT1019_SYS_CLK: + case RT1019_BIAS_CUR_1: + case RT1019_BIAS_CUR_2: + case RT1019_BIAS_CUR_3: + case RT1019_BIAS_CUR_4: + case RT1019_CHOP_CLK_DAC: + case RT1019_CHOP_CLK_ADC: + case RT1019_LDO_CTRL_1: + case RT1019_LDO_CTRL_2: + case RT1019_PM_ANA_1: + case RT1019_PM_ANA_2: + case RT1019_PM_ANA_3: + case RT1019_PLL_1: + case RT1019_PLL_2: + case RT1019_PLL_3: + case RT1019_PLL_INT_1: + case RT1019_PLL_INT_3: + case RT1019_MIXER: + case RT1019_CLD_OUT_1: + case RT1019_CLD_OUT_2: + case RT1019_CLD_OUT_3: + case RT1019_CLD_OUT_4: + case RT1019_CLD_OUT_5: + case RT1019_CLD_OUT_6: + case RT1019_CLS_INT_REG_1: + case RT1019_CLS_INT_REG_2: + case RT1019_CLS_INT_REG_3: + case RT1019_CLS_INT_REG_4: + case RT1019_CLS_INT_REG_5: + case RT1019_CLS_INT_REG_6: + case RT1019_CLS_INT_REG_7: + case RT1019_CLS_INT_REG_8: + case RT1019_CLS_INT_REG_9: + case RT1019_CLS_INT_REG_10: + case RT1019_TDM_1: + case RT1019_TDM_2: + case RT1019_TDM_3: + case RT1019_TDM_4: + case RT1019_TDM_5: + case RT1019_TDM_6: + case RT1019_DVOL_1: + case RT1019_DVOL_2: + case RT1019_DVOL_3: + case RT1019_DVOL_4: + case RT1019_DMIX_MONO_1: + case RT1019_DMIX_MONO_2: + case RT1019_CAL_TOP_1: + case RT1019_CAL_TOP_2: + case RT1019_CAL_TOP_3: + case RT1019_CAL_TOP_4: + case RT1019_CAL_TOP_5: + case RT1019_CAL_TOP_6: + case RT1019_CAL_TOP_7: + case RT1019_CAL_TOP_8: + case RT1019_CAL_TOP_9: + case RT1019_CAL_TOP_10: + case RT1019_CAL_TOP_11: + case RT1019_CAL_TOP_12: + case RT1019_CAL_TOP_13: + case RT1019_CAL_TOP_14: + case RT1019_CAL_TOP_15: + case RT1019_CAL_TOP_16: + case RT1019_CAL_TOP_17: + case RT1019_CAL_TOP_18: + case RT1019_CAL_TOP_19: + case RT1019_CAL_TOP_20: + case RT1019_CAL_TOP_21: + case RT1019_CAL_TOP_22: + case RT1019_MDRE_CTRL_1: + case RT1019_MDRE_CTRL_2: + case RT1019_MDRE_CTRL_3: + case RT1019_MDRE_CTRL_4: + case RT1019_MDRE_CTRL_5: + case RT1019_MDRE_CTRL_6: + case RT1019_MDRE_CTRL_7: + case RT1019_MDRE_CTRL_8: + case RT1019_MDRE_CTRL_9: + case RT1019_MDRE_CTRL_10: + case RT1019_SCC_CTRL_1: + case RT1019_SCC_CTRL_2: + case RT1019_SCC_CTRL_3: + case RT1019_SCC_DUMMY: + case RT1019_SIL_DET_1: + case RT1019_SIL_DET_2: + case RT1019_PWM_DC_DET_1: + case RT1019_PWM_DC_DET_2: + case RT1019_PWM_DC_DET_3: + case RT1019_PWM_DC_DET_4: + case RT1019_BEEP_1: + case RT1019_BEEP_2: + case RT1019_PMC_1: + case RT1019_PMC_2: + case RT1019_PMC_3: + case RT1019_PMC_4: + case RT1019_PMC_5: + case RT1019_PMC_6: + case RT1019_PMC_7: + case RT1019_PMC_8: + case RT1019_PMC_9: + case RT1019_SPKDRC_1: + case RT1019_SPKDRC_2: + case RT1019_SPKDRC_3: + case RT1019_SPKDRC_4: + case RT1019_SPKDRC_5: + case RT1019_SPKDRC_6: + case RT1019_SPKDRC_7: + case RT1019_HALF_FREQ_1: + case RT1019_HALF_FREQ_2: + case RT1019_HALF_FREQ_3: + case RT1019_HALF_FREQ_4: + case RT1019_HALF_FREQ_5: + case RT1019_HALF_FREQ_6: + case RT1019_HALF_FREQ_7: + case RT1019_CUR_CTRL_1: + case RT1019_CUR_CTRL_2: + case RT1019_CUR_CTRL_3: + case RT1019_CUR_CTRL_4: + case RT1019_CUR_CTRL_5: + case RT1019_CUR_CTRL_6: + case RT1019_CUR_CTRL_7: + case RT1019_CUR_CTRL_8: + case RT1019_CUR_CTRL_9: + case RT1019_CUR_CTRL_10: + case RT1019_CUR_CTRL_11: + case RT1019_CUR_CTRL_12: + case RT1019_CUR_CTRL_13: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9525, 75, 0); + +static const char * const rt1019_din_source_select[] = { + "Left", + "Right", + "Left + Right average", +}; + +static SOC_ENUM_SINGLE_DECL(rt1019_mono_lr_sel, RT1019_IDS_CTRL, 0, + rt1019_din_source_select); + +static const struct snd_kcontrol_new rt1019_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", RT1019_DMIX_MONO_1, 0, + 127, 0, dac_vol_tlv), + SOC_ENUM("Mono LR Select", rt1019_mono_lr_sel), +}; + +static int r1019_dac_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); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, RT1019_SDB_CTRL, 0xb); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write(component, RT1019_SDB_CTRL, 0xa); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt1019_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, + r1019_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt1019_dapm_routes[] = { + { "DAC", NULL, "AIFRX" }, + { "SPO", NULL, "DAC" }, +}; + +static int rt1019_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + int pre_div, bclk_ms, frame_size; + unsigned int val_len = 0, sys_div_da_filter = 0; + unsigned int sys_dac_osr = 0, sys_fifo_clk = 0; + unsigned int sys_clk_cal = 0, sys_asrc_in = 0; + + rt1019->lrck = params_rate(params); + pre_div = rl6231_get_clk_info(rt1019->sysclk, rt1019->lrck); + if (pre_div < 0) { + dev_err(component->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + bclk_ms = frame_size > 32; + rt1019->bclk = rt1019->lrck * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt1019->bclk, rt1019->lrck); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (pre_div) { + case 0: + sys_div_da_filter = RT1019_SYS_DIV_DA_FIL_DIV1; + sys_dac_osr = RT1019_SYS_DA_OSR_DIV1; + sys_asrc_in = RT1019_ASRC_256FS_DIV1; + sys_fifo_clk = RT1019_SEL_FIFO_DIV1; + sys_clk_cal = RT1019_SEL_CLK_CAL_DIV1; + break; + case 1: + sys_div_da_filter = RT1019_SYS_DIV_DA_FIL_DIV2; + sys_dac_osr = RT1019_SYS_DA_OSR_DIV2; + sys_asrc_in = RT1019_ASRC_256FS_DIV2; + sys_fifo_clk = RT1019_SEL_FIFO_DIV2; + sys_clk_cal = RT1019_SEL_CLK_CAL_DIV2; + break; + case 3: + sys_div_da_filter = RT1019_SYS_DIV_DA_FIL_DIV4; + sys_dac_osr = RT1019_SYS_DA_OSR_DIV4; + sys_asrc_in = RT1019_ASRC_256FS_DIV4; + sys_fifo_clk = RT1019_SEL_FIFO_DIV4; + sys_clk_cal = RT1019_SEL_CLK_CAL_DIV4; + break; + default: + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len = RT1019_I2S_DL_20; + break; + case 24: + val_len = RT1019_I2S_DL_24; + break; + case 32: + val_len = RT1019_I2S_DL_32; + break; + case 8: + val_len = RT1019_I2S_DL_8; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1019_TDM_2, RT1019_I2S_DL_MASK, + val_len); + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_SEL_FIFO_MASK, sys_fifo_clk); + snd_soc_component_update_bits(component, RT1019_CLK_TREE_2, + RT1019_SYS_DIV_DA_FIL_MASK | RT1019_SYS_DA_OSR_MASK | + RT1019_ASRC_256FS_MASK, sys_div_da_filter | sys_dac_osr | + sys_asrc_in); + snd_soc_component_update_bits(component, RT1019_CLK_TREE_3, + RT1019_SEL_CLK_CAL_MASK, sys_clk_cal); + + return 0; +} + +static int rt1019_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int reg_val = 0, reg_val2 = 0; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val2 |= RT1019_TDM_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT1019_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT1019_I2S_DF_PCM_A_R; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT1019_I2S_DF_PCM_B_R; + break; + + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1019_TDM_2, + RT1019_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT1019_TDM_1, + RT1019_TDM_BCLK_MASK, reg_val2); + + return 0; +} + +static int rt1019_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0; + + if (freq == rt1019->sysclk && clk_id == rt1019->sysclk_src) + return 0; + + switch (clk_id) { + case RT1019_SCLK_S_BCLK: + reg_val |= RT1019_CLK_SYS_PRE_SEL_BCLK; + break; + + case RT1019_SCLK_S_PLL: + reg_val |= RT1019_CLK_SYS_PRE_SEL_PLL; + break; + + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + rt1019->sysclk = freq; + rt1019->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_CLK_SYS_PRE_SEL_MASK, reg_val); + + return 0; +} + +static int rt1019_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + rt1019->pll_in = 0; + rt1019->pll_out = 0; + return 0; + } + + if (source == rt1019->pll_src && freq_in == rt1019->pll_in && + freq_out == rt1019->pll_out) + return 0; + + switch (source) { + case RT1019_PLL_S_BCLK: + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_PLL_SRC_MASK, RT1019_PLL_SRC_SEL_BCLK); + break; + + case RT1019_PLL_S_RC25M: + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_PLL_SRC_MASK, RT1019_PLL_SRC_SEL_RC); + break; + + default: + dev_err(component->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_update_bits(component, RT1019_PWR_STRP_2, + RT1019_AUTO_BITS_SEL_MASK | RT1019_AUTO_CLK_SEL_MASK, + RT1019_AUTO_BITS_SEL_MANU | RT1019_AUTO_CLK_SEL_MANU); + snd_soc_component_update_bits(component, RT1019_PLL_1, + RT1019_PLL_M_MASK | RT1019_PLL_M_BP_MASK | RT1019_PLL_Q_8_8_MASK, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT1019_PLL_M_SFT | + pll_code.m_bp << RT1019_PLL_M_BP_SFT | + ((pll_code.n_code >> 8) & RT1019_PLL_Q_8_8_MASK)); + snd_soc_component_update_bits(component, RT1019_PLL_2, + RT1019_PLL_Q_7_0_MASK, pll_code.n_code & RT1019_PLL_Q_7_0_MASK); + snd_soc_component_update_bits(component, RT1019_PLL_3, + RT1019_PLL_K_MASK, pll_code.k_code); + + rt1019->pll_in = freq_in; + rt1019->pll_out = freq_out; + rt1019->pll_src = source; + + return 0; +} + +static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 4: + val |= RT1019_I2S_TX_4CH; + break; + case 6: + val |= RT1019_I2S_TX_6CH; + break; + case 8: + val |= RT1019_I2S_TX_8CH; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= RT1019_I2S_DL_20; + break; + case 24: + val |= RT1019_I2S_DL_24; + break; + case 32: + val |= RT1019_I2S_DL_32; + break; + case 8: + val |= RT1019_I2S_DL_8; + break; + case 16: + break; + default: + return -EINVAL; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + /* This is an assumption that the system sends stereo audio to the + * amplifier typically. And the stereo audio is placed in slot 0/2/4/6 + * as the starting slot. The users could select the channel from + * L/R/L+R by "Mono LR Select" control. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1019_TDM_3, + RT1019_TDM_I2S_TX_L_DAC1_1_MASK | + RT1019_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1019_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit + 1) << RT1019_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1019_TDM_3, + RT1019_TDM_I2S_TX_L_DAC1_1_MASK | + RT1019_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit - 1) << RT1019_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1019_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1019_TDM_2, + RT1019_I2S_CH_TX_MASK | RT1019_I2S_DF_MASK, val); + +_set_tdm_err_: + return ret; +} + +static int rt1019_probe(struct snd_soc_component *component) +{ + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + + rt1019->component = component; + snd_soc_component_write(component, RT1019_SDB_CTRL, 0xa); + + return 0; +} + +#define RT1019_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT1019_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt1019_aif_dai_ops = { + .hw_params = rt1019_hw_params, + .set_fmt = rt1019_set_dai_fmt, + .set_sysclk = rt1019_set_dai_sysclk, + .set_pll = rt1019_set_dai_pll, + .set_tdm_slot = rt1019_set_tdm_slot, +}; + +static struct snd_soc_dai_driver rt1019_dai[] = { + { + .name = "rt1019-aif", + .id = 0, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1019_STEREO_RATES, + .formats = RT1019_FORMATS, + }, + .ops = &rt1019_aif_dai_ops, + } +}; + +static const struct snd_soc_component_driver soc_component_dev_rt1019 = { + .probe = rt1019_probe, + .controls = rt1019_snd_controls, + .num_controls = ARRAY_SIZE(rt1019_snd_controls), + .dapm_widgets = rt1019_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1019_dapm_widgets), + .dapm_routes = rt1019_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1019_dapm_routes), +}; + +static const struct regmap_config rt1019_regmap = { + .reg_bits = 16, + .val_bits = 8, + .use_single_read = true, + .use_single_write = true, + .max_register = RT1019_CUR_CTRL_13, + .volatile_reg = rt1019_volatile_register, + .readable_reg = rt1019_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt1019_reg, + .num_reg_defaults = ARRAY_SIZE(rt1019_reg), +}; + +static const struct i2c_device_id rt1019_i2c_id[] = { + { "rt1019", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt1019_i2c_id); + +static const struct of_device_id rt1019_of_match[] = { + { .compatible = "realtek,rt1019", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt1019_of_match); + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1019_acpi_match[] = { + { "10EC1019", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1019_acpi_match); +#endif + +static int rt1019_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt1019_priv *rt1019; + int ret; + unsigned int val, val2, dev_id; + + rt1019 = devm_kzalloc(&i2c->dev, sizeof(struct rt1019_priv), + GFP_KERNEL); + if (!rt1019) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt1019); + + rt1019->regmap = devm_regmap_init_i2c(i2c, &rt1019_regmap); + if (IS_ERR(rt1019->regmap)) { + ret = PTR_ERR(rt1019->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt1019->regmap, RT1019_DEV_ID_1, &val); + regmap_read(rt1019->regmap, RT1019_DEV_ID_2, &val2); + dev_id = val << 8 | val2; + if (dev_id != RT1019_DEVICE_ID_VAL && dev_id != RT1019_DEVICE_ID_VAL2) { + dev_err(&i2c->dev, + "Device with ID register 0x%x is not rt1019\n", dev_id); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1019, rt1019_dai, ARRAY_SIZE(rt1019_dai)); +} + +struct i2c_driver rt1019_i2c_driver = { + .driver = { + .name = "rt1019", + .of_match_table = of_match_ptr(rt1019_of_match), + .acpi_match_table = ACPI_PTR(rt1019_acpi_match), + }, + .probe = rt1019_i2c_probe, + .id_table = rt1019_i2c_id, +}; +module_i2c_driver(rt1019_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT1019 driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1019.h b/sound/soc/codecs/rt1019.h new file mode 100644 index 000000000000..46973e71c963 --- /dev/null +++ b/sound/soc/codecs/rt1019.h @@ -0,0 +1,320 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1019.h -- RT1019 ALSA SoC audio amplifier driver + * + * Copyright(c) 2021 Realtek Semiconductor Corp. + */ + +#ifndef __RT1019_H__ +#define __RT1019_H__ + +#define RT1019_DEVICE_ID_VAL 0x1019 +#define RT1019_DEVICE_ID_VAL2 0x6731 + +#define RT1019_RESET 0x0000 +#define RT1019_PAD_DRV_1 0x0002 +#define RT1019_PAD_DRV_2 0x0003 +#define RT1019_PAD_PULL_1 0x0005 +#define RT1019_PAD_PULL_2 0x0006 +#define RT1019_PAD_PULL_3 0x0007 +#define RT1019_I2C_CTRL_1 0x0008 +#define RT1019_I2C_CTRL_2 0x0009 +#define RT1019_I2C_CTRL_3 0x000a +#define RT1019_IDS_CTRL 0x0011 +#define RT1019_ASEL_CTRL 0x0013 +#define RT1019_PLL_RESET 0x0015 +#define RT1019_PWR_STRP_1 0x0017 +#define RT1019_PWR_STRP_2 0x0019 +#define RT1019_BEEP_TONE 0x001b +#define RT1019_SIL_DET_GAT 0x001d +#define RT1019_CLASSD_TIME 0x001f +#define RT1019_CLASSD_OCP 0x0021 +#define RT1019_PHASE_SYNC 0x0023 +#define RT1019_STAT_MACH_1 0x0025 +#define RT1019_STAT_MACH_2 0x0026 +#define RT1019_EFF_CTRL 0x0028 +#define RT1019_FS_DET_1 0x002a +#define RT1019_FS_DET_2 0x002b +#define RT1019_FS_DET_3 0x002c +#define RT1019_FS_DET_4 0x002d +#define RT1019_FS_DET_5 0x002e +#define RT1019_FS_DET_6 0x002f +#define RT1019_FS_DET_7 0x0030 +#define RT1019_ANA_CTRL 0x0053 +#define RT1019_DUMMY_A 0x0055 +#define RT1019_DUMMY_B 0x0056 +#define RT1019_DUMMY_C 0x0057 +#define RT1019_DUMMY_D 0x0058 +#define RT1019_ANA_READ 0x005a +#define RT1019_VER_ID 0x005c +#define RT1019_CUSTOM_ID 0x005d +#define RT1019_VEND_ID_1 0x005e +#define RT1019_VEND_ID_2 0x005f +#define RT1019_DEV_ID_1 0x0061 +#define RT1019_DEV_ID_2 0x0062 +#define RT1019_TEST_PAD 0x0064 +#define RT1019_SDB_CTRL 0x0066 +#define RT1019_TEST_CTRL_1 0x0068 +#define RT1019_TEST_CTRL_2 0x006a +#define RT1019_TEST_CTRL_3 0x006c +#define RT1019_SCAN_MODE 0x006e +#define RT1019_CLK_TREE_1 0x0100 +#define RT1019_CLK_TREE_2 0x0101 +#define RT1019_CLK_TREE_3 0x0102 +#define RT1019_CLK_TREE_4 0x0103 +#define RT1019_CLK_TREE_5 0x0104 +#define RT1019_CLK_TREE_6 0x0105 +#define RT1019_CLK_TREE_7 0x0106 +#define RT1019_CLK_TREE_8 0x0107 +#define RT1019_CLK_TREE_9 0x0108 +#define RT1019_ASRC_1 0x0200 +#define RT1019_ASRC_2 0x0201 +#define RT1019_ASRC_3 0x0202 +#define RT1019_ASRC_4 0x0203 +#define RT1019_SYS_CLK 0x0300 +#define RT1019_BIAS_CUR_1 0x0301 +#define RT1019_BIAS_CUR_2 0x0302 +#define RT1019_BIAS_CUR_3 0x0303 +#define RT1019_BIAS_CUR_4 0x0304 +#define RT1019_CHOP_CLK_DAC 0x0306 +#define RT1019_CHOP_CLK_ADC 0x0308 +#define RT1019_LDO_CTRL_1 0x030a +#define RT1019_LDO_CTRL_2 0x030b +#define RT1019_PM_ANA_1 0x030d +#define RT1019_PM_ANA_2 0x030e +#define RT1019_PM_ANA_3 0x030f +#define RT1019_PLL_1 0x0311 +#define RT1019_PLL_2 0x0312 +#define RT1019_PLL_3 0x0313 +#define RT1019_PLL_INT_1 0x0315 +#define RT1019_PLL_INT_3 0x0318 +#define RT1019_MIXER 0x031a +#define RT1019_CLD_OUT_1 0x031c +#define RT1019_CLD_OUT_2 0x031d +#define RT1019_CLD_OUT_3 0x031e +#define RT1019_CLD_OUT_4 0x031f +#define RT1019_CLD_OUT_5 0x0320 +#define RT1019_CLD_OUT_6 0x0321 +#define RT1019_CLS_INT_REG_1 0x0323 +#define RT1019_CLS_INT_REG_2 0x0324 +#define RT1019_CLS_INT_REG_3 0x0325 +#define RT1019_CLS_INT_REG_4 0x0326 +#define RT1019_CLS_INT_REG_5 0x0327 +#define RT1019_CLS_INT_REG_6 0x0328 +#define RT1019_CLS_INT_REG_7 0x0329 +#define RT1019_CLS_INT_REG_8 0x0330 +#define RT1019_CLS_INT_REG_9 0x0331 +#define RT1019_CLS_INT_REG_10 0x0332 +#define RT1019_TDM_1 0x0400 +#define RT1019_TDM_2 0x0401 +#define RT1019_TDM_3 0x0402 +#define RT1019_TDM_4 0x0403 +#define RT1019_TDM_5 0x0404 +#define RT1019_TDM_6 0x0405 +#define RT1019_DVOL_1 0x0500 +#define RT1019_DVOL_2 0x0501 +#define RT1019_DVOL_3 0x0502 +#define RT1019_DVOL_4 0x0503 +#define RT1019_DMIX_MONO_1 0x0504 +#define RT1019_DMIX_MONO_2 0x0505 +#define RT1019_CAL_TOP_1 0x0600 +#define RT1019_CAL_TOP_2 0x0601 +#define RT1019_CAL_TOP_3 0x0602 +#define RT1019_CAL_TOP_4 0x0603 +#define RT1019_CAL_TOP_5 0x0604 +#define RT1019_CAL_TOP_6 0x0605 +#define RT1019_CAL_TOP_7 0x0606 +#define RT1019_CAL_TOP_8 0x0607 +#define RT1019_CAL_TOP_9 0x0608 +#define RT1019_CAL_TOP_10 0x0609 +#define RT1019_CAL_TOP_11 0x060a +#define RT1019_CAL_TOP_12 0x060b +#define RT1019_CAL_TOP_13 0x060c +#define RT1019_CAL_TOP_14 0x060d +#define RT1019_CAL_TOP_15 0x060e +#define RT1019_CAL_TOP_16 0x060f +#define RT1019_CAL_TOP_17 0x0610 +#define RT1019_CAL_TOP_18 0x0611 +#define RT1019_CAL_TOP_19 0x0612 +#define RT1019_CAL_TOP_20 0x0613 +#define RT1019_CAL_TOP_21 0x0614 +#define RT1019_CAL_TOP_22 0x0615 +#define RT1019_MDRE_CTRL_1 0x0700 +#define RT1019_MDRE_CTRL_2 0x0701 +#define RT1019_MDRE_CTRL_3 0x0702 +#define RT1019_MDRE_CTRL_4 0x0703 +#define RT1019_MDRE_CTRL_5 0x0704 +#define RT1019_MDRE_CTRL_6 0x0705 +#define RT1019_MDRE_CTRL_7 0x0706 +#define RT1019_MDRE_CTRL_8 0x0707 +#define RT1019_MDRE_CTRL_9 0x0708 +#define RT1019_MDRE_CTRL_10 0x0709 +#define RT1019_SCC_CTRL_1 0x0800 +#define RT1019_SCC_CTRL_2 0x0801 +#define RT1019_SCC_CTRL_3 0x0802 +#define RT1019_SCC_DUMMY 0x0803 +#define RT1019_SIL_DET_1 0x0900 +#define RT1019_SIL_DET_2 0x0901 +#define RT1019_PWM_DC_DET_1 0x0a00 +#define RT1019_PWM_DC_DET_2 0x0a01 +#define RT1019_PWM_DC_DET_3 0x0a02 +#define RT1019_PWM_DC_DET_4 0x0a03 +#define RT1019_BEEP_1 0x0b00 +#define RT1019_BEEP_2 0x0b01 +#define RT1019_PMC_1 0x0c00 +#define RT1019_PMC_2 0x0c01 +#define RT1019_PMC_3 0x0c02 +#define RT1019_PMC_4 0x0c03 +#define RT1019_PMC_5 0x0c04 +#define RT1019_PMC_6 0x0c05 +#define RT1019_PMC_7 0x0c06 +#define RT1019_PMC_8 0x0c07 +#define RT1019_PMC_9 0x0c08 +#define RT1019_SPKDRC_1 0x0d00 +#define RT1019_SPKDRC_2 0x0d01 +#define RT1019_SPKDRC_3 0x0d02 +#define RT1019_SPKDRC_4 0x0d03 +#define RT1019_SPKDRC_5 0x0d04 +#define RT1019_SPKDRC_6 0x0d05 +#define RT1019_SPKDRC_7 0x0d06 +#define RT1019_HALF_FREQ_1 0x0e00 +#define RT1019_HALF_FREQ_2 0x0e01 +#define RT1019_HALF_FREQ_3 0x0e02 +#define RT1019_HALF_FREQ_4 0x0e03 +#define RT1019_HALF_FREQ_5 0x0e04 +#define RT1019_HALF_FREQ_6 0x0e05 +#define RT1019_HALF_FREQ_7 0x0e06 +#define RT1019_CUR_CTRL_1 0x0f00 +#define RT1019_CUR_CTRL_2 0x0f01 +#define RT1019_CUR_CTRL_3 0x0f02 +#define RT1019_CUR_CTRL_4 0x0f03 +#define RT1019_CUR_CTRL_5 0x0f04 +#define RT1019_CUR_CTRL_6 0x0f05 +#define RT1019_CUR_CTRL_7 0x0f06 +#define RT1019_CUR_CTRL_8 0x0f07 +#define RT1019_CUR_CTRL_9 0x0f08 +#define RT1019_CUR_CTRL_10 0x0f09 +#define RT1019_CUR_CTRL_11 0x0f0a +#define RT1019_CUR_CTRL_12 0x0f0b +#define RT1019_CUR_CTRL_13 0x0f0c + +/* 0x0019 Power On Strap Control-2 */ +#define RT1019_AUTO_BITS_SEL_MASK (0x1 << 5) +#define RT1019_AUTO_BITS_SEL_AUTO (0x1 << 5) +#define RT1019_AUTO_BITS_SEL_MANU (0x0 << 5) +#define RT1019_AUTO_CLK_SEL_MASK (0x1 << 4) +#define RT1019_AUTO_CLK_SEL_AUTO (0x1 << 4) +#define RT1019_AUTO_CLK_SEL_MANU (0x0 << 4) + +/* 0x0100 Clock Tree Control-1 */ +#define RT1019_CLK_SYS_PRE_SEL_MASK (0x1 << 7) +#define RT1019_CLK_SYS_PRE_SEL_SFT 7 +#define RT1019_CLK_SYS_PRE_SEL_BCLK (0x0 << 7) +#define RT1019_CLK_SYS_PRE_SEL_PLL (0x1 << 7) +#define RT1019_PLL_SRC_MASK (0x1 << 4) +#define RT1019_PLL_SRC_SFT 4 +#define RT1019_PLL_SRC_SEL_BCLK (0x0 << 4) +#define RT1019_PLL_SRC_SEL_RC (0x1 << 4) +#define RT1019_SEL_FIFO_MASK (0x3 << 2) +#define RT1019_SEL_FIFO_DIV1 (0x0 << 2) +#define RT1019_SEL_FIFO_DIV2 (0x1 << 2) +#define RT1019_SEL_FIFO_DIV4 (0x2 << 2) + +/* 0x0101 clock tree control-2 */ +#define RT1019_SYS_DIV_DA_FIL_MASK (0x7 << 5) +#define RT1019_SYS_DIV_DA_FIL_DIV1 (0x2 << 5) +#define RT1019_SYS_DIV_DA_FIL_DIV2 (0x3 << 5) +#define RT1019_SYS_DIV_DA_FIL_DIV4 (0x4 << 5) +#define RT1019_SYS_DA_OSR_MASK (0x3 << 2) +#define RT1019_SYS_DA_OSR_DIV1 (0x0 << 2) +#define RT1019_SYS_DA_OSR_DIV2 (0x1 << 2) +#define RT1019_SYS_DA_OSR_DIV4 (0x2 << 2) +#define RT1019_ASRC_256FS_MASK 0x3 +#define RT1019_ASRC_256FS_DIV1 0x0 +#define RT1019_ASRC_256FS_DIV2 0x1 +#define RT1019_ASRC_256FS_DIV4 0x2 + +/* 0x0102 clock tree control-3 */ +#define RT1019_SEL_CLK_CAL_MASK (0x3 << 6) +#define RT1019_SEL_CLK_CAL_DIV1 (0x0 << 6) +#define RT1019_SEL_CLK_CAL_DIV2 (0x1 << 6) +#define RT1019_SEL_CLK_CAL_DIV4 (0x2 << 6) + +/* 0x0311 PLL-1 */ +#define RT1019_PLL_M_MASK (0xf << 4) +#define RT1019_PLL_M_SFT 4 +#define RT1019_PLL_M_BP_MASK (0x1 << 1) +#define RT1019_PLL_M_BP_SFT 1 +#define RT1019_PLL_Q_8_8_MASK (0x1) + +/* 0x0312 PLL-2 */ +#define RT1019_PLL_Q_7_0_MASK 0xff + +/* 0x0313 PLL-3 */ +#define RT1019_PLL_K_MASK 0x1f + +/* 0x0400 TDM Control-1 */ +#define RT1019_TDM_BCLK_MASK (0x1 << 6) +#define RT1019_TDM_BCLK_NORM (0x0 << 6) +#define RT1019_TDM_BCLK_INV (0x1 << 6) + +/* 0x0401 TDM Control-2 */ +#define RT1019_I2S_CH_TX_MASK (0x3 << 6) +#define RT1019_I2S_CH_TX_SFT 6 +#define RT1019_I2S_TX_2CH (0x0 << 6) +#define RT1019_I2S_TX_4CH (0x1 << 6) +#define RT1019_I2S_TX_6CH (0x2 << 6) +#define RT1019_I2S_TX_8CH (0x3 << 6) +#define RT1019_I2S_DF_MASK (0x7 << 3) +#define RT1019_I2S_DF_SFT 3 +#define RT1019_I2S_DF_I2S (0x0 << 3) +#define RT1019_I2S_DF_LEFT (0x1 << 3) +#define RT1019_I2S_DF_PCM_A_R (0x2 << 3) +#define RT1019_I2S_DF_PCM_B_R (0x3 << 3) +#define RT1019_I2S_DF_PCM_A_F (0x6 << 3) +#define RT1019_I2S_DF_PCM_B_F (0x7 << 3) +#define RT1019_I2S_DL_MASK 0x7 +#define RT1019_I2S_DL_SFT 0 +#define RT1019_I2S_DL_16 0x0 +#define RT1019_I2S_DL_20 0x1 +#define RT1019_I2S_DL_24 0x2 +#define RT1019_I2S_DL_32 0x3 +#define RT1019_I2S_DL_8 0x4 + +/* TDM1 Control-3 (0x0402) */ +#define RT1019_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 4) +#define RT1019_TDM_I2S_TX_R_DAC1_1_MASK 0x7 +#define RT1019_TDM_I2S_TX_L_DAC1_1_SFT 4 +#define RT1019_TDM_I2S_TX_R_DAC1_1_SFT 0 + +/* System Clock Source */ +enum { + RT1019_SCLK_S_BCLK, + RT1019_SCLK_S_PLL, +}; + +/* PLL1 Source */ +enum { + RT1019_PLL_S_BCLK, + RT1019_PLL_S_RC25M, +}; + +enum { + RT1019_AIF1, + RT1019_AIFS +}; + +struct rt1019_priv { + struct snd_soc_component *component; + struct regmap *regmap; + int sysclk; + int sysclk_src; + int lrck; + int bclk; + int pll_src; + int pll_in; + int pll_out; + unsigned int bclk_ratio; +}; + +#endif /* __RT1019_H__ */ -- 2.29.0

3 2

[PATCH] soundwire: intel: move to auxiliary bus
by Bard Liao 26 Mar '21

26 Mar '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Now that the auxiliary_bus exists, there's no reason to use platform devices as children of a PCI device any longer. This patch refactors the code by extending a basic auxiliary device with Intel link-specific structures that need to be passed between controller and link levels. This refactoring is much cleaner with no need for cross-pointers between device and link structures. Note that the auxiliary bus API has separate init and add steps, which requires more attention in the error unwinding paths. The main loop needs to deal with kfree() and auxiliary_device_uninit() for the current iteration before jumping to the common label which releases everything allocated in prior iterations. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 52 ++++---- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 190 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 175 insertions(+), 88 deletions(-) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 016e74230bb7..2b7795233282 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -25,6 +25,7 @@ config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE select SOUNDWIRE_GENERIC_ALLOCATION + select AUXILIARY_BUS depends on ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index d2254ee2fee2..039a101982c9 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -11,7 +11,7 @@ #include <linux/module.h> #include <linux/interrupt.h> #include <linux/io.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <sound/pcm_params.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -1331,9 +1331,10 @@ static int intel_init(struct sdw_intel *sdw) /* * probe and init */ -static int intel_master_probe(struct platform_device *pdev) +static int intel_link_probe(struct auxiliary_device *auxdev, const struct auxiliary_device_id *id) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); struct sdw_intel *sdw; struct sdw_cdns *cdns; struct sdw_bus *bus; @@ -1346,14 +1347,14 @@ static int intel_master_probe(struct platform_device *pdev) cdns = &sdw->cdns; bus = &cdns->bus; - sdw->instance = pdev->id; - sdw->link_res = dev_get_platdata(dev); + sdw->instance = auxdev->id; + sdw->link_res = &ldev->link_res; cdns->dev = dev; cdns->registers = sdw->link_res->registers; cdns->instance = sdw->instance; cdns->msg_count = 0; - bus->link_id = pdev->id; + bus->link_id = auxdev->id; sdw_cdns_probe(cdns); @@ -1386,10 +1387,10 @@ static int intel_master_probe(struct platform_device *pdev) return 0; } -int intel_master_startup(struct platform_device *pdev) +int intel_link_startup(struct auxiliary_device *auxdev) { struct sdw_cdns_stream_config config; - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1526,9 +1527,9 @@ int intel_master_startup(struct platform_device *pdev) return ret; } -static int intel_master_remove(struct platform_device *pdev) +static void intel_link_remove(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1544,19 +1545,17 @@ static int intel_master_remove(struct platform_device *pdev) snd_soc_unregister_component(dev); } sdw_bus_master_delete(bus); - - return 0; } -int intel_master_process_wakeen_event(struct platform_device *pdev) +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_intel *sdw; struct sdw_bus *bus; void __iomem *shim; u16 wake_sts; - sdw = platform_get_drvdata(pdev); + sdw = dev_get_drvdata(dev); bus = &sdw->cdns.bus; if (bus->prop.hw_disabled) { @@ -1978,17 +1977,22 @@ static const struct dev_pm_ops intel_pm = { SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) }; -static struct platform_driver sdw_intel_drv = { - .probe = intel_master_probe, - .remove = intel_master_remove, +static const struct auxiliary_device_id intel_link_id_table[] = { + { .name = "soundwire_intel.link" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, intel_link_id_table); + +static struct auxiliary_driver sdw_intel_drv = { + .probe = intel_link_probe, + .remove = intel_link_remove, .driver = { - .name = "intel-sdw", + /* auxiliary_driver_register() sets .name to be the modname */ .pm = &intel_pm, - } + }, + .id_table = intel_link_id_table }; - -module_platform_driver(sdw_intel_drv); +module_auxiliary_driver(sdw_intel_drv); MODULE_LICENSE("Dual BSD/GPL"); -MODULE_ALIAS("platform:intel-sdw"); -MODULE_DESCRIPTION("Intel Soundwire Master Driver"); +MODULE_DESCRIPTION("Intel Soundwire Link Driver"); diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 06bac8ba14e9..0b47b148da3f 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -7,7 +7,6 @@ /** * struct sdw_intel_link_res - Soundwire Intel link resource structure, * typically populated by the controller driver. - * @pdev: platform_device * @mmio_base: mmio base of SoundWire registers * @registers: Link IO registers base * @shim: Audio shim pointer @@ -23,7 +22,6 @@ * @list: used to walk-through all masters exposed by the same controller */ struct sdw_intel_link_res { - struct platform_device *pdev; void __iomem *mmio_base; /* not strictly needed, useful for debug */ void __iomem *registers; void __iomem *shim; @@ -48,7 +46,15 @@ struct sdw_intel { #endif }; -int intel_master_startup(struct platform_device *pdev); -int intel_master_process_wakeen_event(struct platform_device *pdev); +int intel_link_startup(struct auxiliary_device *auxdev); +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev); + +struct sdw_intel_link_dev { + struct auxiliary_device auxdev; + struct sdw_intel_link_res link_res; +}; + +#define auxiliary_dev_to_sdw_intel_link_dev(auxiliary_dev) \ + container_of(auxiliary_dev, struct sdw_intel_link_dev, auxdev) #endif /* __SDW_INTEL_LOCAL_H */ diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 05b726cdfebc..3cb74060ccae 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -12,7 +12,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/module.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <linux/pm_runtime.h> #include <linux/soundwire/sdw_intel.h> #include "cadence_master.h" @@ -24,28 +24,65 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 +static void intel_link_auxdev_release(struct device *dev) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); + + kfree(ldev); +} + +static int intel_link_dev_init(struct sdw_intel_link_dev *ldev, + struct device *parent, + struct fwnode_handle *fwnode, + const char *name, + int link_id) +{ + struct auxiliary_device *auxdev; + int ret; + + auxdev = &ldev->auxdev; + auxdev->name = name; + auxdev->dev.parent = parent; + auxdev->dev.fwnode = fwnode; + auxdev->dev.release = intel_link_auxdev_release; + + /* we don't use an IDA since we already have a link ID */ + auxdev->id = link_id; + + ret = auxiliary_device_init(auxdev); + if (ret < 0) + dev_err(parent, "failed to initialize link dev %s link_id %d\n", + name, link_id); + + return ret; +} + +static void intel_link_dev_unregister(struct sdw_intel_link_dev *ldev) +{ + auxiliary_device_delete(&ldev->auxdev); + auxiliary_device_uninit(&ldev->auxdev); +} + static int sdw_intel_cleanup(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link = ctx->links; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!link) - return 0; - link_mask = ctx->link_mask; - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - if (link->pdev) { - pm_runtime_disable(&link->pdev->dev); - platform_device_unregister(link->pdev); - } + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) - pm_runtime_put_noidle(link->dev); + pm_runtime_disable(&ldev->auxdev.dev); + intel_link_dev_unregister(ldev); + + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); } return 0; @@ -91,9 +128,8 @@ EXPORT_SYMBOL_NS(sdw_intel_thread, SOUNDWIRE_INTEL_INIT); static struct sdw_intel_ctx *sdw_intel_probe_controller(struct sdw_intel_res *res) { - struct platform_device_info pdevinfo; - struct platform_device *pdev; struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; struct sdw_intel_ctx *ctx; struct acpi_device *adev; struct sdw_slave *slave; @@ -103,6 +139,7 @@ static struct sdw_intel_ctx int num_slaves = 0; int count; int i; + int ret; if (!res) return NULL; @@ -116,35 +153,72 @@ static struct sdw_intel_ctx count = res->count; dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - ctx = devm_kzalloc(&adev->dev, sizeof(*ctx), GFP_KERNEL); + /* + * we need to alloc/free memory manually and can't use devm: + * this routine may be called from a workqueue, and not from + * the parent .probe. + * If devm_ was used, the memory might never be freed on errors. + */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return NULL; ctx->count = count; - ctx->links = devm_kcalloc(&adev->dev, ctx->count, - sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) + + /* + * allocate the array of pointers. The link-specific data is allocated + * as part of the first loop below and released with the auxiliary_device_uninit(). + * If some links are disabled, the link pointer will remain NULL. Given that the + * number of links is small, this is simpler than using a list to keep track of links. + */ + ctx->ldev = kcalloc(ctx->count, sizeof(*ctx->ldev), GFP_KERNEL); + if (!ctx->ldev) { + kfree(ctx); return NULL; + } - ctx->count = count; ctx->mmio_base = res->mmio_base; ctx->link_mask = res->link_mask; ctx->handle = res->handle; mutex_init(&ctx->shim_lock); - link = ctx->links; link_mask = ctx->link_mask; INIT_LIST_HEAD(&ctx->link_list); - /* Create SDW Master devices */ - for (i = 0; i < count; i++, link++) { - if (!(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); + for (i = 0; i < count; i++) { + if (!(link_mask & BIT(i))) continue; + + /* Alloc and init devices */ + ldev = kzalloc(sizeof(*ldev), GFP_KERNEL); + if (!ldev) + goto err; + + /* + * keep a handle on the allocated memory, to be used in all other functions. + * Since the same pattern is used to skip links that are not enabled, there is + * not need to check if ctx->ldev[i] is NULL later on. + */ + ctx->ldev[i] = ldev; + + ret = intel_link_dev_init(ldev, + res->parent, + acpi_fwnode_handle(adev), + "link", /* prefixed by core with "soundwire_intel." */ + i); + if (ret < 0) { + /* + * We need to free the memory allocated in the current iteration + */ + kfree(ldev); + + goto err; } + link = &ldev->link_res; + + /* now set link information */ link->mmio_base = res->mmio_base; link->registers = res->mmio_base + SDW_LINK_BASE + (SDW_LINK_SIZE * i); @@ -159,24 +233,16 @@ static struct sdw_intel_ctx link->shim_mask = &ctx->shim_mask; link->link_mask = link_mask; - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "intel-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - pdevinfo.data = link; - pdevinfo.size_data = sizeof(*link); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); + ret = auxiliary_device_add(&ldev->auxdev); + if (ret < 0) { + dev_err(&adev->dev, "failed to add link dev %s link_id %d\n", + ldev->auxdev.name, i); + /* ldev will be freed with the put_device() and .release sequence */ + auxiliary_device_uninit(&ldev->auxdev); goto err; } - link->pdev = pdev; - link->cdns = platform_get_drvdata(pdev); + + link->cdns = dev_get_drvdata(&ldev->auxdev.dev); list_add_tail(&link->list, &ctx->link_list); bus = &link->cdns->bus; @@ -185,8 +251,7 @@ static struct sdw_intel_ctx num_slaves++; } - ctx->ids = devm_kcalloc(&adev->dev, num_slaves, - sizeof(*ctx->ids), GFP_KERNEL); + ctx->ids = kcalloc(num_slaves, sizeof(*ctx->ids), GFP_KERNEL); if (!ctx->ids) goto err; @@ -204,8 +269,14 @@ static struct sdw_intel_ctx return ctx; err: - ctx->count = i; - sdw_intel_cleanup(ctx); + while (i--) { + if (!(link_mask & BIT(i))) + continue; + ldev = ctx->ldev[i]; + intel_link_dev_unregister(ldev); + } + kfree(ctx->ldev); + kfree(ctx); return NULL; } @@ -213,7 +284,7 @@ static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { struct acpi_device *adev; - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; @@ -232,27 +303,28 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) return -EINVAL; } - if (!ctx->links) + if (!ctx->ldev) return -EINVAL; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_startup(link->pdev); + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) { + intel_link_startup(&ldev->auxdev); + + if (!ldev->link_res.clock_stop_quirks) { /* * we need to prevent the parent PCI device * from entering pm_runtime suspend, so that * power rails to the SoundWire IP are not * turned off. */ - pm_runtime_get_noresume(link->dev); + pm_runtime_get_noresume(ldev->link_res.dev); } } @@ -297,27 +369,31 @@ EXPORT_SYMBOL_NS(sdw_intel_startup, SOUNDWIRE_INTEL_INIT); void sdw_intel_exit(struct sdw_intel_ctx *ctx) { sdw_intel_cleanup(ctx); + kfree(ctx->ids); + kfree(ctx->ldev); + kfree(ctx); } EXPORT_SYMBOL_NS(sdw_intel_exit, SOUNDWIRE_INTEL_INIT); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!ctx->links) + if (!ctx->ldev) return; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_process_wakeen_event(link->pdev); + ldev = ctx->ldev[i]; + + intel_link_process_wakeen_event(&ldev->auxdev); } } EXPORT_SYMBOL_NS(sdw_intel_process_wakeen_event, SOUNDWIRE_INTEL_INIT); diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3a5446ac014a..1ebea7764011 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -58,7 +58,7 @@ struct sdw_intel_acpi_info { u32 link_mask; }; -struct sdw_intel_link_res; +struct sdw_intel_link_dev; /* Intel clock-stop/pm_runtime quirk definitions */ @@ -109,7 +109,7 @@ struct sdw_intel_slave_id { * Controller * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle - * @links: information for each link (controller-specific and kept + * @ldev: information for each link (controller-specific and kept * opaque here) * @ids: array of slave_id, representing Slaves exposed across all enabled * links @@ -123,7 +123,7 @@ struct sdw_intel_ctx { u32 link_mask; int num_slaves; acpi_handle handle; - struct sdw_intel_link_res *links; + struct sdw_intel_link_dev **ldev; struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ -- 2.17.1

4 13

[RESEND PATCH 00/11] soundwire: some cleanup patches
by Bard Liao 26 Mar '21

26 Mar '21

To make soundwire driver more decent and less Cppcheck complaint. Pierre-Louis Bossart (11): soundwire: bus: use correct driver name in error messages soundwire: bus: test read status soundwire: bus: use consistent tests for return values soundwire: bus: demote clock stop prepare log to dev_dbg() soundwire: bus: uniquify dev_err() for SCP_INT access soundwire: bus: remove useless initialization soundwire: generic_bandwidth_allocation: remove useless init soundwire: intel: remove useless readl soundwire: qcom: check of_property_read status soundwire: stream: remove useless initialization soundwire: stream: remove useless bus initializations drivers/soundwire/bus.c | 54 +++++++++++-------- drivers/soundwire/bus_type.c | 15 ++++-- .../soundwire/generic_bandwidth_allocation.c | 4 +- drivers/soundwire/intel.c | 2 - drivers/soundwire/qcom.c | 3 ++ drivers/soundwire/stream.c | 8 +-- 6 files changed, 52 insertions(+), 34 deletions(-) -- 2.17.1

1 11

[PATCH V4] ALSA: pcm: Fix couple of typos
by Bhaskar Chowdhury 26 Mar '21

26 Mar '21

s/unconditonally/unconditionally/ s/succesful/successful/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Changes from V3: Yet another try to make it work sound/core/pcm_native.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..8dbe86cf2e4f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1425,7 +1425,7 @@ static int snd_pcm_do_stop(struct snd_pcm_substream *substream, substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP); substream->runtime->stop_operating = true; } - return 0; /* unconditonally stop all substreams */ + return 0; /* unconditionally stop all substreams */ } static void snd_pcm_post_stop(struct snd_pcm_substream *substream, @@ -1469,7 +1469,7 @@ EXPORT_SYMBOL(snd_pcm_stop); * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. * - * Return: Zero if succesful, or a negative error code. + * Return: Zero if successful, or a negative error code. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) { -- 2.26.2

2 1

[PATCH v3 1/2] ALSA: usb-audio: Carve out connector value checking into a helper
by Kai-Heng Feng 26 Mar '21

26 Mar '21

This is preparation for next patch, no functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v3: - No change. v2: - Only return early when ret < 0. sound/usb/mixer.c | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b004b2e63a5d..5a2d9a768f70 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1446,13 +1446,11 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } -/* get the connectors status and report it as boolean type */ -static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int get_connector_value(struct usb_mixer_elem_info *cval, + char *name, int *val) { - struct usb_mixer_elem_info *cval = kcontrol->private_data; struct snd_usb_audio *chip = cval->head.mixer->chip; - int idx = 0, validx, ret, val; + int idx = 0, validx, ret; validx = cval->control << 8 | 0; @@ -1467,21 +1465,24 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac2_conn, sizeof(uac2_conn)); - val = !!uac2_conn.bNrChannels; + if (val) + *val = !!uac2_conn.bNrChannels; } else { /* UAC_VERSION_3 */ struct uac3_insertion_ctl_blk uac3_conn; ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac3_conn, sizeof(uac3_conn)); - val = !!uac3_conn.bmConInserted; + if (val) + *val = !!uac3_conn.bmConInserted; } snd_usb_unlock_shutdown(chip); if (ret < 0) { - if (strstr(kcontrol->id.name, "Speaker")) { - ucontrol->value.integer.value[0] = 1; + if (name && strstr(name, "Speaker")) { + if (val) + *val = 1; return 0; } error: @@ -1491,6 +1492,21 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, return filter_error(cval, ret); } + return ret; +} + +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + int ret, val; + + ret = get_connector_value(cval, kcontrol->id.name, &val); + + if (ret < 0) + return ret; + ucontrol->value.integer.value[0] = val; return 0; } -- 2.30.2

2 3

Re: [PATCH] ALSA: usb-audio: Don't store returned value and check for errors
by Takashi Iwai 26 Mar '21

26 Mar '21

On Thu, 25 Mar 2021 09:36:41 +0100, Muhammad Usama Anjum wrote: > > The returned value by create_autodetect_quirk isn't being used other > than in the success check on next line. Remove the return value > assignement. Check for error values instead of success check. > > Signed-off-by: Muhammad Usama Anjum <musamaanjum(a)gmail.com> > --- > sound/usb/quirks.c | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 9e5e37eff10e..5ce5f4ecb9d0 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -425,9 +425,9 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, > USB_CLASS_VENDOR_SPEC) > continue; > > - err = create_autodetect_quirk(chip, iface, driver); > - if (err >= 0) > - usb_driver_claim_interface(driver, iface, (void *)-1L); > + if (create_autodetect_quirk(chip, iface, driver) < 0) > + continue; > + usb_driver_claim_interface(driver, iface, (void *)-1L); Hm, basically this change is only for readability (the code optimization makes almost no sense), and the readability itself isn't improved significantly by this change, either. That is, I'm fine to take this, but OTOH, I see no compelling reason to do so... thanks, Takashi

1 0

[PATCH v4 2/2] ASoC: rt715:add micmute led state control supports
by Perry Yuan 26 Mar '21

26 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing When micmute hotkey pressed by user, soft mute will need to be enabled firstly in case of pop noise, and codec driver need to react to mic mute event to EC(embedded controller) notifying that SW mute is completed Then EC will do the hardware mute physically within the timeout reached This patch allow codec rt715 and rt715 sdca driver to change the local micmute led state. Dell privacy led trigger driver will ack EC when micmute key pressed through this micphone led control interface like hda_generic provided ACPI method defined in dell-privacy micmute led trigger will be called for notifying the EC that software mute has been completed, then hardware audio circuit solution controlled by EC will switch the audio input source off/on Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> -------- v3 -> v4: * remove unused debug log * remove compile flag of DELL privacy * move the micmute_led to local from rt715_priv * when Jaroslav upstream his gerneric LED trigger driver,I will rebase this patch,please consider merge this at first https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ v2 -> v3: * simplify the patch to reuse some val value * add more detail to the commit info v1 -> v2: * fix some format issue -------- --- sound/soc/codecs/rt715-sdca.c | 12 ++++++++++++ sound/soc/codecs/rt715.c | 12 ++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index b43ac8559e45..816348ae11a1 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -12,6 +12,7 @@ #include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -269,6 +270,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, unsigned int reg = mc->reg; unsigned int max = mc->max; int err; + bool micmute_led; val = ucontrol->value.integer.value[0]; if (invert) @@ -287,6 +289,16 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, return err; } + /* Micmute LED state changed by muted/unmute switch */ + if (mc->invert) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) { + micmute_led = LED_OFF; + } else { + micmute_led = LED_ON; + } + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } + return 0; } diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index cdcba70146da..db2c0d2ff9d2 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -88,6 +89,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt; + bool micmute_led; if (strstr(ucontrol->id.name, "Main Capture Switch") || strstr(ucontrol->id.name, "Main Capture Volume")) @@ -95,6 +97,16 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, else loop_cnt = 1; + /* Micmute LED state changed by muted/unmute switch */ + if (mc->invert) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) { + micmute_led = LED_OFF; + } else { + micmute_led = LED_ON; + } + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } + for (j = 0; j < loop_cnt; j++) { /* Can't use update bit function, so read the original value first */ if (loop_cnt == 1) { -- 2.25.1

6 10

[PATCH V3] ALSA: pcm: Fix couple of typos
by Bhaskar Chowdhury 26 Mar '21

26 Mar '21

s/unconditonally/unconditionally/ s/succesful/successful/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Changes from V2: Takashi pointed out that the patch was not applicable due to some unwanted stuff get into it. Resending it with the new patch creation. sound/core/pcm_native.c | 82 ++++++++++++++++++++--------------------- 1 file changed, 41 insertions(+), 41 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..afb670d7fd53 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1425,7 +1425,7 @@ static int snd_pcm_do_stop(struct snd_pcm_substream *substream, substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP); substream->runtime->stop_operating = true; } - return 0; /* unconditonally stop all substreams */ + return 0; /* unconditionally stop all substreams */ } static void snd_pcm_post_stop(struct snd_pcm_substream *substream, @@ -1469,7 +1469,7 @@ EXPORT_SYMBOL(snd_pcm_stop); * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. * - * Return: Zero if succesful, or a negative error code. + * Return: Zero if successful, or a negative error code. -- 2.30.1

3 6

[PATCH v3 0/7] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 26 Mar '21

26 Mar '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Even though this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Am planning to send support for MBHC once this driver gets accepted! Thanks, srini Many thanks for reviewing v2. Changes since v2: - fixed dt_binding_check error Srinivas Kandagatla (7): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x.yaml | 165 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 350 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 291 ++ sound/soc/codecs/wcd938x.c | 3623 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 676 +++ 8 files changed, 5122 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

3 16

[PATCH 0/1] sound: rawmidi: Add framing mode
by David Henningsson 25 Mar '21

25 Mar '21

Hi, When writing an application that records midi (and e g saves it to disk), ultra-low latency isn't really needed, all we need to know is exactly when the midi message came in. The application can then wake up once a second or so, to write the incoming data, including an accurate timestamp for every event, to disk. As far as I can see, the rawmidi interface does not support such a feature at all. There is a snd_rawmidi_status syscall, but its timestamp field is not even filled by the kernel (!). But even if that was fixed, it would not fix the problem as there could be several midi events in the buffer with different timestamps. You could use the seq interface, it does support timestamps, but I can see at least two potential problems with this: First, the seq code runs in a work queue, not in the actual IRQ. This means that midi event is timestamped too late, especially so if the work is delayed for some reason. Second, seq hard-codes the timestamp type to monotonic - there is no monotonic_raw, so the timestamp would be affected by NTP changes. Also, the timespec uses 32-bit for sec and nsec, but I suspect this is less of a problem (unless people constantly record midi for sixty years or so...). So here's a patch that adds a new "framing" mode that stuffs all MIDI data into 16 byte frames, 8 bytes of timestamp, one byte length, up to seven bytes of data. I'll follow up with an alsa-lib patch if this gets merged. David Henningsson (1): sound: rawmidi: Add framing mode include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 18 ++++++++++++++- sound/core/rawmidi.c | 45 ++++++++++++++++++++++++++++++++++++- 3 files changed, 62 insertions(+), 2 deletions(-) -- 2.25.1

2 4

[PATCH v3] sound: rawmidi: Add framing mode
by David Henningsson 25 Mar '21

25 Mar '21

This commit adds a new framing mode that frames all MIDI data into 16-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Signed-off-by: David Henningsson <coding(a)diwic.se> --- Changes in v3: * Bumped protocol version * More checks for invalid combinations in set_params * Frame: Timestamp is 64 bit for seconds and 3 bytes for data * Sets clock and framing individually. (Clock_type is unsigned char instead of clockid_t to avoid padding issues.) * Supports both monotonic and monotonic_raw timestamps * Moved spin lock to after timestamp retreival. Both to decrease time spent in spin lock, and to get more accurate timestamp in case the spin lock is actually busy. include/sound/rawmidi.h | 2 ++ include/uapi/sound/asound.h | 21 +++++++++++-- sound/core/rawmidi.c | 63 +++++++++++++++++++++++++++++++++++-- 3 files changed, 82 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..b0057a193c31 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame input data */ + clockid_t clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..a965a975aec3 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,29 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 3 + +struct snd_rawmidi_framing_tstamp { + unsigned long long tv_sec; /* seconds */ + unsigned int tv_nsec; /* nanoseconds */ + unsigned char length; + unsigned char data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char clock_type; /* Type of clock to use for framing, same as clockid_t */ + unsigned char reserved[14]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..269363286b3b 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -683,6 +683,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (params->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP && params->buffer_size & 0xf) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +722,16 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + if (params->framing) { + if (params->framing > SNDRV_RAWMIDI_FRAMING_LAST) + return -EINVAL; + /* framing requires a valid clock type */ + if (params->clock_type != CLOCK_MONOTONIC_RAW && params->clock_type != CLOCK_MONOTONIC) + return -EINVAL; + } snd_rawmidi_drain_input(substream); + substream->framing = params->framing; + substream->clock_type = params->clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +974,45 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp frame; + struct snd_rawmidi_framing_tstamp *dest_ptr; + + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + frame.tv_sec = tstamp->tv_sec; + frame.tv_nsec = tstamp->tv_nsec; + + if (snd_BUG_ON(runtime->hw_ptr & 0xf || runtime->buffer_size & 0xf || frame_size != 0x10)) + return -EINVAL; + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + return dest_frames * frame_size; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1027,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64; int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1038,15 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } - spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + spin_lock_irqsave(&runtime->lock, flags); + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ + spin_lock_irqsave(&runtime->lock, flags); substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -999,6 +1057,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, runtime->xruns++; } } else { + spin_lock_irqsave(&runtime->lock, flags); substream->bytes += count; count1 = runtime->buffer_size - runtime->hw_ptr; if (count1 > count) -- 2.25.1

1 0

Patch for some Roland devices' USB digital audio
by Lucas 25 Mar '21

25 Mar '21

A couple days ago, I found this excellent informational post, and that Mike Oliphant's patch is in mainline now: http://blog.nostatic.org/2020/01/getting-boss-gt-1-effects-processor-to.html Since I was fairly certain it applies to the Roland devices I own, I added my USB IDs. Wonderfully, now USB audio capture seems to be working perfectly from the three devices I can test: the INTEGRA-7, Boutique D-05, & VG-99! I don't know if the D-05 and VG-99 have digital audio playback/output, as I only tried it with the INTEGRA-7. It does work, but only when the INTEGRA-7's "Sampling Rate" is set to "44.1 kHz" and "Ext Part Source Select" is set to "USB AUDIO". I searched a bit for other Roland devices people have had USB digital audio issues with over the years, and took the liberty of adding them to the patch too. Likely there are still many more, especially in the Boutique line, but I couldn't find their USB IDs. I hope this can be added to the mainline kernel too, as this support should help many that have struggled. I, certainly, have been hoping somebody would figure this out for quite a while, and even bought a USB through/monitoring device to try to develop it myself (sadly, the software supporting that has its own bugs, so I probably should've just used wireshark). Suffice it to say, I'm quite happy to have this figured out, and to not have to delve into stuff I know nothing about! :) Thanks, Lucas

1 0

[PATCH v5 0/6] Add audio driver base on rpmsg on i.MX platform
by Shengjiu Wang 25 Mar '21

25 Mar '21

On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core, Linux is running on A core, RTOS is running on M core. The audio hardware device can be controlled by Cortex-M device, So audio playback/capture can be handled by M core. Rpmsg is the interface for sending and receiving msg to and from M core, that we can create a virtual sound on Cortex-A core side. A core will tell the Cortex-M core sound format/rate/channel, where is the data buffer, what is the period size, when to start, when to stop and when suspend or resume happen, each of this behavior there is defined rpmsg command. Especially we designed the low power audio case, that is to allocate a large buffer and fill the data, then Cortex-A core can go to sleep mode, Cortex-M core continue to play the sound, when the buffer is consumed, Cortex-M core will trigger the Cortex-A core to wakeup to fill data. changes in v5: - remove unneeded property in binding doc and driver - update binding doc according to Rob's comments. - Fix link issue reported by kernel test robot changes in v4: - remove the sound card node, merge the property to cpu dai node according to Rob's comments. - sound card device will be registered by cpu dai driver. - Fix do_div issue reported by kernel test robot changes in v3: - add local refcount for clk enablement in hw_params() - update the document according Rob's comments changes in v2: - update codes and comments according to Mark's comments Shengjiu Wang (6): ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg audio device ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg .../devicetree/bindings/sound/fsl,rpmsg.yaml | 108 +++ include/sound/soc-component.h | 3 + sound/soc/fsl/Kconfig | 30 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_rpmsg.c | 279 ++++++ sound/soc/fsl/fsl_rpmsg.h | 35 + sound/soc/fsl/imx-audio-rpmsg.c | 140 +++ sound/soc/fsl/imx-pcm-rpmsg.c | 918 ++++++++++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++ sound/soc/fsl/imx-rpmsg.c | 150 +++ sound/soc/soc-component.c | 14 + sound/soc/soc-pcm.c | 2 + 12 files changed, 2197 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c -- 2.27.0

2 7

[PATCH] ASoC: rt1015: Add bclk detection and dc detection
by Jack Yu 25 Mar '21

25 Mar '21

Add bclk detection and dc detection before playback. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 18 +++++++++++++++++- sound/soc/codecs/rt1015.h | 10 ++++++++++ 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 10a7b7745f4b..9238f12999aa 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -669,8 +669,23 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + unsigned int ret, ret2; switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = snd_soc_component_read(component, RT1015_CLK_DET); + ret2 = snd_soc_component_read(component, RT1015_SPK_DC_DETECT1); + if (!((ret >> 15) & 0x1)) { + snd_soc_component_update_bits(component, RT1015_CLK_DET, + RT1015_EN_BCLK_DET_MASK, RT1015_EN_BCLK_DET); + dev_dbg(component->dev, "BCLK Detection Enabled.\n"); + } + if (!((ret2 >> 12) & 0x1)) { + snd_soc_component_update_bits(component, RT1015_SPK_DC_DETECT1, + RT1015_EN_CLA_D_DC_DET_MASK, RT1015_EN_CLA_D_DC_DET); + dev_dbg(component->dev, "Class-D DC Detection Enabled.\n"); + } + break; case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); @@ -690,7 +705,8 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, - rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU), + rt1015_amp_drv_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPO"), }; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 2aeaf65ba793..14344532048e 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -209,6 +209,11 @@ #define RT1015_PLL_K_MASK (RT1015_PLL_K_MAX) #define RT1015_PLL_K_SFT 0 +/* 0x0020 */ +#define RT1015_EN_BCLK_DET_MASK (0x1 << 15) +#define RT1015_EN_BCLK_DET (0x1 << 15) +#define RT1015_DIS_BCLK_DET (0x0 << 15) + /* 0x007a */ #define RT1015_ID_MASK 0xff #define RT1015_ID_VERA 0x0 @@ -374,6 +379,11 @@ #define RT1015_PWR_SWR (0x1 << 12) #define RT1015_PWR_SWR_BIT 12 +/* 0x0519 */ +#define RT1015_EN_CLA_D_DC_DET_MASK (0x1 << 12) +#define RT1015_EN_CLA_D_DC_DET (0x1 << 12) +#define RT1015_DIS_CLA_D_DC_DET (0x0 << 12) + /* 0x1300 */ #define RT1015_PWR_CLSD (0x1 << 12) #define RT1015_PWR_CLSD_BIT 12 -- 2.29.0

2 1

[PATCH 1/2] ASoC: fsl-asoc-card: Add support for WM8958 codec
by Shengjiu Wang 25 Mar '21

25 Mar '21

WM8958 codec is used on some i.MX based platform. So add it support in this generic driver. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/Kconfig | 2 ++ sound/soc/fsl/fsl-asoc-card.c | 17 +++++++++++++++-- 2 files changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c71c6024320b..0917d65d6921 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -310,6 +310,8 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI select SND_SOC_FSL_SSI + select SND_SOC_WM8994 + select MFD_WM8994 help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888, diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index f62f81ceab0d..c62bfd1c3ac7 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -25,6 +25,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" +#include "../codecs/wm8994.h" #define CS427x_SYSCLK_MCLK 0 @@ -37,12 +38,14 @@ /** * struct codec_priv - CODEC private data * @mclk_freq: Clock rate of MCLK + * @free_freq: Clock rate of MCLK for hw_free() * @mclk_id: MCLK (or main clock) id for set_sysclk() * @fll_id: FLL (or secordary clock) id for set_sysclk() * @pll_id: PLL id for set_pll() */ struct codec_priv { unsigned long mclk_freq; + unsigned long free_freq; u32 mclk_id; u32 fll_id; u32 pll_id; @@ -235,10 +238,10 @@ 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) { - /* Force freq to be 0 to avoid error message in codec */ + /* 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, - 0, + codec_priv->free_freq, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "failed to switch away from FLL: %d\n", ret); @@ -665,6 +668,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->card.dapm_routes = audio_map_rx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) { + codec_dai_name = "wm8994-aif1"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1; + priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1; + priv->codec_priv.pll_id = WM8994_FLL1; + priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; + priv->card.dapm_routes = NULL; + priv->card.num_dapm_routes = 0; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -882,6 +894,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, { .compatible = "fsl,imx-audio-si476x", }, + { .compatible = "fsl,imx-audio-wm8958", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.27.0

2 2

[PATCH 1/3] ASoC:codec:max98373: Changed amp shutdown register as volatile
by Ryan Lee 25 Mar '21

25 Mar '21

0x20FF(amp global enable) register was defined as non-volatile, but it is not. Overheating, overcurrent can cause amp shutdown in hardware. 'regmap_write' compare register readback value before writing to avoid same value writing. 'regmap_read' just read cache not actual hardware value for the non-volatile register. When amp is internally shutdown by some reason, next 'AMP ON' command can be ignored because regmap think amp is already ON. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98373-i2c.c | 1 + sound/soc/codecs/max98373-sdw.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 85f6865019d4..ddb6436835d7 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -446,6 +446,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: return true; default: diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index d8c47667a9ea..f3a12205cd48 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -220,6 +220,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: /* SoundWire Control Port Registers */ case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: -- 2.17.1

2 4

[PATCH v2] ASoC: wm8960: Remove bitclk relax condition in wm8960_configure_sysclk
by Shengjiu Wang 25 Mar '21

25 Mar '21

The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. With this case: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav the required bitclk is 48000 * 24 * 2 = 2304000, bitclk got from wm8960_configure_sysclk is 3072000, but if go to wm8960_configure_pll. it can get correct bitclk 2304000. So bitclk relax condition should be removed in wm8960_configure_sysclk, then wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- changes in v2: - Find the real reason for the case, just patially remove bitclk relax sound/soc/codecs/wm8960.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..847ca16b9841 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; -- 2.27.0

3 2

[PATCH 0/3] ASoC: soc.h: small cleanups
by Kuninori Morimoto 25 Mar '21

25 Mar '21

Hi Mark These are small cleanups for soc.h Kuninori Morimoto (3): ASoC: soc.h: add asoc_link_to_cpu/codec/platform() macro ASoC: soc.h: fixup return timing for snd_soc_fixup_dai_links_platform_name() ASoC: soc.h: return error if multi platform at snd_soc_fixup_dai_links_platform_name() include/sound/soc.h | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

2 4

[PATCH v2 1/2] ALSA: usb-audio: Carve out connector value checking into a helper
by Kai-Heng Feng 25 Mar '21

25 Mar '21

This is preparation for next patch, no functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: - Only return early when ret < 0. sound/usb/mixer.c | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b004b2e63a5d..5a2d9a768f70 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1446,13 +1446,11 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } -/* get the connectors status and report it as boolean type */ -static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int get_connector_value(struct usb_mixer_elem_info *cval, + char *name, int *val) { - struct usb_mixer_elem_info *cval = kcontrol->private_data; struct snd_usb_audio *chip = cval->head.mixer->chip; - int idx = 0, validx, ret, val; + int idx = 0, validx, ret; validx = cval->control << 8 | 0; @@ -1467,21 +1465,24 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac2_conn, sizeof(uac2_conn)); - val = !!uac2_conn.bNrChannels; + if (val) + *val = !!uac2_conn.bNrChannels; } else { /* UAC_VERSION_3 */ struct uac3_insertion_ctl_blk uac3_conn; ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac3_conn, sizeof(uac3_conn)); - val = !!uac3_conn.bmConInserted; + if (val) + *val = !!uac3_conn.bmConInserted; } snd_usb_unlock_shutdown(chip); if (ret < 0) { - if (strstr(kcontrol->id.name, "Speaker")) { - ucontrol->value.integer.value[0] = 1; + if (name && strstr(name, "Speaker")) { + if (val) + *val = 1; return 0; } error: @@ -1491,6 +1492,21 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, return filter_error(cval, ret); } + return ret; +} + +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + int ret, val; + + ret = get_connector_value(cval, kcontrol->id.name, &val); + + if (ret < 0) + return ret; + ucontrol->value.integer.value[0] = val; return 0; } -- 2.30.2

2 5

Directions please...
by GitHub issues - opened 25 Mar '21

25 Mar '21

alsa-project/alsa-utils issue #84 was opened from giggywithit: Hello I design and build audio equipment and would like to know where or how I would go about getting access to the sound states to control a GPIO for a raspberry Pi. What I would like ideally is to control a GPIO state based upon actual non-zero data playback. My desire is to use the GPIO to control a mute function to silence the interactive periods in DAC activity. I know little to nothing about programming but imagine the function is intelligent enough to discern the difference between an open device and actual music playback. Other methods keep open the device to keep from switching off and on, however, I am looking for a more professional and detailed method to create. Any pointers helpful thank you S Issue URL : https://github.com/alsa-project/alsa-utils/issues/84 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH AUTOSEL 4.4 05/10] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index a3dd7030f629..321b1ac52bfd 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -78,7 +78,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

1 0

[PATCH AUTOSEL 4.4 04/10] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 1d4031818966..883b93f0bd38 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -286,9 +286,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 4.4 03/10] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b1c8bb39cdf1..db7734e45dd1 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -341,9 +341,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 4.9 12/13] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 635818fcda00..21a007c26407 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3389,12 +3389,17 @@ static int rt5659_set_dai_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

1 0

[PATCH AUTOSEL 4.9 05/13] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0c2a1413a8f5..14e564e38f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -75,7 +75,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

1 0

[PATCH AUTOSEL 4.9 04/13] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f0c9e2562474..acd5c30694e8 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -287,9 +287,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 4.9 03/13] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 3cc1135fc2cd..81fbbcaf8121 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -341,9 +341,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 4.14 15/16] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index fa66b11df8d4..ae626d57c1ad 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3391,12 +3391,17 @@ static int rt5659_set_dai_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

1 0

[PATCH AUTOSEL 4.14 08/16] ASoC: cs42l42: Always wait at least 3ms after reset
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 3 ++- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index ec322fda3c18..39adb2fdd003 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1809,7 +1809,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1936,6 +1936,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index d87a0a5322d5..3d5fa343db96 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -743,6 +743,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", -- 2.30.1

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[PATCH AUTOSEL 4.14 07/16] ASoC: cs42l42: Fix mixer volume control
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a2324a0e72ee..ec322fda3c18 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { }; static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -462,7 +462,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, -- 2.30.1

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[PATCH AUTOSEL 4.14 06/16] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ] Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 949dbdc0445e..0410f2e5183c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -56,13 +56,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

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[PATCH AUTOSEL 4.14 05/16] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d64cb28e8dc5..b7a0002d9872 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -75,7 +75,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 4.14 04/16] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 57c2add323c4..38510fd06458 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -287,9 +287,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 4.14 03/16] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 438fe52a12df..5af5dfc0fd46 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -341,9 +341,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH AUTOSEL 4.19 17/20] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 1c1a521c73cb..b331b3ba61a9 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3466,12 +3466,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

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[PATCH AUTOSEL 4.19 10/20] ASoC: cs42l42: Always wait at least 3ms after reset
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 3 ++- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a8ba518ba043..fddfd227a9c0 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1800,7 +1800,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1925,6 +1925,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 23b1a63315ca..bcaf4f22408d 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -744,6 +744,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", -- 2.30.1

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[PATCH AUTOSEL 4.19 09/20] ASoC: cs42l42: Fix mixer volume control
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a5bd9cff7085..a8ba518ba043 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { }; static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -462,7 +462,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, -- 2.30.1

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[PATCH AUTOSEL 4.19 08/20] ASoC: cs42l42: Fix channel width support
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ] Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index c7baa19bf317..a5bd9cff7085 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -695,24 +695,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_CLK_OASRC_SEL_MASK, CS42L42_CLK_OASRC_SEL_12 << CS42L42_CLK_OASRC_SEL_SHIFT); - /* channel 1 on low LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH1_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_LOW << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); - /* Channel 2 on high LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH2_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_HI << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -828,14 +810,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - int retval; + unsigned int width = (params_width(params) / 8) - 1; + unsigned int val = 0; cs42l42->srate = params_rate(params); - cs42l42->swidth = params_width(params); - retval = cs42l42_pll_config(component); + switch(substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + val |= width << CS42L42_ASP_RX_CH_RES_SHIFT; + /* channel 1 on low LRCLK */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + /* Channel 2 on high LRCLK */ + val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT; + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + break; + default: + break; + } - return retval; + return cs42l42_pll_config(component); } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, @@ -900,9 +897,9 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) return 0; } -#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE ) static const struct snd_soc_dai_ops cs42l42_ops = { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9d04ed75e5c8..23b1a63315ca 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -761,7 +761,6 @@ struct cs42l42_private { struct completion pdn_done; u32 sclk; u32 srate; - u32 swidth; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.30.1

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[PATCH AUTOSEL 4.19 07/20] ASoC: cs42l42: Fix Bitclock polarity inversion
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ] The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 20 ++++++++------------ sound/soc/codecs/cs42l42.h | 11 ++++++----- 2 files changed, 14 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 651329bf9743..c7baa19bf317 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -801,27 +801,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Bitclock/frame inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; break; case SND_SOC_DAIFMT_NB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; case SND_SOC_DAIFMT_IB_NF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; break; case SND_SOC_DAIFMT_IB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; } - snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, - CS42L42_ASP_MODE_MASK | - CS42L42_ASP_SCPOL_IN_DAC_MASK | - CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val); + snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK | + CS42L42_ASP_SCPOL_MASK | + CS42L42_ASP_LCPOL_MASK, + asp_cfg_val); return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 09b0a93203ef..9d04ed75e5c8 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -262,11 +262,12 @@ #define CS42L42_ASP_SLAVE_MODE 0x00 #define CS42L42_ASP_MODE_SHIFT 4 #define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT) -#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2 -#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT) -#define CS42L42_ASP_LCPOL_IN_SHIFT 0 -#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT) -#define CS42L42_ASP_POL_INV 1 +#define CS42L42_ASP_SCPOL_SHIFT 2 +#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT) +#define CS42L42_ASP_SCPOL_NOR 3 +#define CS42L42_ASP_LCPOL_SHIFT 0 +#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT) +#define CS42L42_ASP_LCPOL_INV 3 #define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08) #define CS42L42_ASP_STP_SHIFT 4 -- 2.30.1

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[PATCH AUTOSEL 4.19 06/20] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ] Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 9ebe77c3784a..57130edaf3ab 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -56,13 +56,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

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[PATCH AUTOSEL 4.19 05/20] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7c0a06b487f7..17255e9683f5 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 4.19 04/20] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852fd9723..318a4c9b380f 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -288,9 +288,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH AUTOSEL 4.19 03/20] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 974e1a449172..63e19a6a9790 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -342,9 +342,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH AUTOSEL 5.4 21/24] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index e66d08398f74..afd61599d94c 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3463,12 +3463,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

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[PATCH AUTOSEL 5.4 13/24] ASoC: cs42l42: Always wait at least 3ms after reset
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 3 ++- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 95930018a1a4..dcd2acb2c3ce 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1796,7 +1796,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1921,6 +1921,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9b017b76828a..866d7c873e3c 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -740,6 +740,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", -- 2.30.1

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[PATCH AUTOSEL 5.4 12/24] ASoC: cs42l42: Fix mixer volume control
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 26ca5acf0136..95930018a1a4 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = { }; static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, -- 2.30.1

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[PATCH AUTOSEL 5.4 11/24] ASoC: cs42l42: Fix channel width support
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ] Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index c78e2ce37930..26ca5acf0136 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -691,24 +691,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_CLK_OASRC_SEL_MASK, CS42L42_CLK_OASRC_SEL_12 << CS42L42_CLK_OASRC_SEL_SHIFT); - /* channel 1 on low LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH1_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_LOW << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); - /* Channel 2 on high LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH2_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_HI << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -824,14 +806,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - int retval; + unsigned int width = (params_width(params) / 8) - 1; + unsigned int val = 0; cs42l42->srate = params_rate(params); - cs42l42->swidth = params_width(params); - retval = cs42l42_pll_config(component); + switch(substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + val |= width << CS42L42_ASP_RX_CH_RES_SHIFT; + /* channel 1 on low LRCLK */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + /* Channel 2 on high LRCLK */ + val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT; + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + break; + default: + break; + } - return retval; + return cs42l42_pll_config(component); } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, @@ -896,9 +893,9 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) return 0; } -#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE ) static const struct snd_soc_dai_ops cs42l42_ops = { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 1f0d67c95a9a..9b017b76828a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -757,7 +757,6 @@ struct cs42l42_private { struct completion pdn_done; u32 sclk; u32 srate; - u32 swidth; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.30.1

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[PATCH AUTOSEL 5.4 10/24] ASoC: cs42l42: Fix Bitclock polarity inversion
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ] The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 20 ++++++++------------ sound/soc/codecs/cs42l42.h | 11 ++++++----- 2 files changed, 14 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5125bb9b37b5..c78e2ce37930 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -797,27 +797,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Bitclock/frame inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; break; case SND_SOC_DAIFMT_NB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; case SND_SOC_DAIFMT_IB_NF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; break; case SND_SOC_DAIFMT_IB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; } - snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, - CS42L42_ASP_MODE_MASK | - CS42L42_ASP_SCPOL_IN_DAC_MASK | - CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val); + snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK | + CS42L42_ASP_SCPOL_MASK | + CS42L42_ASP_LCPOL_MASK, + asp_cfg_val); return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9e3cc528dcff..1f0d67c95a9a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -258,11 +258,12 @@ #define CS42L42_ASP_SLAVE_MODE 0x00 #define CS42L42_ASP_MODE_SHIFT 4 #define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT) -#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2 -#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT) -#define CS42L42_ASP_LCPOL_IN_SHIFT 0 -#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT) -#define CS42L42_ASP_POL_INV 1 +#define CS42L42_ASP_SCPOL_SHIFT 2 +#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT) +#define CS42L42_ASP_SCPOL_NOR 3 +#define CS42L42_ASP_LCPOL_SHIFT 0 +#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT) +#define CS42L42_ASP_LCPOL_INV 3 #define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08) #define CS42L42_ASP_STP_SHIFT 4 -- 2.30.1

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[PATCH AUTOSEL 5.4 09/24] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ] Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 36eef1fb3d18..b781b28de012 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

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[PATCH AUTOSEL 5.4 08/24] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f5b59305c957..8a1e485982d8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 5.4 07/24] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index c506c9305043..829cf552fe3e 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH AUTOSEL 5.4 06/24] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 747ca248bf10..3bc63fbcb188 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH v2] ASoC: core: Don't set platform name when of_node is set
by Daniel Baluta 25 Mar '21

25 Mar '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> A DAI link has 3 components: * CPU * platform * codec(s) A component is specified via: * name * of_node * dai_name In order to avoid confusion when building a sound card we disallow matching by both name and of_node (1). soc_check_tplg_fes allows overriding certain BE links by overriding BE link name. This doesn't work well if BE link was specified via DT, because we end up with a link with both name and of_node specified which is conflicting with (1). In order to fix this we need to: * override of_node if component was specified via DT * override name, otherwise. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 522bae73640a..7a1c011a7fe0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1648,7 +1648,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platforms->name = component->name; + + if (component->dev->of_node) + dai_link->platforms->of_node = component->dev->of_node; + else + dai_link->platforms->name = component->name; /* convert non BE into BE */ if (!dai_link->no_pcm) { -- 2.27.0

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[PATCH AUTOSEL 5.10 29/39] ASoC: rt711: add snd_soc_component remove callback
by Sasha Levin 25 Mar '21

25 Mar '21

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 899b12542b0897f92de9ba30944937c39ebb246d ] We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index a9b1b4180c47..93d86f7558e0 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } +static void rt711_remove(struct snd_soc_component *component) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt711->regmap, true); +} + static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, + .remove = rt711_remove, }; static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, -- 2.30.1

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[PATCH AUTOSEL 5.10 28/39] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 41e5917b16a5..91a4ef7f620c 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

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[PATCH AUTOSEL 5.10 18/39] ASoC: cs42l42: Always wait at least 3ms after reset
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 3 ++- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5078ce79fad..4d82d24c7828 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1794,7 +1794,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1919,6 +1919,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9b017b76828a..866d7c873e3c 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -740,6 +740,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", -- 2.30.1

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[PATCH AUTOSEL 5.10 17/39] ASoC: cs42l42: Fix mixer volume control
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 4f9ad9547929..d5078ce79fad 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = { }; static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, -- 2.30.1

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[PATCH AUTOSEL 5.10 16/39] ASoC: cs42l42: Fix channel width support
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ] Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index df0d5fec0287..4f9ad9547929 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -691,24 +691,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_CLK_OASRC_SEL_MASK, CS42L42_CLK_OASRC_SEL_12 << CS42L42_CLK_OASRC_SEL_SHIFT); - /* channel 1 on low LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH1_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_LOW << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); - /* Channel 2 on high LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH2_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_HI << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -824,14 +806,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - int retval; + unsigned int width = (params_width(params) / 8) - 1; + unsigned int val = 0; cs42l42->srate = params_rate(params); - cs42l42->swidth = params_width(params); - retval = cs42l42_pll_config(component); + switch(substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + val |= width << CS42L42_ASP_RX_CH_RES_SHIFT; + /* channel 1 on low LRCLK */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + /* Channel 2 on high LRCLK */ + val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT; + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + break; + default: + break; + } - return retval; + return cs42l42_pll_config(component); } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, @@ -896,9 +893,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction) return 0; } -#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE ) static const struct snd_soc_dai_ops cs42l42_ops = { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 1f0d67c95a9a..9b017b76828a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -757,7 +757,6 @@ struct cs42l42_private { struct completion pdn_done; u32 sclk; u32 srate; - u32 swidth; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.30.1

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[PATCH AUTOSEL 5.10 15/39] ASoC: cs42l42: Fix Bitclock polarity inversion
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ] The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 20 ++++++++------------ sound/soc/codecs/cs42l42.h | 11 ++++++----- 2 files changed, 14 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 210fcbedf241..df0d5fec0287 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -797,27 +797,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Bitclock/frame inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; break; case SND_SOC_DAIFMT_NB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; case SND_SOC_DAIFMT_IB_NF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; break; case SND_SOC_DAIFMT_IB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; } - snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, - CS42L42_ASP_MODE_MASK | - CS42L42_ASP_SCPOL_IN_DAC_MASK | - CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val); + snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK | + CS42L42_ASP_SCPOL_MASK | + CS42L42_ASP_LCPOL_MASK, + asp_cfg_val); return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9e3cc528dcff..1f0d67c95a9a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -258,11 +258,12 @@ #define CS42L42_ASP_SLAVE_MODE 0x00 #define CS42L42_ASP_MODE_SHIFT 4 #define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT) -#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2 -#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT) -#define CS42L42_ASP_LCPOL_IN_SHIFT 0 -#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT) -#define CS42L42_ASP_POL_INV 1 +#define CS42L42_ASP_SCPOL_SHIFT 2 +#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT) +#define CS42L42_ASP_SCPOL_NOR 3 +#define CS42L42_ASP_LCPOL_SHIFT 0 +#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT) +#define CS42L42_ASP_LCPOL_INV 3 #define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08) #define CS42L42_ASP_STP_SHIFT 4 -- 2.30.1

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[PATCH AUTOSEL 5.10 14/39] ASoC: soc-core: Prevent warning if no DMI table is present
by Sasha Levin 25 Mar '21

25 Mar '21

From: Jon Hunter <jonathanh(a)nvidia.com> [ Upstream commit 7de14d581dbed57c2b3c6afffa2c3fdc6955a3cd ] Many systems do not use ACPI and hence do not provide a DMI table. On non-ACPI systems a warning, such as the following, is printed on boot. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The variable 'dmi_available' is not exported and so currently cannot be used by kernel modules without adding an accessor. However, it is possible to use the function is_acpi_device_node() to determine if the sound card is an ACPI device and hence indicate if we expect a DMI table to be present. Therefore, call is_acpi_device_node() to see if we are using ACPI and only parse the DMI table if we are booting with ACPI. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 05a085f6dc7c..bf65cba232e6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -31,6 +31,7 @@ #include <linux/of.h> #include <linux/of_graph.h> #include <linux/dmi.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -1581,6 +1582,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (card->long_name) return 0; /* long name already set by driver or from DMI */ + if (!is_acpi_device_node(card->dev->fwnode)) + return 0; + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { -- 2.30.1

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[PATCH AUTOSEL 5.10 13/39] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ] Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index bd5d230c5df2..609459077f9d 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

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[PATCH AUTOSEL 5.10 12/39] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 4d6ff8114622..4c0e87e22b97 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 5.10 11/39] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index d198e191fb0c..e59fdc81dbd4 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH AUTOSEL 5.10 10/39] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1414ad15d01c..a5674c227b3a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

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[PATCH AUTOSEL 5.10 09/39] ASoC: rt1015: fix i2c communication error
by Sasha Levin 25 Mar '21

25 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 9e0bdaa9fcb8c64efc1487a7fba07722e7bc515e ] Remove 0x100 cache re-sync to solve i2c communication error. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 3db07293c70b..2627910060dc 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_VENDOR_ID: case RT1015_DEVICE_ID: case RT1015_PRO_ALT: + case RT1015_MAN_I2C: case RT1015_DAC3: case RT1015_VBAT_TEST_OUT1: case RT1015_VBAT_TEST_OUT2: -- 2.30.1

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[PATCH AUTOSEL 5.11 32/44] ASoC: rt711: add snd_soc_component remove callback
by Sasha Levin 25 Mar '21

25 Mar '21

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 899b12542b0897f92de9ba30944937c39ebb246d ] We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 85f744184a60..047f4e677d78 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } +static void rt711_remove(struct snd_soc_component *component) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt711->regmap, true); +} + static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, + .remove = rt711_remove, }; static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, -- 2.30.1

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[PATCH AUTOSEL 5.11 31/44] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 41e5917b16a5..91a4ef7f620c 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

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[PATCH AUTOSEL 5.11 20/44] ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge
by Sasha Levin 25 Mar '21

25 Mar '21

From: Jiaxin Yu <jiaxin.yu(a)mediatek.com> [ Upstream commit 8d06b9633a66f41fed520f6eebd163189518ba79 ] This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3]. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> Link: https://lore.kernel.org/r/1615516005-781-1-git-send-email-jiaxin.yu@mediate… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 +++- sound/soc/mediatek/mt8192/mt8192-reg.h | 8 +++++--- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c index 8383536b7ae0..504293de2c0d 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c @@ -555,7 +555,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, /* set tdm */ if (tdm_priv->bck_invert) - tdm_con |= 1 << BCK_INVERSE_SFT; + regmap_update_bits(afe->regmap, AUDIO_TOP_CON3, + BCK_INVERSE_MASK_SFT, + 0x1 << BCK_INVERSE_SFT); if (tdm_priv->lck_invert) tdm_con |= 1 << LRCK_INVERSE_SFT; diff --git a/sound/soc/mediatek/mt8192/mt8192-reg.h b/sound/soc/mediatek/mt8192/mt8192-reg.h index 562f25c79c34..b9fb80d4afec 100644 --- a/sound/soc/mediatek/mt8192/mt8192-reg.h +++ b/sound/soc/mediatek/mt8192/mt8192-reg.h @@ -21,6 +21,11 @@ enum { /***************************************************************************** * R E G I S T E R D E F I N I T I O N *****************************************************************************/ +/* AUDIO_TOP_CON3 */ +#define BCK_INVERSE_SFT 3 +#define BCK_INVERSE_MASK 0x1 +#define BCK_INVERSE_MASK_SFT (0x1 << 3) + /* AFE_DAC_CON0 */ #define VUL12_ON_SFT 31 #define VUL12_ON_MASK 0x1 @@ -2079,9 +2084,6 @@ enum { #define TDM_EN_SFT 0 #define TDM_EN_MASK 0x1 #define TDM_EN_MASK_SFT (0x1 << 0) -#define BCK_INVERSE_SFT 1 -#define BCK_INVERSE_MASK 0x1 -#define BCK_INVERSE_MASK_SFT (0x1 << 1) #define LRCK_INVERSE_SFT 2 #define LRCK_INVERSE_MASK 0x1 #define LRCK_INVERSE_MASK_SFT (0x1 << 2) -- 2.30.1

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[PATCH AUTOSEL 5.11 18/44] ASoC: cs42l42: Always wait at least 3ms after reset
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 3 ++- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5078ce79fad..4d82d24c7828 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1794,7 +1794,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1919,6 +1919,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9b017b76828a..866d7c873e3c 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -740,6 +740,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", -- 2.30.1

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[PATCH AUTOSEL 5.11 17/44] ASoC: cs42l42: Fix mixer volume control
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 4f9ad9547929..d5078ce79fad 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = { }; static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, -- 2.30.1

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[PATCH AUTOSEL 5.11 16/44] ASoC: cs42l42: Fix channel width support
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ] Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index df0d5fec0287..4f9ad9547929 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -691,24 +691,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_CLK_OASRC_SEL_MASK, CS42L42_CLK_OASRC_SEL_12 << CS42L42_CLK_OASRC_SEL_SHIFT); - /* channel 1 on low LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH1_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_LOW << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); - /* Channel 2 on high LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH2_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_HI << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -824,14 +806,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - int retval; + unsigned int width = (params_width(params) / 8) - 1; + unsigned int val = 0; cs42l42->srate = params_rate(params); - cs42l42->swidth = params_width(params); - retval = cs42l42_pll_config(component); + switch(substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + val |= width << CS42L42_ASP_RX_CH_RES_SHIFT; + /* channel 1 on low LRCLK */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + /* Channel 2 on high LRCLK */ + val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT; + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + break; + default: + break; + } - return retval; + return cs42l42_pll_config(component); } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, @@ -896,9 +893,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction) return 0; } -#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE ) static const struct snd_soc_dai_ops cs42l42_ops = { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 1f0d67c95a9a..9b017b76828a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -757,7 +757,6 @@ struct cs42l42_private { struct completion pdn_done; u32 sclk; u32 srate; - u32 swidth; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.30.1

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[PATCH AUTOSEL 5.11 15/44] ASoC: cs42l42: Fix Bitclock polarity inversion
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ] The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 20 ++++++++------------ sound/soc/codecs/cs42l42.h | 11 ++++++----- 2 files changed, 14 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 210fcbedf241..df0d5fec0287 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -797,27 +797,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Bitclock/frame inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; break; case SND_SOC_DAIFMT_NB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; case SND_SOC_DAIFMT_IB_NF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; break; case SND_SOC_DAIFMT_IB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; } - snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, - CS42L42_ASP_MODE_MASK | - CS42L42_ASP_SCPOL_IN_DAC_MASK | - CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val); + snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK | + CS42L42_ASP_SCPOL_MASK | + CS42L42_ASP_LCPOL_MASK, + asp_cfg_val); return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9e3cc528dcff..1f0d67c95a9a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -258,11 +258,12 @@ #define CS42L42_ASP_SLAVE_MODE 0x00 #define CS42L42_ASP_MODE_SHIFT 4 #define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT) -#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2 -#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT) -#define CS42L42_ASP_LCPOL_IN_SHIFT 0 -#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT) -#define CS42L42_ASP_POL_INV 1 +#define CS42L42_ASP_SCPOL_SHIFT 2 +#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT) +#define CS42L42_ASP_SCPOL_NOR 3 +#define CS42L42_ASP_LCPOL_SHIFT 0 +#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT) +#define CS42L42_ASP_LCPOL_INV 3 #define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08) #define CS42L42_ASP_STP_SHIFT 4 -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 14/44] ASoC: soc-core: Prevent warning if no DMI table is present
by Sasha Levin 25 Mar '21

25 Mar '21

From: Jon Hunter <jonathanh(a)nvidia.com> [ Upstream commit 7de14d581dbed57c2b3c6afffa2c3fdc6955a3cd ] Many systems do not use ACPI and hence do not provide a DMI table. On non-ACPI systems a warning, such as the following, is printed on boot. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The variable 'dmi_available' is not exported and so currently cannot be used by kernel modules without adding an accessor. However, it is possible to use the function is_acpi_device_node() to determine if the sound card is an ACPI device and hence indicate if we expect a DMI table to be present. Therefore, call is_acpi_device_node() to see if we are using ACPI and only parse the DMI table if we are booting with ACPI. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..0cffc9527e28 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -31,6 +31,7 @@ #include <linux/of.h> #include <linux/of_graph.h> #include <linux/dmi.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1573,6 +1574,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (card->long_name) return 0; /* long name already set by driver or from DMI */ + if (!is_acpi_device_node(card->dev->fwnode)) + return 0; + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 13/44] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ] Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index f9ec5cf82599..ec2f11ff8a84 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 12/44] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 4d6ff8114622..4c0e87e22b97 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 11/44] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index d198e191fb0c..e59fdc81dbd4 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 10/44] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1414ad15d01c..a5674c227b3a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } 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 */ -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 09/44] ASoC: rt1015: fix i2c communication error
by Sasha Levin 25 Mar '21

25 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 9e0bdaa9fcb8c64efc1487a7fba07722e7bc515e ] Remove 0x100 cache re-sync to solve i2c communication error. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 32e6bcf763d1..4607039a16e7 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_VENDOR_ID: case RT1015_DEVICE_ID: case RT1015_PRO_ALT: + case RT1015_MAN_I2C: case RT1015_DAC3: case RT1015_VBAT_TEST_OUT1: case RT1015_VBAT_TEST_OUT2: -- 2.30.1

1 0

[PATCH V2] ALSA: pcm: Fix couple of typos
by Bhaskar Chowdhury 25 Mar '21

25 Mar '21

s/unconditonally/unconditionally/ s/succesful/successful/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Changes from V1: Randy's finding incorporated ,plus the subject line adjusted. sound/core/pcm_native.c | 82 ++++++++++++++++++++--------------------- 1 file changed, 41 insertions(+), 41 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..afb670d7fd53 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1425,7 +1425,7 @@ static int snd_pcm_do_stop(struct snd_pcm_substream *substream, substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP); substream->runtime->stop_operating = true; } - return 0; /* unconditonally stop all substreams */ + return 0; /* unconditionally stop all substreams */ } static void snd_pcm_post_stop(struct snd_pcm_substream *substream, @@ -1469,7 +1469,7 @@ EXPORT_SYMBOL(snd_pcm_stop); * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. * - * Return: Zero if succesful, or a negative error code. + * Return: Zero if successful, or a negative error code. -- 2.30.1

2 2

[PATCH 1/2] ALSA: usb-audio: Carve out connector value checking into a helper
by Kai-Heng Feng 25 Mar '21

25 Mar '21

This is preparation for next patch, no functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/mixer.c | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b004b2e63a5d..98f5417a70e4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1446,13 +1446,11 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } -/* get the connectors status and report it as boolean type */ -static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int get_connector_value(struct usb_mixer_elem_info *cval, + char *name, int *val) { - struct usb_mixer_elem_info *cval = kcontrol->private_data; struct snd_usb_audio *chip = cval->head.mixer->chip; - int idx = 0, validx, ret, val; + int idx = 0, validx, ret; validx = cval->control << 8 | 0; @@ -1467,21 +1465,24 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac2_conn, sizeof(uac2_conn)); - val = !!uac2_conn.bNrChannels; + if (val) + *val = !!uac2_conn.bNrChannels; } else { /* UAC_VERSION_3 */ struct uac3_insertion_ctl_blk uac3_conn; ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac3_conn, sizeof(uac3_conn)); - val = !!uac3_conn.bmConInserted; + if (val) + *val = !!uac3_conn.bmConInserted; } snd_usb_unlock_shutdown(chip); if (ret < 0) { - if (strstr(kcontrol->id.name, "Speaker")) { - ucontrol->value.integer.value[0] = 1; + if (name && strstr(name, "Speaker")) { + if (val) + *val = 1; return 0; } error: @@ -1491,6 +1492,21 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, return filter_error(cval, ret); } + return ret; +} + +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + int ret, val; + + ret = get_connector_value(cval, kcontrol->id.name, &val); + + if (ret) + return ret; + ucontrol->value.integer.value[0] = val; return 0; } -- 2.30.2

2 3

[PATCH] ALSA: hda/hdmi: fix max DP-MST dev_num for Intel TGL+ platforms
by Kai Vehmanen 25 Mar '21

25 Mar '21

Increase the device select range to 4 on platforms supporting 4 concurrent displays. This fixes a problem in scenario where total of 4 displays are active, and 3 of these are audio capable DP receivers and connected to a DP-MST hub. Due to incorrect range for device select, audio could not be played to the 3rd monitor in DP-MST hub. BugLink: https://github.com/thesofproject/linux/issues/2798 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 45ae845e82df..5de3666a7101 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1848,16 +1848,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) */ if (spec->intel_hsw_fixup) { /* - * On Intel platforms, device entries number is - * changed dynamically. If there is a DP MST - * hub connected, the device entries number is 3. - * Otherwise, it is 1. - * Here we manually set dev_num to 3, so that - * we can initialize all the device entries when - * bootup statically. + * On Intel platforms, device entries count returned + * by AC_PAR_DEVLIST_LEN is dynamic, and depends on + * the type of receiver that is connected. Allocate pin + * structures based on worst case. */ - dev_num = 3; - spec->dev_num = 3; + dev_num = spec->dev_num; } else if (spec->dyn_pcm_assign && codec->dp_mst) { dev_num = snd_hda_get_num_devices(codec, pin_nid) + 1; /* @@ -2942,7 +2938,7 @@ static int parse_intel_hdmi(struct hda_codec *codec) /* Intel Haswell and onwards; audio component with eld notifier */ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, - const int *port_map, int port_num) + const int *port_map, int port_num, int dev_num) { struct hdmi_spec *spec; int err; @@ -2957,6 +2953,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, spec->port_map = port_map; spec->port_num = port_num; spec->intel_hsw_fixup = true; + spec->dev_num = dev_num; intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); @@ -2982,12 +2979,12 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x08, NULL, 0); + return intel_hsw_common_init(codec, 0x08, NULL, 0, 3); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x0b, NULL, 0); + return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3); } static int patch_i915_icl_hdmi(struct hda_codec *codec) @@ -2998,7 +2995,7 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) */ static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3); } static int patch_i915_tgl_hdmi(struct hda_codec *codec) @@ -3010,7 +3007,7 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; int ret; - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4); if (!ret) { struct hdmi_spec *spec = codec->spec; base-commit: 950dc7fe37b99e77160154f6ce767b987816deaf -- 2.29.2

2 1

Re: [PATCH] ALSA: usb-audio: Fix missing return assignment
by Dan Carpenter 25 Mar '21

25 Mar '21

On Thu, Mar 25, 2021 at 01:24:23AM +0500, Muhammad Usama Anjum wrote: > On Wed, 2021-03-24 at 21:50 +0300, Dan Carpenter wrote: > > > - usb_driver_claim_interface(driver, iface, (void *)-1L); > > > + err = usb_driver_claim_interface(driver, iface, (void *)-1L); > > > > This is in a loop so only the last return value is used. Which seems > > sort of weird and pointless that the last value would matter more than > > the others. > > > Correct. Lets not store the return value. To stop the static analyzers > to report the missing return assignment, can we add (void) in start of > this function call? I've not seen use of (void) this way in the > kernel. Is there any other way used in the kernel? Don't add (void). Don't add any code just to help static checkers, only do it if it helps humans. The (void) stuff is ugly. We have a __must_check annotation for functions where it's a bug not to check the return and the usb_driver_claim_interface() is not a __must_check function. Just ignore the static checker when it's wrong. When I'm reviewing static checker warnings, I only look at the new ones. Then after I've looked at them, I mark them as old. I currently have 65k old ignored warnings. regards, dan carpenter

1 0

[PATCH] ALSA: pcm: A typo fix
by Bhaskar Chowdhury 25 Mar '21

25 Mar '21

s/unconditonally/unconditionally/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/core/pcm_native.c | 80 ++++++++++++++++++++--------------------- 1 file changed, 40 insertions(+), 40 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..9884961bca7d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c - return 0; /* unconditonally stop all substreams */ + return 0; /* unconditionally stop all substreams */ } -- 2.30.1

2 2

[PATCH v4 resend 00/13] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 25 Mar '21

25 Mar '21

Hi All, Here is v4 of my series to rework the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. As discussed before here is a resend rebased on 5.12-rc2, making sure that all patches this depends on are in place. Lee, can you pick-up patches 1-6 through the MFD tree and then send a pull-req to Mark so that Mark can merge the Asoc parts throught the ASoC tree ? Patches 2-6 touch drivers/extcon, these all have an Ack from Chanwoo Choi for merging these through the MFD tree. Here is some more generic info on this series from the previous cover-letter: This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. This was also tested by Charles Keepax, one of the Cirrus Codec folks. Regards, Hans Hans de Goede (13): mfd: arizona: Drop arizona-extcon cells extcon: arizona: Fix some issues when HPDET IRQ fires after the jack has been unplugged extcon: arizona: Fix various races on driver unbind extcon: arizona: Fix flags parameter to the gpiod_get("wlf,micd-pol") call extcon: arizona: Always use pm_runtime_get_sync() when we need the device to be awake ASoC/extcon: arizona: Move arizona jack code to sound/soc/codecs/arizona-jack.c ASoC: arizona-jack: Move jack-detect variables to struct arizona_priv ASoC: arizona-jack: Use arizona->dev for runtime-pm ASoC: arizona-jack: convert into a helper library for codec drivers ASoC: arizona-jack: Use snd_soc_jack to report jack events ASoC: arizona-jack: Cleanup logging ASoC: arizona: Make the wm5102, wm5110, wm8997 and wm8998 drivers use the new jack library ASoC: Intel: bytcr_wm5102: Add jack detect support MAINTAINERS | 3 +- drivers/extcon/Kconfig | 8 - drivers/extcon/Makefile | 1 - drivers/mfd/arizona-core.c | 20 - sound/soc/codecs/Makefile | 2 +- .../soc/codecs/arizona-jack.c | 577 +++++++----------- sound/soc/codecs/arizona.h | 44 ++ sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8997.c | 14 +- sound/soc/codecs/wm8998.c | 9 + sound/soc/intel/boards/bytcr_wm5102.c | 28 +- 12 files changed, 325 insertions(+), 405 deletions(-) rename drivers/extcon/extcon-arizona.c => sound/soc/codecs/arizona-jack.c (76%) -- 2.30.1

3 18

[PATCH v2 0/2] ASoC: intel: atom: 2 bug-fixes for the atom SST driver
by Hans de Goede 25 Mar '21

25 Mar '21

Hi All, Here is v2 of my bugfix series for the ASoC Intel Atom SST driver. The patches are unchanged except for adding Pierre-Louis' Acked-By and adding a fixed (subject fixed) version of the Fixes: tag which Pierre-Louis suggested. The first patch fixes a bug which is causing audio to now work with pipewire and both Fedora and Arch Linux are moving towards using pipewire as the default sound-server. As such it would be good if we can get these 2 bugfixes added to a 5.12-rc# release. Regards, Hans Hans de Goede (2): ASoC: intel: atom: Stop advertising non working S24LE support ASoC: intel: atom: Remove 44100 sample-rate from the media and deep-buffer DAI descriptions sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) -- 2.30.2

2 3

[PATCH 1/2] ASoC: intel: atom: Stop advertising non working S24LE support
by Hans de Goede 25 Mar '21

25 Mar '21

The SST firmware's media and deep-buffer inputs are hardcoded to S16LE, the corresponding DAIs don't have a hw_params callback and their prepare callback also does not take the format into account. So far the advertising of non working S24LE support has not caused issues because pulseaudio defaults to S16LE, but changing pulse-audio's config to use S24LE will result in broken sound. Pipewire is replacing pulse now and pipewire prefers S24LE over S16LE when available, causing the problem of the broken S24LE support to come to the surface now. Cc: stable(a)vger.kernel.org BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/866 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 9e9b05883557..aa5dd590ddd5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -488,14 +488,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -506,7 +506,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { -- 2.30.1

3 7

[PATCH 0/6] ASoC: fsl: Don't use devm_regmap_init_mmio_clk
by Shengjiu Wang 25 Mar '21

25 Mar '21

When there is power domain bind with ipg clock, The call flow: devm_regmap_init_mmio_clk - clk_prepare() - clk_pm_runtime_get() cause the power domain of clock always be enabled after regmap_init(). which impact the power consumption. So use devm_regmap_init_mmio instead of devm_regmap_init_mmio_clk. Shengjiu Wang (6): ASoC: fsl_esai: Don't use devm_regmap_init_mmio_clk ASoC: fsl_spdif: Don't use devm_regmap_init_mmio_clk ASoC: fsl_asrc: Don't use devm_regmap_init_mmio_clk ASoC: fsl_easrc: Don't use devm_regmap_init_mmio_clk ASoC: fsl_audmix: Don't use devm_regmap_init_mmio_clk ASoC: fsl_micfil: Don't use devm_regmap_init_mmio_clk sound/soc/fsl/fsl_asrc.c | 57 +++++++++++++++++++++++++++++--------- sound/soc/fsl/fsl_audmix.c | 3 +- sound/soc/fsl/fsl_easrc.c | 3 +- sound/soc/fsl/fsl_esai.c | 48 ++++++++++++++++++++++++-------- sound/soc/fsl/fsl_micfil.c | 25 +++++++++++++---- sound/soc/fsl/fsl_spdif.c | 3 +- 6 files changed, 103 insertions(+), 36 deletions(-) -- 2.27.0

2 7

[RESEND PATCH v2] ASoC: wm8960: Remove bitclk relax condition in wm8960_configure_sysclk
by Shengjiu Wang 24 Mar '21

24 Mar '21

The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. With this case: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav the required bitclk is 48000 * 24 * 2 = 2304000, bitclk got from wm8960_configure_sysclk is 3072000, but if go to wm8960_configure_pll. it can get correct bitclk 2304000. So bitclk relax condition should be removed in wm8960_configure_sysclk, then wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- changes in resend v2 - Add acked-by Charles changes in v2: - Find the real reason for the case, just patially remove bitclk relax sound/soc/codecs/wm8960.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..847ca16b9841 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; -- 2.27.0

3 3

[PATCH v5 1/2] ASoC: amd: Add support for RT5682 codec in machine driver
by Vijendar Mukunda 24 Mar '21

24 Mar '21

Add support for RT5682 codec in machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v4->v5:add checks for clk_set_rate() API return value fixed multiple definition error for acp_soc_is_rltk_max() API v3->v4: fixed clk_prepare_enable() for wclk instead of bclk v2->v3: updated SPDX license v1->v2: updated kconfig, spdx license, removed unnecessary initialization and updated comment sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-da7219-max98357a.c | 357 +++++++++++++++++++++++++++++++++-- 2 files changed, 345 insertions(+), 17 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index a6ce000..43f5d29 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -5,14 +5,15 @@ config SND_SOC_AMD_ACP This option enables ACP DMA support on AMD platform. config SND_SOC_AMD_CZ_DA7219MX98357_MACH - tristate "AMD CZ support for DA7219 and MAX9835" + tristate "AMD CZ support for DA7219, RT5682 and MAX9835" select SND_SOC_DA7219 + select SND_SOC_RT5682_I2C select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help - This option enables machine driver for DA7219 and MAX9835. + This option enables machine driver for DA7219, RT5682 and MAX9835. config SND_SOC_AMD_CZ_RT5645_MACH tristate "AMD CZ support for RT5645" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 849288d..e65e007 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -41,14 +41,19 @@ #include "acp.h" #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" +#include "../codecs/rt5682.h" #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +#define RT5682_PLL_FREQ (48000 * 512) static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; extern bool bt_uart_enable; +void *acp_soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -128,6 +133,96 @@ static void da7219_clk_disable(void) clk_disable_unprepare(da7219_dai_bclk); } +static int cz_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + dev_info(codec_dai->dev, "codec dai name = %s\n", codec_dai->name); + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CZ_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(rt5682_dai_wclk)) + return PTR_ERR(rt5682_dai_wclk); + + rt5682_dai_bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(rt5682_dai_bclk)) + return PTR_ERR(rt5682_dai_bclk); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &cz_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &cz_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + return 0; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." RT5682 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + ret = clk_set_rate(rt5682_dai_wclk, 48000); + if (ret) { + dev_err(rtd->dev, "Error setting wclk rate: %d\n", ret); + return ret; + } + ret = clk_set_rate(rt5682_dai_bclk, 48000 * 64); + if (ret) { + dev_err(rtd->dev, "Error setting bclk rate: %d\n", ret); + return ret; + } + ret = clk_prepare_enable(rt5682_dai_wclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable wclk %d\n", ret); + return ret; + } + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_wclk); +} + static const unsigned int channels[] = { DUAL_CHANNEL, }; @@ -260,6 +355,118 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_rt5682_play_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return rt5682_clk_enable(substream); +} + +static void cz_rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -285,6 +492,31 @@ static const struct snd_soc_ops cz_dmic1_cap_ops = { .shutdown = cz_da7219_shutdown, }; +static const struct snd_soc_ops cz_rt5682_play_ops = { + .startup = cz_rt5682_play_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_cap_ops = { + .startup = cz_rt5682_cap_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_max_play_ops = { + .startup = cz_rt5682_max_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic0_cap_ops = { + .startup = cz_rt5682_dmic0_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { + .startup = cz_rt5682_dmic1_startup, + .shutdown = cz_rt5682_shutdown, +}; + SND_SOC_DAILINK_DEF(designware1, DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); SND_SOC_DAILINK_DEF(designware2, @@ -294,6 +526,8 @@ SND_SOC_DAILINK_DEF(designware3, SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(mx, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); SND_SOC_DAILINK_DEF(adau, @@ -353,6 +587,57 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { }, }; +static struct snd_soc_dai_link cz_dai_5682_98357[] = { + { + .name = "amd-rt5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = cz_rt5682_init, + .dpcm_playback = 1, + .ops = &cz_rt5682_play_ops, + SND_SOC_DAILINK_REG(designware1, rt5682, platform), + }, + { + .name = "amd-rt5682-cap", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_cap_ops, + SND_SOC_DAILINK_REG(designware2, rt5682, platform), + }, + { + .name = "amd-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &cz_rt5682_max_play_ops, + SND_SOC_DAILINK_REG(designware3, mx, platform), + }, + { + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic0_cap_ops, + SND_SOC_DAILINK_REG(designware3, adau, platform), + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic1_cap_ops, + SND_SOC_DAILINK_REG(designware2, adau, platform), + }, +}; + static const struct snd_soc_dapm_widget cz_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -368,6 +653,14 @@ static const struct snd_soc_dapm_route cz_audio_route[] = { {"PDM_DAT", NULL, "Int Mic"}, }; +static const struct snd_soc_dapm_route cz_rt5682_audio_route[] = { + {"Headphones", NULL, "HPOL"}, + {"Headphones", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Speakers", NULL, "Speaker"}, + {"PDM_DAT", NULL, "Int Mic"}, +}; + static const struct snd_kcontrol_new cz_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), @@ -388,6 +681,28 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct snd_soc_card cz_rt5682_card = { + .name = "acpr5682m98357", + .owner = THIS_MODULE, + .dai_link = cz_dai_5682_98357, + .num_links = ARRAY_SIZE(cz_dai_5682_98357), + .dapm_widgets = cz_widgets, + .num_dapm_widgets = ARRAY_SIZE(cz_widgets), + .dapm_routes = cz_rt5682_audio_route, + .controls = cz_mc_controls, + .num_controls = ARRAY_SIZE(cz_mc_controls), +}; + +void *acp_soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static struct regulator_consumer_supply acp_da7219_supplies[] = { REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), @@ -425,29 +740,39 @@ static int cz_probe(struct platform_device *pdev) struct snd_soc_card *card; struct acp_platform_info *machine; struct regulator_dev *rdev; - - acp_da7219_cfg.dev = &pdev->dev; - rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, - &acp_da7219_cfg); - if (IS_ERR(rdev)) { - dev_err(&pdev->dev, "Failed to register regulator: %d\n", - (int)PTR_ERR(rdev)); - return -EINVAL; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)acp_soc_is_rltk_max(dev); + if (!card) + return -ENODEV; + if (!strcmp(card->name, "acpd7219m98357")) { + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &cz_card; - cz_card.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - cz_card.name, ret); + card->name, ret); + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); return ret; } bt_uart_enable = !device_property_read_bool(&pdev->dev, @@ -457,7 +782,8 @@ static int cz_probe(struct platform_device *pdev) #ifdef CONFIG_ACPI static const struct acpi_device_id cz_audio_acpi_match[] = { - { "AMD7219", 0 }, + { "AMD7219", (unsigned long)&cz_card }, + { "AMDI5682", (unsigned long)&cz_rt5682_card}, {}, }; MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match); @@ -475,5 +801,6 @@ static struct platform_driver cz_pcm_driver = { module_platform_driver(cz_pcm_driver); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("DA7219 & MAX98357A audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("DA7219, RT5682 & MAX98357A audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

3 4

Re: [PATCH] ALSA: usb-audio: Fix missing return assignment
by Dan Carpenter 24 Mar '21

24 Mar '21

On Wed, Mar 24, 2021 at 10:26:04PM +0500, Muhammad Usama Anjum wrote: > Return value of usb_driver_claim_interface should not be ignored. > Instead it should be stored in err variable and returned from > this function. > > Signed-off-by: Muhammad Usama Anjum <musamaanjum(a)gmail.com> > --- > sound/usb/quirks.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 9e5e37eff10e..dd32ceaef18a 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -427,10 +427,10 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, > > err = create_autodetect_quirk(chip, iface, driver); > if (err >= 0) create_autodetect_quirk() never returns positive values. Flip this condition. (Always do error handling, don't do success handling). if (err) continue; > - usb_driver_claim_interface(driver, iface, (void *)-1L); > + err = usb_driver_claim_interface(driver, iface, (void *)-1L); This is in a loop so only the last return value is used. Which seems sort of weird and pointless that the last value would matter more than the others. > } > > - return 0; > + return err; regards, dan carpenter

1 0

man: alsaloop man page error
by GitHub issues - edited 24 Mar '21

24 Mar '21

alsa-project/alsa-utils issue #82 was edited from nootc: There is a small typo in the alsaloop man page: https://github.com/alsa-project/alsa-utils/blob/c9e9a79c6cfef3212bdb5f9be4f… "-c" is used for both "channels" and "rate". Thankfully, the source thinks otherwise and "-r" is used for "rate". Issue URL : https://github.com/alsa-project/alsa-utils/issues/82 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

man: Correct "rate" option
by GitHub pull_request - edited 24 Mar '21

24 Mar '21

alsa-project/alsa-utils pull request #83 was edited from nootc: Resolves https://github.com/alsa-project/alsa-utils/issues/82 Change "rate" option from "-c" to "-r". Request URL : https://github.com/alsa-project/alsa-utils/pull/83 Patch URL : https://github.com/alsa-project/alsa-utils/pull/83.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

man: Correct "rate" option
by GitHub pull_request - edited 24 Mar '21

24 Mar '21

alsa-project/alsa-utils pull request #83 was edited from nootc: https://github.com/alsa-project/alsa-utils/issues/82 Change "rate" option from "-c" to "-r". Request URL : https://github.com/alsa-project/alsa-utils/pull/83 Patch URL : https://github.com/alsa-project/alsa-utils/pull/83.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: dt-bindings: nvidia, tegra210-ahub: Add missing child nodes
by Rob Herring 24 Mar '21

24 Mar '21

The nvidia,tegra210-ahub binding is missing schema for child nodes. This results in warnings if 'additionalProperties: false' is set (or when the tools implement 'unevaluatedProperties' support). Add the child nodes and reference their schema if one exists. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Jonathan Hunter <jonathanh(a)nvidia.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-tegra(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- This patch ideally should be applied before this series[1]. [1] https://lore.kernel.org/r/20210323163634.877511-1-robh@kernel.org/ .../bindings/sound/nvidia,tegra210-ahub.yaml | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml index e568d6c7dddd..d7a5eb77ed4f 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml @@ -69,6 +69,18 @@ properties: $ref: audio-graph-port.yaml# unevaluatedProperties: false +patternProperties: + '^i2s@[0-9a-f]+$': + type: object + + '^dmic@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-dmic.yaml# + + '^admaif@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-admaif.yaml# + required: - compatible - reg -- 2.27.0

2 1

[PATCH v4 0/9] soundwire: qcom: various improvements
by Srinivas Kandagatla 24 Mar '21

24 Mar '21

Thanks for reviewing v3 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Patch to add Auto Enumeration supported by the controller is also part of this series. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v3: - Fixed setting assigned bit during autoenumeration Srinivas Kandagatla (9): dt-bindings: soundwire: qcom: clarify data port bus parameters soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts soundwire: export sdw_compare_devid() and sdw_extract_slave_id() soundwire: qcom: add auto enumeration support soundwire: qcom: wait for enumeration to be complete in probe .../bindings/soundwire/qcom,sdw.txt | 20 + drivers/soundwire/bus.c | 4 +- drivers/soundwire/qcom.c | 529 ++++++++++++++---- include/linux/soundwire/sdw.h | 2 + 4 files changed, 442 insertions(+), 113 deletions(-) -- 2.21.0

4 12

[PATCH] ASoC: wm8960: Fix wrong bclk and lrclk with pll enabled for some chips
by Shengjiu Wang 24 Mar '21

24 Mar '21

The input MCLK is 12.288MHz, the desired output sysclk is 11.2896MHz and sample rate is 44100Hz, with the configuration pllprescale=2, postscale=sysclkdiv=1, some chip may have wrong bclk and lrclk output with pll enabled in master mode, but with the configuration pllprescale=1, postscale=2, the output clock is correct. >From Datasheet, the PLL performs best when f2 is between 90MHz and 100MHz when the desired sysclk output is 11.2896MHz or 12.288MHz, so sysclkdiv = 2 (f2/8) is the best choice. So search available sysclk_divs from 2 to 1 other than from 1 to 2. Fixes: 84fdc00d519f ("ASoC: codec: wm9860: Refactor PLL out freq search") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..cda9cd935d4f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -707,7 +707,13 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + /* + * From Datasheet, the PLL performs best when f2 is between + * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz + * or 12.288MHz, then sysclkdiv = 2 is the best choice. + * So search sysclk_divs from 2 to 1 other than from 1 to 2. + */ + for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) { if (sysclk_divs[i] == -1) continue; for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { -- 2.17.1

3 2

[PATCH] ASoC: SOF: Intel: hda: drop display power on/off in D0i3 flows
by Kai Vehmanen 24 Mar '21

24 Mar '21

Controller needs to ensure display power is enabled only for HDA controller reset. Drop the display power-up/down calls from D0i3 entry/exit paths. This was previously not possible as codec drivers could not resume the links, and instead controller kept the reference to display power. The state of display power had be maintained in the D0i3 entry/exit code. With commit 87fc20e4a0cb ("ASoC: SOF: Intel: hda: use hdac_ext fine-grained link management"), this is no longer needed and the code can be cleaned up. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Keyon Jie <yang.jie(a)intel.com> --- sound/soc/sof/intel/hda-dsp.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 6e24e1cb13f9..df4cb29b9a1b 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -727,8 +727,6 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) /* resume from D0I3 */ if (sdev->dsp_power_state.state == SOF_DSP_PM_D0) { - hda_codec_i915_display_power(sdev, true); - #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* power up links that were active before suspend */ list_for_each_entry(hlink, &bus->hlink_list, list) { @@ -839,9 +837,6 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) cancel_delayed_work_sync(&hda->d0i3_work); if (target_state == SOF_DSP_PM_D0) { - /* we can't keep a wakeref to display driver at suspend */ - hda_codec_i915_display_power(sdev, false); - /* Set DSP power state */ ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state); if (ret < 0) { base-commit: bf939446c357242b3306e88c5f48976940d29679 -- 2.29.2

2 1

[PATCH 0/8] ASoC: SOF: Intel: shutdown and core handling corrections
by Pierre-Louis Bossart 24 Mar '21

24 Mar '21

This patchset fixes a logical flow in the core status, improves shutdown support on Intel SOF platforms with an HDaudio controller and cleans-up ElkhartLake support. Guennadi Liakhovetski (1): ASoC: SOF: Intel: HDA: fix core status verification Libin Yang (5): ASoC: SOF: Intel: TGL: fix EHL ops ASoC: SOF: Intel: TGL: set shutdown callback to hda_dsp_shutdown ASoC: SOF: Intel: ICL: set shutdown callback to hda_dsp_shutdown ASoC: SOF: Intel: CNL: set shutdown callback to hda_dsp_shutdown ASoC: SOF: Intel: APL: set shutdown callback to hda_dsp_shutdown Pierre-Louis Bossart (2): ASoC: SOF: core: harden shutdown helper ASoC: SOF: Intel: move ELH chip info sound/soc/sof/core.c | 8 +++++++- sound/soc/sof/intel/apl.c | 3 ++- sound/soc/sof/intel/cnl.c | 19 ++----------------- sound/soc/sof/intel/hda-dsp.c | 21 +++++++++++++++++---- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 3 ++- sound/soc/sof/intel/pci-tgl.c | 2 +- sound/soc/sof/intel/tgl.c | 18 +++++++++++++++++- 8 files changed, 49 insertions(+), 26 deletions(-) -- 2.25.1

2 9

Intro and prep for Topology2.0
by GitHub pull_request - edited 23 Mar '21

23 Mar '21

alsa-project/alsa-lib pull request #129 was edited from ranj063: This pull request includes the preparatory commits paving the way for introducing Topology2.0. Existing alsatplg compiler functionality is unmodified. An example implementation of topology with Topology2.0 can be found here: https://github.com/thesofproject/sof/pull/3936 **Introduction to Topology 2.0** ----- Topology2.0 is a high level keyword extension on top of the existing ALSA conf topology format designed to: 1) Simplify the ALSA conf topology definitions by providing high level "classes" so topology designers need to write less config for common object definitions. 2) Allow simple reuse of objects. Define once and reuse (like M4) with the ability to alter objects configuration attributes from defaults. 3) Allow data type and value verification. This is not done today and frequently crops up in FW bug reports. **Common Topology Classes** ----------------------- Topology today has some common classes that are often reused throughout with slightly altered configurations. i.e. widgets (components), pipelines, dais and controls. Topology2.0 introduces the high level concept of reusable "class" like definition for a AIF_IN/AIF_OUT type object that can be used to create topology objects. **Common Topology Attributes** -------------------------- Topology defines a lot of attributes per object with different types and constraints. Today there is no easy way to validate type or constraints and this can lead to many hard to find problems in FW at runtime. A new keyword "DefineAttribute" has been added to define attribute type, size, min value, max value, enum_values. This then allows alsatplg to validate each topology object attribute. Topology Classes define the list of attributes that they use and whether the attribute is mandatory, can be overridden by parent users or is immutable. This also helps alsatplg emit the appropriate errors for attribute misuse. **Common Topology Arguments** ------------------------- Arguments are used to pass essential data needed for instantiating an object particularly needed for the object name. A new keyword "DefineArgument" has been added to define the arguments. The order in which the arguments are defined determines the name for the widget. For example, in the case of the host widget, the name would be constructed as "host.1.playback" where "1" is the pipeline_id argument value and "playback" is the direction argument value. **Attribute Inheritance:** ---------------------- One of the key features of Topology2.0 is howthe attribute values are propagated from a parent object to a child object. This is accomplished by adding attributes/arguments with the same name for a parent and an object. By doing so, when creating a child object, the value for the common attribute is populated from the parent. If the value is provided in the child object instance, then it overrides the value coming from the parent. **ALSA Conf Parser** ---------------- All the changes being proposed and discussed here must be 100% compliant with the ALSA conf parser. i.e. no syntax changes or changes to semantics for any existing keyword. It's intended that there will be NO changes to the ALSA conf parser and all topology building changes will be in the alsatplg compiler. Request URL : https://github.com/alsa-project/alsa-lib/pull/129 Patch URL : https://github.com/alsa-project/alsa-lib/pull/129.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Intro and prep for Topology2.0
by GitHub pull_request - edited 23 Mar '21

23 Mar '21

alsa-project/alsa-lib pull request #129 was edited from ranj063: This pull request includes the preparatory commits paving the way for introducing Topology2.0. Existing alsatplg compiler functionality is unmodified. An example implementation of topology with Topology2.0 can be found here: https://github.com/thesofproject/sof/pull/3936 **Introduction to Topology 2.0** ----- Topology2.0 is a high level keyword extension on top of the existing ALSA conf topology format designed to: 1) Simplify the ALSA conf topology definitions by providing high level "classes" so topology designers need to write less config for common object definitions. 2) Allow simple reuse of objects. Define once and reuse (like M4) with the ability to alter objects configuration attributes from defaults. 3) Allow data type and value verification. This is not done today and frequently crops up in FW bug reports. **Common Topology Classes** ----------------------- Topology today has some common classes that are often reused throughout with slightly altered configurations. i.e. widgets (components), pipelines, dais and controls. This PR introduces the high level concept of reusable "class" like definition for a AIF_IN/AIF_OUT type object that can be used to create topology objects. **Common Topology Attributes** -------------------------- Topology defines a lot of attributes per object with different types and constraints. Today there is no easy way to validate type or constraints and this can lead to many hard to find problems in FW at runtime. A new keyword "DefineAttribute" has been added to define attribute type, size, min value, max value, enum_values. This then allows alsatplg to validate each topology object attribute. Topology Classes define the list of attributes that they use and whether the attribute is mandatory, can be overridden by parent users or is immutable. This also helps alsatplg emit the appropriate errors for attribute misuse. **Common Topology Arguments** ------------------------- Arguments are used to pass essential data needed for instantiating an object particularly needed for the object name. A new keyword "DefineArgument" has been added to define the arguments. The order in which the arguments are defined determines the name for the widget. For example, in the case of the host widget, the name would be constructed as "host.1.playback" where "1" is the pipeline_id argument value and "playback" is the direction argument value. **Attribute Inheritance:** ---------------------- One of the key features of Topology2.0 is howthe attribute values are propagated from a parent object to a child object. This is accomplished by adding attributes/arguments with the same name for a parent and an object. By doing so, when creating a child object, the value for the common attribute is populated from the parent. If the value is provided in the child object instance, then it overrides the value coming from the parent. **ALSA Conf Parser** ---------------- All the changes being proposed and discussed here must be 100% compliant with the ALSA conf parser. i.e. no syntax changes or changes to semantics for any existing keyword. It's intended that there will be NO changes to the ALSA conf parser and all topology building changes will be in the alsatplg compiler. Request URL : https://github.com/alsa-project/alsa-lib/pull/129 Patch URL : https://github.com/alsa-project/alsa-lib/pull/129.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Intro and prep for Topology2.0
by GitHub pull_request - opened 23 Mar '21

23 Mar '21

alsa-project/alsa-lib pull request #129 was opened from ranj063: This pull request includes the preparatory commits paving the way for introducing Topology2.0. An example implementation of topology with Topology2.0 can be found here: https://github.com/thesofproject/sof/pull/3936 **Introduction to Topology 2.0** ----- Topology2.0 is a high level keyword extension on top of the existing ALSA conf topology format designed to: 1) Simplify the ALSA conf topology definitions by providing high level "classes" so topology designers need to write less config for common object definitions. 2) Allow simple reuse of objects. Define once and reuse (like M4) with the ability to alter objects configuration attributes from defaults. 3) Allow data type and value verification. This is not done today and frequently crops up in FW bug reports. **Common Topology Classes** ----------------------- Topology today has some common classes that are often reused throughout with slightly altered configurations. i.e. widgets (components), pipelines, dais and controls. This PR introduces the high level concept of reusable "class" like definition for a AIF_IN/AIF_OUT type object that can be used to create topology objects. **Common Topology Attributes** -------------------------- Topology defines a lot of attributes per object with different types and constraints. Today there is no easy way to validate type or constraints and this can lead to many hard to find problems in FW at runtime. A new keyword "DefineAttribute" has been added to define attribute type, size, min value, max value, enum_values. This then allows alsatplg to validate each topology object attribute. Topology Classes define the list of attributes that they use and whether the attribute is mandatory, can be overridden by parent users or is immutable. This also helps alsatplg emit the appropriate errors for attribute misuse. **Common Topology Arguments** ------------------------- Arguments are used to pass essential data needed for instantiating an object particularly needed for the object name. A new keyword "DefineArgument" has been added to define the arguments. The order in which the arguments are defined determines the name for the widget. For example, in the case of the host widget, the name would be constructed as "host.1.playback" where "1" is the pipeline_id argument value and "playback" is the direction argument value. **Attribute Inheritance:** ---------------------- One of the key features of Topology2.0 is howthe attribute values are propagated from a parent object to a child object. This is accomplished by adding attributes/arguments with the same name for a parent and an object. By doing so, when creating a child object, the value for the common attribute is populated from the parent. If the value is provided in the child object instance, then it overrides the value coming from the parent. **ALSA Conf Parser** ---------------- All the changes being proposed and discussed here must be 100% compliant with the ALSA conf parser. i.e. no syntax changes or changes to semantics for any existing keyword. It's intended that there will be NO changes to the ALSA conf parser and all topology building changes will be in the alsatplg compiler. Request URL : https://github.com/alsa-project/alsa-lib/pull/129 Patch URL : https://github.com/alsa-project/alsa-lib/pull/129.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] sound/i2c: Simplify the return expression of snd_i2c_bit_sendbyte()
by caizhichao 23 Mar '21

23 Mar '21

From: Zhichao Cai <caizhichao(a)yulong.com> Simplify the return expression. Signed-off-by: Zhichao Cai <caizhichao(a)yulong.com> --- sound/i2c/i2c.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index 847e3b6..dfcc87e 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -235,9 +235,7 @@ static int snd_i2c_bit_sendbyte(struct snd_i2c_bus *bus, unsigned char data) for (i = 7; i >= 0; i--) snd_i2c_bit_send(bus, !!(data & (1 << i))); err = snd_i2c_bit_ack(bus); - if (err < 0) - return err; - return 0; + return err < 0 ? err : 0; } static int snd_i2c_bit_readbyte(struct snd_i2c_bus *bus, int last) -- 1.9.1

2 1

[PATCH 0/5] soundwire: add missing \n in dev_err()
by Bard Liao 23 Mar '21

23 Mar '21

We fixed a lot of warnings in 2019 but the magic of copy-paste keeps adding new ones... Pierre-Louis Bossart (5): soundwire: intel: add missing \n in dev_err() soundwire: bandwidth_allocation: add missing \n in dev_err() soundwire: cadence: add missing \n in dev_err() soundwire: stream: add missing \n in dev_err() soundwire: qcom: add missing \n in dev_err() drivers/soundwire/cadence_master.c | 2 +- .../soundwire/generic_bandwidth_allocation.c | 4 ++-- drivers/soundwire/intel.c | 18 +++++++++--------- drivers/soundwire/qcom.c | 2 +- drivers/soundwire/stream.c | 10 +++++----- 5 files changed, 18 insertions(+), 18 deletions(-) -- 2.17.1

2 6

[PATCH v4 1/2] platform/x86: dell-privacy: Add support for Dell hardware privacy
by Perry Yuan 23 Mar '21

23 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> add support for Dell privacy driver for the Dell units equipped hardware privacy design, which protect users privacy of audio and camera from hardware level. Once the audio or camera privacy mode activated, any applications will not get any audio or video stream when user pressed ctrl+F4 hotkey, audio privacy mode will be enabled, micmute led will be also changed accordingly The micmute led is fully controlled by hardware & EC(embedded controller) and camera mute hotkey is Ctrl+F9. Currently design only emmits SW_CAMERA_LENS_COVER event while the camera lens shutter will be changed by EC & hw(hadware) control *The flow is like this: 1) User presses key. HW does stuff with this key (timeout timer is started) 2) WMI event is emitted from BIOS to kernel 3) WMI event is received by dell-privacy 4) KEY_MICMUTE emitted from dell-privacy 5) Userland picks up key and modifies kcontrol for SW mute 6) Codec kernel driver catches and calls ledtrig_audio_set, like this: ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led ? LED_ON :LED_OFF); 7) If "LED" is set to on dell-privacy notifies EC, and timeout is cancelled, HW mic mute activated. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- v3 -> v4: * fix format for Kconfig * add sysfs document * add flow comments to the privacy wmi/acpi driver * addressed feedback from Barnabás Pőcze[Thanks very much] * export privacy_valid to make the global state simpler to query * fix one issue which will block the dell-laptop driver to load when privacy driver invalid * addressed feedback from Pierre-Louis Bossart,remove the EC ID match v2 -> v3: * add sysfs attributes doc v1 -> v2: * query EC handle from EC driver directly. * fix some code style. * add KEY_END to keymap array. * clean platform device when cleanup called * use hexadecimal format for log print in dev_dbg * remove __set_bit for the report keys from probe. * fix keymap leak * add err_free_keymap in dell_privacy_wmi_probe * wmi driver will be unregistered if privacy_acpi_init() fails * add sysfs attribute files for user space query. * add leds micmute driver to privacy acpi * add more design info the commit info --- --- .../testing/sysfs-platform-dell-privacy-wmi | 32 ++ drivers/platform/x86/Kconfig | 17 + drivers/platform/x86/Makefile | 4 +- drivers/platform/x86/dell-laptop.c | 26 +- drivers/platform/x86/dell-privacy-acpi.c | 164 +++++++++ drivers/platform/x86/dell-privacy-wmi.c | 340 ++++++++++++++++++ drivers/platform/x86/dell-privacy-wmi.h | 35 ++ drivers/platform/x86/dell-wmi.c | 29 +- 8 files changed, 631 insertions(+), 16 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell-privacy-acpi.c create mode 100644 drivers/platform/x86/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell-privacy-wmi.h diff --git a/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi new file mode 100644 index 000000000000..20c15a51ec38 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi @@ -0,0 +1,32 @@ +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/devices_supported +Date: Feb 2021 +KernelVersion: 5.12 +Contact: "perry.yuan(a)dell.com>" +Description: + Display which dell hardware level privacy devices are supported + “Dell Privacy” is a set of HW, FW, and SW features to enhance + Dell’s commitment to platform privacy for MIC, Camera, and + ePrivacy screens. + The supported hardware privacy devices are: + - 0 = Not Supported + - 1 = Supported + - Bit0 -> Microphone + - Bit1 -> Camera + - Bit2 -> ePrivacy Screen + +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/current_state +Date: Feb 2021 +KernelVersion: 5.12 +Contact: "perry.yuan(a)dell.com>" +Description: + Allow user space to check current dell privacy device state. + Describes the Device State class exposed by BIOS which can be + consumed by various applications interested in knowing the Privacy + feature capabilities + There are three Bits for available states: + - 0 = Enabled + - 1 = Disabled + - Bit0 -> Microphone + - Bit1 -> Camera + - Bit2 -> ePrivacyScreen + diff --git a/drivers/platform/x86/Kconfig b/drivers/platform/x86/Kconfig index 91e6176cdfbd..e20f79389a39 100644 --- a/drivers/platform/x86/Kconfig +++ b/drivers/platform/x86/Kconfig @@ -491,6 +491,23 @@ config DELL_WMI_LED This adds support for the Latitude 2100 and similar notebooks that have an external LED. +config DELL_PRIVACY + tristate "Dell Hardware Privacy Support" + depends on ACPI + depends on ACPI_WMI + depends on INPUT + depends on DELL_LAPTOP + depends on LEDS_TRIGGER_AUDIO + select DELL_WMI + help + This driver provides support for the "Dell Hardware Privacy" feature + of Dell laptops. + Support for a micmute and camera mute privacy will be provided as + hardware button Ctrl+F4 and Ctrl+F9 hotkey. + + To compile this driver as a module, choose M here: the module will + be called dell_privacy. + config AMILO_RFKILL tristate "Fujitsu-Siemens Amilo rfkill support" depends on RFKILL diff --git a/drivers/platform/x86/Makefile b/drivers/platform/x86/Makefile index 581475f59819..18c430456de7 100644 --- a/drivers/platform/x86/Makefile +++ b/drivers/platform/x86/Makefile @@ -51,7 +51,9 @@ obj-$(CONFIG_DELL_WMI_DESCRIPTOR) += dell-wmi-descriptor.o obj-$(CONFIG_DELL_WMI_AIO) += dell-wmi-aio.o obj-$(CONFIG_DELL_WMI_LED) += dell-wmi-led.o obj-$(CONFIG_DELL_WMI_SYSMAN) += dell-wmi-sysman/ - +obj-$(CONFIG_DELL_PRIVACY) += dell-privacy.o +dell-privacy-objs := dell-privacy-wmi.o \ + dell-privacy-acpi.o # Fujitsu obj-$(CONFIG_AMILO_RFKILL) += amilo-rfkill.o obj-$(CONFIG_FUJITSU_LAPTOP) += fujitsu-laptop.o diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index 70edc5bb3a14..ec0dcc7fc17c 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -31,6 +31,8 @@ #include "dell-rbtn.h" #include "dell-smbios.h" +#include "dell-privacy-wmi.h" + struct quirk_entry { bool touchpad_led; bool kbd_led_not_present; @@ -90,10 +92,12 @@ static struct rfkill *wifi_rfkill; static struct rfkill *bluetooth_rfkill; static struct rfkill *wwan_rfkill; static bool force_rfkill; +static bool has_privacy; module_param(force_rfkill, bool, 0444); MODULE_PARM_DESC(force_rfkill, "enable rfkill on non whitelisted models"); + static const struct dmi_system_id dell_device_table[] __initconst = { { .ident = "Dell laptop", @@ -2205,11 +2209,17 @@ static int __init dell_init(void) dell_laptop_register_notifier(&dell_laptop_notifier); if (dell_smbios_find_token(GLOBAL_MIC_MUTE_DISABLE) && - dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); - if (ret < 0) - goto fail_led; + dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { + if (!privacy_valid) + has_privacy = true; + else + has_privacy = false; + if (!has_privacy) { + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); + if (ret < 0) + goto fail_led; + } } if (acpi_video_get_backlight_type() != acpi_backlight_vendor) @@ -2257,7 +2267,8 @@ static int __init dell_init(void) fail_get_brightness: backlight_device_unregister(dell_backlight_device); fail_backlight: - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); fail_led: dell_cleanup_rfkill(); fail_rfkill: @@ -2278,7 +2289,8 @@ static void __exit dell_exit(void) touchpad_led_exit(); kbd_led_exit(); backlight_device_unregister(dell_backlight_device); - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); dell_cleanup_rfkill(); if (platform_device) { platform_device_unregister(platform_device); diff --git a/drivers/platform/x86/dell-privacy-acpi.c b/drivers/platform/x86/dell-privacy-acpi.c new file mode 100644 index 000000000000..4c785b62c7c4 --- /dev/null +++ b/drivers/platform/x86/dell-privacy-acpi.c @@ -0,0 +1,164 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/bits.h> +#include <linux/device.h> +#include <linux/fs.h> +#include <linux/kernel.h> +#include <linux/leds.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/platform_device.h> +#include <linux/string.h> +#include <linux/sysfs.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define PRIVACY_PLATFORM_NAME "dell-privacy-acpi" +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" + +struct privacy_acpi_priv { + struct device *dev; + struct platform_device *platform_device; + struct led_classdev cdev; +}; +static struct privacy_acpi_priv *privacy_acpi; + +static int dell_privacy_micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct privacy_acpi_priv *priv = privacy_acpi; + acpi_status status; + acpi_handle handle; + static char *acpi_method = (char *)"ECAK"; + + handle = ec_get_handle(); + if (!handle) + return -EIO; + if (!acpi_has_method(handle, acpi_method)) + return -EIO; + status = acpi_evaluate_object(handle, acpi_method, NULL, NULL); + if (ACPI_FAILURE(status)) { + dev_err(priv->dev, "Error setting privacy EC ack value: %s\n", + acpi_format_exception(status)); + return -EIO; + } + pr_debug("set dell privacy micmute ec ack event done\n"); + return 0; +} + +static int dell_privacy_acpi_remove(struct platform_device *pdev) +{ + struct privacy_acpi_priv *priv = dev_get_drvdata(privacy_acpi->dev); + + led_classdev_unregister(&priv->cdev); + + return 0; +} +/* + * Pressing the mute key activates a time delayed circuit to physically cut + * off the mute. The LED is in the same circuit, so it reflects the true + * state of the HW mute. The reason for the EC "ack" is so that software + * can first invoke a SW mute before the HW circuit is cut off. Without SW + * cutting this off first does not affect the time delayed muting or status + * of the LED but there is a possibility of a "popping" noise. + * + * If the EC receives the SW ack, the circuit will be activated before the + * delay completed. + * + * Exposing as an LED device allows the codec drivers notification path to + * EC ACK to work + */ +static int dell_privacy_leds_setup(struct device *dev) +{ + struct privacy_acpi_priv *priv = dev_get_drvdata(dev); + int ret = 0; + + priv->cdev.name = "dell-privacy::micmute"; + priv->cdev.max_brightness = 1; + priv->cdev.brightness_set_blocking = dell_privacy_micmute_led_set; + priv->cdev.default_trigger = "audio-micmute"; + priv->cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = devm_led_classdev_register(dev, &priv->cdev); + if (ret) + return ret; + return 0; +} + +static int dell_privacy_acpi_probe(struct platform_device *pdev) +{ + int ret; + + platform_set_drvdata(pdev, privacy_acpi); + privacy_acpi->dev = &pdev->dev; + privacy_acpi->platform_device = pdev; + + ret = dell_privacy_leds_setup(&pdev->dev); + if (ret) + return -EIO; + + return 0; +} + +static struct platform_driver dell_privacy_platform_drv = { + .driver = { + .name = PRIVACY_PLATFORM_NAME, + }, + .probe = dell_privacy_acpi_probe, + .remove = dell_privacy_acpi_remove, +}; + +int __init dell_privacy_acpi_init(void) +{ + int err; + struct platform_device *pdev; + + if (!wmi_has_guid(DELL_PRIVACY_GUID)) + return -ENODEV; + + privacy_acpi = kzalloc(sizeof(*privacy_acpi), GFP_KERNEL); + if (!privacy_acpi) + return -ENOMEM; + + err = platform_driver_register(&dell_privacy_platform_drv); + if (err) + goto pdrv_err; + + pdev = platform_device_register_simple( + PRIVACY_PLATFORM_NAME, PLATFORM_DEVID_NONE, NULL, 0); + if (IS_ERR(pdev)) { + err = PTR_ERR(pdev); + goto pdev_err; + } + + return 0; + +pdev_err: + platform_device_unregister(pdev); +pdrv_err: + kfree(privacy_acpi); + return err; +} + +void __exit dell_privacy_acpi_exit(void) +{ + struct platform_device *pdev = to_platform_device(privacy_acpi->dev); + + platform_device_unregister(pdev); + platform_driver_unregister(&dell_privacy_platform_drv); + kfree(privacy_acpi); +} + +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("DELL Privacy ACPI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell-privacy-wmi.c b/drivers/platform/x86/dell-privacy-wmi.c new file mode 100644 index 000000000000..f90cb9d066de --- /dev/null +++ b/drivers/platform/x86/dell-privacy-wmi.c @@ -0,0 +1,340 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/input.h> +#include <linux/input/sparse-keymap.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" +#define MICROPHONE_STATUS BIT(0) +#define CAMERA_STATUS BIT(1) +#define PRIVACY_SCREEN_STATUS BIT(2) + +/* + * The wmi_list is used to store the privacy_priv struct with mutex protecting + */ +static LIST_HEAD(wmi_list); +static DEFINE_MUTEX(list_mutex); +/* + * global privacy state for other modules to query if the privacy driver loaded successfully + * or no privacy WMI device was detected at initial probe phase + */ +int privacy_valid = -EPROBE_DEFER; +EXPORT_SYMBOL_GPL(privacy_valid); + +struct privacy_wmi_data { + struct input_dev *input_dev; + struct wmi_device *wdev; + struct list_head list; + u32 features_present; + u32 last_status; +}; + +/* + * Keymap for WMI privacy events of type 0x0012 + */ +static const struct key_entry dell_wmi_keymap_type_0012[] = { + /* privacy mic mute */ + { KE_KEY, 0x0001, { KEY_MICMUTE } }, + /* privacy camera mute */ + { KE_SW, 0x0002, { SW_CAMERA_LENS_COVER } }, + { KE_END, 0}, +}; + +int dell_privacy_state(void) +{ + return privacy_valid; +} +EXPORT_SYMBOL_GPL(dell_privacy_state); +/* + * The flow of privacy event: + * 1) User presses key. HW does stuff with this key (timeout is started) + * 2) WMI event is emitted from BIOS + * 3) WMI event is received by dell-privacy + * 4) KEY_MICMUTE emitted from dell-privacy + * 5) Userland picks up key and modifies kcontrol for SW mute + * 6) Codec kernel driver catches and calls ledtrig_audio_set defined by + * dell-privacy-acpi driver. + * codec driver will call like this to switch micmute led state. + * ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led ? LED_ON :LED_OFF); + * 7) If "LED" is set to on dell-privacy notifies EC,and timeout is cancelled, + * HW mic mute activated. + */ +void dell_privacy_process_event(int type, int code, int status) +{ + struct privacy_wmi_data *priv; + const struct key_entry *key; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + goto error; + } + key = sparse_keymap_entry_from_scancode(priv->input_dev, (type << 16) | code); + if (!key) { + dev_dbg(&priv->wdev->dev, "Unknown key with type 0x%04x and code 0x%04x pressed\n", + type, code); + goto error; + } + switch (code) { + case DELL_PRIVACY_TYPE_AUDIO: /* Mic mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + break; + case DELL_PRIVACY_TYPE_CAMERA: /* Camera mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + break; + default: + dev_dbg(&priv->wdev->dev, "unknown event type 0x%04x 0x%04x", + type, code); + } +error: + mutex_unlock(&list_mutex); +} +EXPORT_SYMBOL_GPL(dell_privacy_process_event); + +static ssize_t devices_supported_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + return sysfs_emit(buf, "%x\n", priv->features_present); +} + +static ssize_t current_state_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + return sysfs_emit(buf, "%x\n", priv->last_status); +} + +static DEVICE_ATTR_RO(devices_supported); +static DEVICE_ATTR_RO(current_state); + +static struct attribute *privacy_attributes[] = { + &dev_attr_devices_supported.attr, + &dev_attr_current_state.attr, + NULL, +}; + +static const struct attribute_group privacy_attribute_group = { + .attrs = privacy_attributes +}; + +/* + * Describes the Device State class exposed by BIOS which can be consumed by + * various applications interested in knowing the Privacy feature capabilities. + * class DeviceState + * { + * [key, read] string InstanceName; + * [read] boolean ReadOnly; + * [WmiDataId(1), read] uint32 DevicesSupported; + * 0 – None, 0x1 – Microphone, 0x2 – Camera, 0x4 -ePrivacy Screen + * [WmiDataId(2), read] uint32 CurrentState; + * 0:Off; 1:On. Bit0 – Microphone, Bit1 – Camera, Bit2 - ePrivacyScreen + * }; + */ + +static int get_current_status(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + union acpi_object *obj_present; + u32 *buffer; + int ret = 0; + + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + return -EINVAL; + } + /* check privacy support features and device states */ + obj_present = wmidev_block_query(wdev, 0); + if (!obj_present) { + dev_err(&wdev->dev, "failed to read Binary MOF\n"); + ret = -EIO; + privacy_valid = ret; + return ret; + } + + if (obj_present->type != ACPI_TYPE_BUFFER) { + dev_err(&wdev->dev, "Binary MOF is not a buffer!\n"); + ret = -EIO; + privacy_valid = ret; + goto obj_free; + } + /* Although it's not technically a failure, this would lead to + * unexpected behavior + */ + if (obj_present->buffer.length != 8) { + dev_err(&wdev->dev, "Dell privacy buffer has unexpected length (%d)!\n", + obj_present->buffer.length); + ret = -EINVAL; + privacy_valid = ret; + goto obj_free; + } + buffer = (u32 *)obj_present->buffer.pointer; + priv->features_present = buffer[0]; + priv->last_status = buffer[1]; + privacy_valid = 0; + +obj_free: + kfree(obj_present); + return ret; +} + +static int dell_privacy_wmi_probe(struct wmi_device *wdev, const void *context) +{ + struct privacy_wmi_data *priv; + struct key_entry *keymap; + int ret, i; + + priv = devm_kzalloc(&wdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&wdev->dev, priv); + priv->wdev = wdev; + /* create evdev passing interface */ + priv->input_dev = devm_input_allocate_device(&wdev->dev); + if (!priv->input_dev) + return -ENOMEM; + /* remap the wmi keymap event to new keymap */ + keymap = kcalloc(ARRAY_SIZE(dell_wmi_keymap_type_0012), + sizeof(struct key_entry), GFP_KERNEL); + if (!keymap) { + ret = -ENOMEM; + goto err_free_dev; + } + /* remap the keymap code with Dell privacy key type 0x12 as prefix + * KEY_MICMUTE scancode will be reported as 0x120001 + */ + for (i = 0; i < ARRAY_SIZE(dell_wmi_keymap_type_0012); i++) { + keymap[i] = dell_wmi_keymap_type_0012[i]; + keymap[i].code |= (0x0012 << 16); + } + ret = sparse_keymap_setup(priv->input_dev, keymap, NULL); + if (ret) + return ret; + priv->input_dev->dev.parent = &wdev->dev; + priv->input_dev->name = "Dell Privacy Driver"; + priv->input_dev->id.bustype = BUS_HOST; + if (input_register_device(priv->input_dev)) { + pr_debug("input_register_device failed to register!\n"); + goto err_free_keymap; + } + mutex_lock(&list_mutex); + list_add_tail(&priv->list, &wmi_list); + mutex_unlock(&list_mutex); + + if (get_current_status(priv->wdev)) + goto err_free_input; + + ret = devm_device_add_group(&wdev->dev, &privacy_attribute_group); + if (ret) + goto err_free_input; + + kfree(keymap); + return 0; + +err_free_input: + input_unregister_device(priv->input_dev); +err_free_keymap: + privacy_valid = -ENODEV; + kfree(keymap); +err_free_dev: + input_free_device(priv->input_dev); + return ret; +} + +static int dell_privacy_wmi_remove(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + + mutex_lock(&list_mutex); + list_del(&priv->list); + mutex_unlock(&list_mutex); + privacy_valid = -ENODEV; + input_unregister_device(priv->input_dev); + + return 0; +} + +static const struct wmi_device_id dell_wmi_privacy_wmi_id_table[] = { + { .guid_string = DELL_PRIVACY_GUID }, + { }, +}; + +static struct wmi_driver dell_privacy_wmi_driver = { + .driver = { + .name = "dell-privacy", + }, + .probe = dell_privacy_wmi_probe, + .remove = dell_privacy_wmi_remove, + .id_table = dell_wmi_privacy_wmi_id_table, +}; + +static int __init init_dell_privacy(void) +{ + int ret; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) { + privacy_valid = -ENODEV; + pr_debug("Unable to detect available Dell privacy devices: %d\n", ret); + return privacy_valid; + } + + ret = wmi_driver_register(&dell_privacy_wmi_driver); + if (ret) { + pr_err("failed to initialize privacy wmi driver: %d\n", ret); + return ret; + } + + ret = dell_privacy_acpi_init(); + if (ret) { + pr_err("failed to initialize privacy acpi driver: %d\n", ret); + goto err_init; + } + + return 0; + +err_init: + wmi_driver_unregister(&dell_privacy_wmi_driver); + return ret; +} + +static void dell_privacy_wmi_exit(void) +{ + wmi_driver_unregister(&dell_privacy_wmi_driver); +} + +static void __exit exit_dell_privacy(void) +{ + dell_privacy_wmi_exit(); + dell_privacy_acpi_exit(); +} + +module_init(init_dell_privacy); +module_exit(exit_dell_privacy); + +MODULE_DEVICE_TABLE(wmi, dell_wmi_privacy_wmi_id_table); +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("Dell Privacy WMI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell-privacy-wmi.h b/drivers/platform/x86/dell-privacy-wmi.h new file mode 100644 index 000000000000..6dda2990df5d --- /dev/null +++ b/drivers/platform/x86/dell-privacy-wmi.h @@ -0,0 +1,35 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#ifndef _DELL_PRIVACY_WMI_H_ +#define _DELL_PRIVACY_WMI_H_ + +#if IS_ENABLED(CONFIG_DELL_PRIVACY) +extern int privacy_valid; +int dell_privacy_state(void); +void dell_privacy_process_event(int type, int code, int status); +#else /* CONFIG_DELL_PRIVACY */ +int privacy_valid = -ENODEV; +static inline int dell_privacy_state(void) +{ + return privacy_valid; +} + +static inline void dell_privacy_process_event(int type, int code, int status) +{} +#endif /* CONFIG_DELL_PRIVACY */ + +int dell_privacy_acpi_init(void); +void dell_privacy_acpi_exit(void); + +/* DELL Privacy Type */ +enum { + DELL_PRIVACY_TYPE_UNKNOWN = 0x0, + DELL_PRIVACY_TYPE_AUDIO, + DELL_PRIVACY_TYPE_CAMERA, +}; +#endif diff --git a/drivers/platform/x86/dell-wmi.c b/drivers/platform/x86/dell-wmi.c index bbdb3e860892..70e83c7c046d 100644 --- a/drivers/platform/x86/dell-wmi.c +++ b/drivers/platform/x86/dell-wmi.c @@ -13,20 +13,22 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt -#include <linux/kernel.h> -#include <linux/module.h> +#include <linux/acpi.h> +#include <linux/dmi.h> #include <linux/init.h> -#include <linux/slab.h> -#include <linux/types.h> #include <linux/input.h> #include <linux/input/sparse-keymap.h> -#include <linux/acpi.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> #include <linux/string.h> -#include <linux/dmi.h> +#include <linux/types.h> #include <linux/wmi.h> #include <acpi/video.h> + #include "dell-smbios.h" #include "dell-wmi-descriptor.h" +#include "dell-privacy-wmi.h" MODULE_AUTHOR("Matthew Garrett <mjg(a)redhat.com>"); MODULE_AUTHOR("Pali Rohár <pali(a)kernel.org>"); @@ -381,6 +383,7 @@ static void dell_wmi_notify(struct wmi_device *wdev, u16 *buffer_entry, *buffer_end; acpi_size buffer_size; int len, i; + int err; if (obj->type != ACPI_TYPE_BUFFER) { pr_warn("bad response type %x\n", obj->type); @@ -427,10 +430,9 @@ static void dell_wmi_notify(struct wmi_device *wdev, switch (buffer_entry[1]) { case 0x0000: /* One key pressed or event occurred */ - case 0x0012: /* Event with extended data occurred */ if (len > 2) dell_wmi_process_key(wdev, buffer_entry[1], - buffer_entry[2]); + buffer_entry[2]); /* Extended data is currently ignored */ break; case 0x0010: /* Sequence of keys pressed */ @@ -439,6 +441,17 @@ static void dell_wmi_notify(struct wmi_device *wdev, dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[i]); break; + case 0x0012: + err = dell_privacy_state(); + if (err == 0) { + dell_privacy_process_event(buffer_entry[1], + buffer_entry[3], buffer_entry[4]); + } else { + if (len > 2) + dell_wmi_process_key(wdev, buffer_entry[1], + buffer_entry[2]); + } + break; default: /* Unknown event */ pr_info("Unknown WMI event type 0x%x\n", (int)buffer_entry[1]); -- 2.25.1

5 7

Correct "rate" option
by GitHub pull_request - opened 23 Mar '21

23 Mar '21

alsa-project/alsa-utils pull request #83 was opened from nootc: https://github.com/alsa-project/alsa-utils/issues/82 Change "rate" option from "-c" to "-r". Request URL : https://github.com/alsa-project/alsa-utils/pull/83 Patch URL : https://github.com/alsa-project/alsa-utils/pull/83.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: Re: [PATCH v5 2/4] ASoC: Add Rockchip rk817 audio CODEC support
by Maciej Matuszczyk 22 Mar '21

22 Mar '21

Thanks! It works fine on my Odroid GO-Super and Odroid GO-Advance Tested-By: Maciej Matuszczyk <maccraft123mc(a)gmail.com>

1 0

alsaloop man page error
by GitHub issues - opened 22 Mar '21

22 Mar '21

alsa-project/alsa-utils issue #82 was opened from nootc: There is a small typo in the alsaloop man page: https://github.com/alsa-project/alsa-utils/blob/c9e9a79c6cfef3212bdb5f9be4f… "-c" is used for both "channels" and "rate". Thankfully, the source thinks otherwise and "-r" is used for "rate". Issue URL : https://github.com/alsa-project/alsa-utils/issues/82 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: Intel: Fix a typo
by Bhaskar Chowdhury 22 Mar '21

22 Mar '21

s/struture/structure/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/soc/intel/atom/sst-mfld-dsp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-dsp.h b/sound/soc/intel/atom/sst-mfld-dsp.h index 102b0e7eafb0..8d9e29b16e57 100644 --- a/sound/soc/intel/atom/sst-mfld-dsp.h +++ b/sound/soc/intel/atom/sst-mfld-dsp.h @@ -256,7 +256,7 @@ struct snd_sst_tstamp { u32 channel_peak[8]; } __packed; -/* Stream type params struture for Alloc stream */ +/* Stream type params structure for Alloc stream */ struct snd_sst_str_type { u8 codec_type; /* Codec type */ u8 str_type; /* 1 = voice 2 = music */ -- 2.31.0

2 1

[PATCH v6 0/5] Enable rk3066a HDMI sound
by Johan Jonker 22 Mar '21

22 Mar '21

Update the rk3066a HDMI documents with a #sound-dai-cells property. Include the code for sound in the HDMI driver. Add a simple-sound-card compatible node to rk3066a.dtsi, because I2S0 and HDMI TX are connected internally. And as last enable rk3066a HDMI sound in the rk3066a-mk808.dts file. make ARCH=arm CROSS_COMPILE=/usr/bin/arm-linux-gnueabi- -j4 cp ./arch/arm/boot/zImage ../zImage-dtb cat ./arch/arm/boot/dts/rk3066a-mk808.dtb >> ../zImage-dtb ../tools/rkcrc -k ../zImage-dtb ../mk808.img sudo ../tools/rkflashtool w 0x4000 0x8000 < ../mk808.img sudo ../tools/rkflashtool b Changed v6: remove patches that are applied to linux-next add platform_device_unregister() restyle Changed v5: removed unused variable fill frame structure Johan Jonker (4): dt-bindings: display: add #sound-dai-cells property to rockchip rk3066 hdmi ARM: dts: rockchip: rk3066a: add #sound-dai-cells to hdmi node ARM: dts: rockchip: add hdmi-sound node to rk3066a.dtsi ARM: dts: rockchip: enable hdmi_sound and i2s0 for rk3066a-mk808 Zheng Yang (1): drm: rockchip: add sound support to rk3066 hdmi driver .../display/rockchip/rockchip,rk3066-hdmi.yaml | 4 + arch/arm/boot/dts/rk3066a-mk808.dts | 8 + arch/arm/boot/dts/rk3066a.dtsi | 17 ++ drivers/gpu/drm/rockchip/Kconfig | 2 + drivers/gpu/drm/rockchip/rk3066_hdmi.c | 277 ++++++++++++++++++++- 5 files changed, 307 insertions(+), 1 deletion(-) -- 2.11.0

2 7

Re: include/linux/unaligned/be_byteshift.h:46:19: error: redefinition of 'get_unaligned_be32'
by Andy Shevchenko 22 Mar '21

22 Mar '21

On Fri, Mar 19, 2021 at 9:05 AM kernel test robot <lkp(a)intel.com> wrote: > > Hi Linus, > > FYI, the error/warning still remains. > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: 8b12a62a4e3ed4ae99c715034f557eb391d6b196 > commit: de8860b1ed4701ea7e6f760f02d79ca6a3b656a1 iio: magnetometer: Add driver for Yamaha YAS530 > date: 10 weeks ago > config: m68k-randconfig-s032-20210318 (attached as .config) > compiler: m68k-linux-gcc (GCC) 9.3.0 > reproduce: > wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # apt-get install sparse > # sparse version: v0.6.3-277-gc089cd2d-dirty > # https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… > git remote add linus https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > git fetch --no-tags linus master > git checkout de8860b1ed4701ea7e6f760f02d79ca6a3b656a1 > # save the attached .config to linux build tree > COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' ARCH=m68k > > If you fix the issue, kindly add following tag as appropriate > Reported-by: kernel test robot <lkp(a)intel.com> > > All errors (new ones prefixed by >>): > > In file included from include/linux/build_bug.h:5, > from include/linux/bitfield.h:10, > from drivers/iio/magnetometer/yamaha-yas530.c:22: Isn't it fixed already somewhere? ... > Kconfig warnings: (for reference only) > WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC > Depends on SOUND && !UML && SND && SND_SOC && SND_ATMEL_SOC && HAS_DMA > Selected by > - SND_ATMEL_SOC_SSC && SOUND && !UML && SND && SND_SOC && SND_ATMEL_SOC > - SND_ATMEL_SOC_SSC_PDC && SOUND && !UML && SND && SND_SOC && SND_ATMEL_SOC && ATMEL_SSC This one though is interesting. Cc Takashi and Mark. -- With Best Regards, Andy Shevchenko

3 2

[PATCH 1/2] ASoC: rt5682: Add a mclk name property to the device tree
by derek.fang＠realtek.com 22 Mar '21

22 Mar '21

From: "derek.fang" <derek(a)realtek.com> The patch adds a new property to specify the mclk name which be used. Signed-off-by: derek.fang <derek(a)realtek.com> diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index cd8c53d8497e..bee935ef2974 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -46,6 +46,8 @@ Optional properties: - realtek,dmic-clk-driving-high : Set the high driving of the DMIC clock out. +- realtek,mclk-name : Specify the mclk name that codec uses as clock provider. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 -- 2.17.1

2 2

[PATCH 1/2] ASoC: SOF: add a helper to get topology configured mclk
by Kai Vehmanen 22 Mar '21

22 Mar '21

From: Keyon Jie <yang.jie(a)linux.intel.com> Add helper sof_dai_ssp_mclk to get the topology configured MCLK from a pcm_runtime, return 0 if it is not available, and error if the dai type is not SSP at the moment. Export the helper for external use, e.g. from machine drivers. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof.h | 1 + sound/soc/sof/sof-audio.c | 27 +++++++++++++++++++++++++++ 2 files changed, 28 insertions(+) diff --git a/include/sound/sof.h b/include/sound/sof.h index 646a655c3c6b..b93bb8038080 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -103,5 +103,6 @@ struct sof_dev_desc { int sof_nocodec_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params)); +int sof_dai_get_mclk(struct snd_soc_pcm_runtime *rtd); #endif diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 3277489fee5e..928d7a46d820 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -433,6 +433,33 @@ struct snd_sof_dai *snd_sof_find_dai(struct snd_soc_component *scomp, return NULL; } +/* + * Helper to get SSP MCLK from a pcm_runtime. + * Return 0 if not exist. + */ +int sof_dai_get_mclk(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); + struct snd_sof_dai *dai = + snd_sof_find_dai(component, (char *)rtd->dai_link->name); + + /* use the tplg configured mclk if existed */ + if (!dai || !dai->dai_config) + return 0; + + switch (dai->dai_config->type) { + case SOF_DAI_INTEL_SSP: + return dai->dai_config->ssp.mclk_rate; + default: + /* not yet implemented for platforms other than the above */ + dev_err(rtd->dev, "mclk for dai_config->type %d not supported yet!\n", + dai->dai_config->type); + return -EINVAL; + } +} +EXPORT_SYMBOL(sof_dai_get_mclk); + /* * SOF Driver enumeration. */ base-commit: bb9faeccbd6c05c6c2525be90f663f8694dd8bf6 -- 2.29.2

2 2

[PATCH v4 1/2] ASoC: amd: Add support for RT5682 codec in machine driver
by Vijendar Mukunda 22 Mar '21

22 Mar '21

Add support for RT5682 codec in machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v3->v4: fixed clk_prepare_enable() for wclk instead of bclk v2->v3: updated SPDX license v1->v2: updated kconfig, spdx license, removed unnecessary initialization and updated comment sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-da7219-max98357a.c | 378 +++++++++++++++++++++++++++++++---- 2 files changed, 342 insertions(+), 41 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index a6ce000..43f5d29 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -5,14 +5,15 @@ config SND_SOC_AMD_ACP This option enables ACP DMA support on AMD platform. config SND_SOC_AMD_CZ_DA7219MX98357_MACH - tristate "AMD CZ support for DA7219 and MAX9835" + tristate "AMD CZ support for DA7219, RT5682 and MAX9835" select SND_SOC_DA7219 + select SND_SOC_RT5682_I2C select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help - This option enables machine driver for DA7219 and MAX9835. + This option enables machine driver for DA7219, RT5682 and MAX9835. config SND_SOC_AMD_CZ_RT5645_MACH tristate "AMD CZ support for RT5645" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 849288d..7fca4e78 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -1,27 +1,8 @@ -/* - * Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec - * - * Copyright 2017 Advanced Micro Devices, Inc. - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR - * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, - * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR - * OTHER DEALINGS IN THE SOFTWARE. - * - */ +// SPDX-License-Identifier: MIT +// +// Machine driver for AMD ACP Audio engine using DA7219, RT5682 & MAX98357 codec +// +//Copyright 2017-2021 Advanced Micro Devices, Inc. #include <sound/core.h> #include <sound/soc.h> @@ -41,14 +22,19 @@ #include "acp.h" #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" +#include "../codecs/rt5682.h" #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +#define RT5682_PLL_FREQ (48000 * 512) static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; extern bool bt_uart_enable; +void *soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -128,6 +114,88 @@ static void da7219_clk_disable(void) clk_disable_unprepare(da7219_dai_bclk); } +static int cz_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + dev_info(codec_dai->dev, "codec dai name = %s\n", codec_dai->name); + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CZ_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(rt5682_dai_wclk)) + return PTR_ERR(rt5682_dai_wclk); + + rt5682_dai_bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(rt5682_dai_bclk)) + return PTR_ERR(rt5682_dai_bclk); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &cz_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &cz_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + return 0; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." RT5682 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_wclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable wclk %d\n", ret); + return ret; + } + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_wclk); +} + static const unsigned int channels[] = { DUAL_CHANNEL, }; @@ -260,6 +328,118 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_rt5682_play_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return rt5682_clk_enable(substream); +} + +static void cz_rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -285,6 +465,31 @@ static const struct snd_soc_ops cz_dmic1_cap_ops = { .shutdown = cz_da7219_shutdown, }; +static const struct snd_soc_ops cz_rt5682_play_ops = { + .startup = cz_rt5682_play_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_cap_ops = { + .startup = cz_rt5682_cap_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_max_play_ops = { + .startup = cz_rt5682_max_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic0_cap_ops = { + .startup = cz_rt5682_dmic0_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { + .startup = cz_rt5682_dmic1_startup, + .shutdown = cz_rt5682_shutdown, +}; + SND_SOC_DAILINK_DEF(designware1, DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); SND_SOC_DAILINK_DEF(designware2, @@ -294,6 +499,8 @@ SND_SOC_DAILINK_DEF(designware3, SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(mx, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); SND_SOC_DAILINK_DEF(adau, @@ -353,6 +560,57 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { }, }; +static struct snd_soc_dai_link cz_dai_5682_98357[] = { + { + .name = "amd-rt5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = cz_rt5682_init, + .dpcm_playback = 1, + .ops = &cz_rt5682_play_ops, + SND_SOC_DAILINK_REG(designware1, rt5682, platform), + }, + { + .name = "amd-rt5682-cap", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_cap_ops, + SND_SOC_DAILINK_REG(designware2, rt5682, platform), + }, + { + .name = "amd-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &cz_rt5682_max_play_ops, + SND_SOC_DAILINK_REG(designware3, mx, platform), + }, + { + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic0_cap_ops, + SND_SOC_DAILINK_REG(designware3, adau, platform), + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic1_cap_ops, + SND_SOC_DAILINK_REG(designware2, adau, platform), + }, +}; + static const struct snd_soc_dapm_widget cz_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -368,6 +626,14 @@ static const struct snd_soc_dapm_route cz_audio_route[] = { {"PDM_DAT", NULL, "Int Mic"}, }; +static const struct snd_soc_dapm_route cz_rt5682_audio_route[] = { + {"Headphones", NULL, "HPOL"}, + {"Headphones", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Speakers", NULL, "Speaker"}, + {"PDM_DAT", NULL, "Int Mic"}, +}; + static const struct snd_kcontrol_new cz_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), @@ -388,6 +654,28 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct snd_soc_card cz_rt5682_card = { + .name = "acpr5682m98357", + .owner = THIS_MODULE, + .dai_link = cz_dai_5682_98357, + .num_links = ARRAY_SIZE(cz_dai_5682_98357), + .dapm_widgets = cz_widgets, + .num_dapm_widgets = ARRAY_SIZE(cz_widgets), + .dapm_routes = cz_rt5682_audio_route, + .controls = cz_mc_controls, + .num_controls = ARRAY_SIZE(cz_mc_controls), +}; + +void *soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static struct regulator_consumer_supply acp_da7219_supplies[] = { REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), @@ -425,29 +713,39 @@ static int cz_probe(struct platform_device *pdev) struct snd_soc_card *card; struct acp_platform_info *machine; struct regulator_dev *rdev; - - acp_da7219_cfg.dev = &pdev->dev; - rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, - &acp_da7219_cfg); - if (IS_ERR(rdev)) { - dev_err(&pdev->dev, "Failed to register regulator: %d\n", - (int)PTR_ERR(rdev)); - return -EINVAL; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)soc_is_rltk_max(dev); + if (!card) + return -ENODEV; + if (!strcmp(card->name, "acpd7219m98357")) { + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &cz_card; - cz_card.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - cz_card.name, ret); + card->name, ret); + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); return ret; } bt_uart_enable = !device_property_read_bool(&pdev->dev, @@ -457,7 +755,8 @@ static int cz_probe(struct platform_device *pdev) #ifdef CONFIG_ACPI static const struct acpi_device_id cz_audio_acpi_match[] = { - { "AMD7219", 0 }, + { "AMD7219", (unsigned long)&cz_card }, + { "AMDI5682", (unsigned long)&cz_rt5682_card}, {}, }; MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match); @@ -475,5 +774,6 @@ static struct platform_driver cz_pcm_driver = { module_platform_driver(cz_pcm_driver); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("DA7219 & MAX98357A audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("DA7219, RT5682 & MAX98357A audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

3 6

[RESEND PATCH v2] soundwire: bus: Fix device found flag correctly
by Srinivas Kandagatla 22 Mar '21

22 Mar '21

found flag is used to indicate SoundWire devices that are both enumerated on the bus and available in the device list. However this flag is not reset correctly after one iteration, This could miss some of the devices that are enumerated on the bus but not in device list. So reset this correctly to fix this issue! Fixes: d52d7a1be02c ("soundwire: Add Slave status handling helpers") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- For some reason I ended up deleting Fixes tag so resending with it! drivers/soundwire/bus.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9ab4a6264e0..188e391c5a83 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -712,7 +712,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) struct sdw_slave *slave, *_s; struct sdw_slave_id id; struct sdw_msg msg; - bool found = false; + bool found; int count = 0, ret; u64 addr; @@ -744,6 +744,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) sdw_extract_slave_id(bus, addr, &id); + found = false; /* Now compare with entries */ list_for_each_entry_safe(slave, _s, &bus->slaves, node) { if (sdw_compare_devid(slave, id) == 0) { -- 2.21.0

2 1

[PATCH v2 0/3] soundwire: clear bus clash/parity interrupt before the mask is enabled
by Bard Liao 22 Mar '21

22 Mar '21

The SoundWire specification allows a Slave device to report a bus clash or parity error with the in-band interrupt mechanism. Unfortunately, on some platforms, these errors are randomly reported and don't seem to be valid. This series suggests the addition of a Master level quirk to discard such interrupts. The quirk should in theory have been added at the Slave level, but since the problem was detected with different generations of Slave devices it's hard to point to a specific IP. The problem might also be board-dependent and hence dealing with a Master quirk is simpler. v2: - Reorder the patches sequence. - Add comments about each quirk. - Use u64 quirks. Bard Liao (3): soundwire: add master quirks for bus clash and parity soundwire: bus: handle master quirks for bus clash and parity soundwire: intel: add master quirks for bus clash and parity drivers/soundwire/bus.c | 39 +++++++++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 3 +++ include/linux/soundwire/sdw.h | 22 ++++++++++++++++++++ 3 files changed, 64 insertions(+) -- 2.17.1

2 5

[PATCH] ALSA: core: avoid -Wempty-body warnings
by Arnd Bergmann 22 Mar '21

22 Mar '21

From: Arnd Bergmann <arnd(a)arndb.de> Building with 'make W=1' shows some warnings about empty function-style macros: sound/core/pcm_memory.c: In function 'preallocate_pages': sound/core/pcm_memory.c:236:49: error: suggest braces around empty body in an 'if' statement [-Werror=empty-body] 236 | preallocate_info_init(substream); sound/core/seq_device.c: In function 'snd_seq_device_dev_register': sound/core/seq_device.c:163:41: error: suggest braces around empty body in an 'if' statement [-Werror=empty-body] 163 | queue_autoload_drivers(); Change them to empty inline functions, which are more robust here. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/core/oss/pcm_oss.c | 8 ++++++-- sound/core/pcm_memory.c | 4 +++- sound/core/seq_device.c | 15 ++++++++++++--- 3 files changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 142fc751a847..86c39ee01aaa 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -3069,8 +3069,12 @@ static void snd_pcm_oss_proc_done(struct snd_pcm *pcm) } } #else /* !CONFIG_SND_VERBOSE_PROCFS */ -#define snd_pcm_oss_proc_init(pcm) -#define snd_pcm_oss_proc_done(pcm) +static inline void snd_pcm_oss_proc_init(struct snd_pcm *pcm) +{ +} +static inline void snd_pcm_oss_proc_done(struct snd_pcm *pcm) +{ +} #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 289dd1fd8fe7..c440f67b34a3 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -210,7 +210,9 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) } #else /* !CONFIG_SND_VERBOSE_PROCFS */ -#define preallocate_info_init(s) +static inline void preallocate_info_init(struct snd_pcm_substream *substream) +{ +} #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index 7ed13cb32ef8..382275c5b193 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -133,10 +133,19 @@ void snd_seq_device_load_drivers(void) flush_work(&autoload_work); } EXPORT_SYMBOL(snd_seq_device_load_drivers); -#define cancel_autoload_drivers() cancel_work_sync(&autoload_work) + +static inline void cancel_autoload_drivers(void) +{ + cancel_work_sync(&autoload_work); +} #else -#define queue_autoload_drivers() /* NOP */ -#define cancel_autoload_drivers() /* NOP */ +static inline void queue_autoload_drivers(void) +{ +} + +static inline void cancel_autoload_drivers(void) +{ +} #endif /* -- 2.29.2

2 1

[PATCH] sound: remove unnecessary blank
by Fengfei Xi 22 Mar '21

22 Mar '21

Remove unnecessary blank when calling kmalloc_array(). Signed-off-by: Fengfei Xi <xi.fengfei(a)h3c.com> --- sound/oss/dmasound/dmasound_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 49679aa86..78bb9dfe2 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -421,7 +421,7 @@ static int sq_allocate_buffers(struct sound_queue *sq, int num, int size) return 0; sq->numBufs = num; sq->bufSize = size; - sq->buffers = kmalloc_array (num, sizeof(char *), GFP_KERNEL); + sq->buffers = kmalloc_array(num, sizeof(char *), GFP_KERNEL); if (!sq->buffers) return -ENOMEM; for (i = 0; i < num; i++) { -- 2.17.1

2 1

[PATCH 0/3] ALSA: bebob: enable MIDI message transmission for multiple ports
by Takashi Sakamoto 22 Mar '21

22 Mar '21

Hi, Although below models supported by ALSA bebob driver have multiple MIDI ports, the driver just adds one pair of MIDI ports for ALSA Rawmidi interface: * M-Audio ProjectMix I/O * ESI Quatafire 610 The cause comes from two bugs: * The driver registers the number of MIDI conformant data channels into AM824 data block processing layer, instead of the number of MIDI ports. * For Quatafire, the driver counts plugs with MIDI type, however the number of physical MIDI ports is expressed in the number of channels on the plugs. This patchset enables MIDI message transmission for multiple ports. Takashi Sakamoto (3): ALSA: bebob: code refactoring for stream format detection ALSA: bebob: detect the number of available MIDI ports ALSA: bebob: enable to deliver MIDI messages for multiple ports sound/firewire/bebob/bebob.h | 2 + sound/firewire/bebob/bebob_command.c | 36 ++++++ sound/firewire/bebob/bebob_stream.c | 163 ++++++++++++++------------- 3 files changed, 120 insertions(+), 81 deletions(-) -- 2.27.0

2 4

[PATCH v2 0/3] ALSA: hdsp and hdspm, don't disable device if not enabled
by Tong Zhang 22 Mar '21

22 Mar '21

This series fixes issues in hdsp and hdspm. The drivers in question want to disable a device that is not enabled on error path. v2: add fix to rme9652 Tong Zhang (3): ALSA: hdsp: don't disable if not enabled ALSA: hdspm: don't disable if not enabled ALSA: rme9652: don't disable if not enabled sound/pci/rme9652/hdsp.c | 10 ++++++---- sound/pci/rme9652/hdspm.c | 10 ++++++---- sound/pci/rme9652/rme9652.c | 10 ++++++---- 3 files changed, 18 insertions(+), 12 deletions(-) -- 2.25.1

2 10

[PATCH v3 1/2] ALSA: hda/realtek: fix a determine_headset_type issue for a Dell AIO
by Hui Wang 22 Mar '21

22 Mar '21

We found a recording issue on a Dell AIO, users plug a headset-mic and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. On this machine, the internal mic is not connected to the codec, the "Input Source" is headset mic by default. And when users plug a headset, the determine_headset_type() will be called immediately, the codec on this AIO is alc274, the delay time for this codec in the determine_headset_type() is only 80ms, the delay is too short to correctly determine the headset type, the fail rate is nearly 99% when users plug the headset with the normal speed. Other codecs set several hundred ms delay time, so here I change the delay time to 850ms for alc2x4 series, after this change, the fail rate is zero unless users plug the headset slowly on purpose. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8239e5efc12d..442e555de44c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5263,7 +5263,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0274: case 0x10ec0294: alc_process_coef_fw(codec, coef0274); - msleep(80); + msleep(850); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x00f0) == 0x00f0; break; -- 2.25.1

2 3

[PATCH v2] soundwire: cadence_master: fix kernel-doc
by Pierre-Louis Bossart 22 Mar '21

22 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. drivers/soundwire/cadence_master.c:914: warning: expecting prototype for To update slave status in a work since we will need to handle(). Prototype was for cdns_update_slave_status_work() instead drivers/soundwire/cadence_master.c:976: warning: expecting prototype for sdw_cdns_enable_slave_interrupt(). Prototype was for cdns_enable_slave_interrupts() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index d05442e646a3..e5c3f304b8f6 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -905,7 +905,7 @@ irqreturn_t sdw_cdns_irq(int irq, void *dev_id) EXPORT_SYMBOL(sdw_cdns_irq); /** - * To update slave status in a work since we will need to handle + * cdns_update_slave_status_work - update slave status in a work since we will need to handle * other interrupts eg. CDNS_MCP_INT_RX_WL during the update slave * process. * @work: cdns worker thread @@ -968,7 +968,7 @@ int sdw_cdns_exit_reset(struct sdw_cdns *cdns) EXPORT_SYMBOL(sdw_cdns_exit_reset); /** - * sdw_cdns_enable_slave_interrupt() - Enable SDW slave interrupts + * cdns_enable_slave_interrupts() - Enable SDW slave interrupts * @cdns: Cadence instance * @state: boolean for true/false */ -- 2.25.1

2 1

[PATCH v5 2/2] ASoC: rt715:add micmute led state control supports
by Perry Yuan 22 Mar '21

22 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing When micmute hotkey pressed by user, soft mute will need to be enabled firstly in case of pop noise, and codec driver need to react to mic mute event to EC(embedded controller) notifying that SW mute is completed Then EC will do the hardware mute physically within the timeout reached This patch allow codec rt715 and rt715 sdca driver to change the local micmute led state. Dell privacy led trigger driver will ack EC when micmute key pressed through this micphone led control interface like hda_generic provided ACPI method defined in dell-privacy micmute led trigger will be called for notifying the EC that software mute has been completed, then hardware audio circuit solution controlled by EC will switch the audio input source off/on Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> -------- v4 -> v5: * rebase to latest 5.12 rc4 upstream kernel v3 -> v4: * remove unused debug log * remove compile flag of DELL privacy * move the micmute_led to local from rt715_priv * when Jaroslav upstream his gerneric LED trigger driver,I will rebase this patch,please consider merge this at first https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ v2 -> v3: * simplify the patch to reuse some val value * add more detail to the commit info v1 -> v2: * fix some format issue -------- --- sound/soc/codecs/rt715-sdca.c | 11 +++++++++++ sound/soc/codecs/rt715.c | 12 ++++++++++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 20528afbdc57..70e1181f1dee 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -11,6 +11,7 @@ #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -358,6 +359,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = p->invert; int err; + bool micmute_led; for (i = 0; i < 4; i++) { if (ucontrol->value.integer.value[i] != rt715->kctl_switch_orig[i]) { @@ -393,6 +395,15 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, if (err < 0) return err; } + /* Micmute LED state changed by muted/unmute switch */ + if (invert) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) { + micmute_led = LED_OFF; + } else { + micmute_led = LED_ON; + } + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } return k_changed; } diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 34c3357e943b..13686c5239b3 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -88,6 +89,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt; + bool micmute_led; if (strstr(ucontrol->id.name, "Main Capture Switch") || strstr(ucontrol->id.name, "Main Capture Volume")) @@ -95,6 +97,16 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, else loop_cnt = 1; + /* Micmute LED state changed by muted/unmute switch */ + if (mc->invert) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) { + micmute_led = LED_OFF; + } else { + micmute_led = LED_ON; + } + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } + for (j = 0; j < loop_cnt; j++) { /* Can't use update bit function, so read the original value first */ if (loop_cnt == 1) { -- 2.25.1

1 0

[PATCH v5 0/2] hardware-privacy-implementation-for-dell-laptop
by Perry Yuan 22 Mar '21

22 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> Hi All, This patch set is a new driver for Dell mobile platform that has the hardware privacy feature including micmute,camera mute. For micmute led control, the hotkey is Fn + F4, it is a hardware based mute state, and new privacy will prevent micphone void input from hardware layer, any application cannot get voice data when micmute activated. Camera mute use a new hardware design to control the camrea shutter. When video is muted, no OS application should be functionally able to capture images of the person/environment in front of the system Older history: [1]https://patchwork.kernel.org/project/platform-driver-x86/patch/2020122813… [2]https://patchwork.kernel.org/project/alsa-devel/patch/20201103125859.8759… [3]https://www.spinics.net/lists/alsa-devel/msg120537.html [4]https://github.com/thesofproject/linux/pull/2660 [5]https://github.com/thesofproject/linux/issues/2496 Perry Yuan (2): platform/x86: dell-privacy: Add support for Dell hardware privacy ASoC: rt715:add micmute led state control supports .../testing/sysfs-platform-dell-privacy-wmi | 32 ++ drivers/platform/x86/dell/Kconfig | 16 + drivers/platform/x86/dell/Makefile | 3 + drivers/platform/x86/dell/dell-laptop.c | 23 +- drivers/platform/x86/dell/dell-privacy-acpi.c | 164 +++++++++ drivers/platform/x86/dell/dell-privacy-wmi.c | 340 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 35 ++ drivers/platform/x86/dell/dell-wmi.c | 27 +- sound/soc/codecs/rt715-sdca.c | 11 + sound/soc/codecs/rt715.c | 12 + 10 files changed, 650 insertions(+), 13 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-acpi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h -- 2.25.1

1 0

[PATCH] ASoC: Intel: Fix a typo
by Bhaskar Chowdhury 22 Mar '21

22 Mar '21

s/struture/structure/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/soc/intel/atom/sst-mfld-dsp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-dsp.h b/sound/soc/intel/atom/sst-mfld-dsp.h index 5795f98e04d4..102b0e7eafb0 100644 --- a/sound/soc/intel/atom/sst-mfld-dsp.h +++ b/sound/soc/intel/atom/sst-mfld-dsp.h @@ -358,7 +358,7 @@ struct snd_wma_params { u8 reserved; /* reserved */ } __packed; -/* Codec params struture */ +/* Codec params structure */ union snd_sst_codec_params { struct snd_pcm_params pcm_params; struct snd_mp3_params mp3_params; -- 2.31.0

1 0

[alsa-lib][PATCH 0/6] add API of equality and comparison for a pair of control element IDs
by Takashi Sakamoto 22 Mar '21

22 Mar '21

Hi, This patchset is a fix for bug issued in the message thread[1]. In this development period, alsa-lib got new API as implementation for one of comparison algorithms to a pair of control element IDs. However, it has several issues. At first, the name, 'snd_ctl_elem_id_compare()', is inappropriate since it implements one of comparison algorithms. The name itself implies the algorithm is single and unique for control element ID. However, the target structure, 'struct snd_ctl_elem_id', is hybrid and compound one. We can not find such single and unique comparison algorithm for it. Secondary, it subtracts a pair of values in fields of 'unsigned int' type in storage size of the type. It brings integer overflow. Tertiary, it has simple bug to compare subdevice field in the same structure. Essentially, equality is different from comparison. In a point of programming, implementation for comparison algorithm can have more overhead than implementation for equality. In this meaning, it's better to add different API for them. This patchset adds new API below: * for equality * snd_ctl_elem_id_equal_by_numid() * snd_ctl_elem_id_equal_by_tuple() * for each comparison algorithm * snd_ctl_elem_id_compare_by_numid() * snd_ctl_elem_id_compare_by_tuple_arithmetic() I've got bothered to decide the name of API for the case to use tuples. Here I use the word, 'tuple', which comes from documentation of alsa-lib[2]. Furthermore, this patchset adds test program for them since equality and comparison are quite basic method to operate data. It's better to have no bug. Finally, the issued API, 'snd_ctl_elem_id_compare()' is dropped. After merging the patchset, I'm going to post additional patch to alsa-utils to fix issued line[3]. [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2021-March/181738.html [2] https://github.com/alsa-project/alsa-lib/blob/master/src/control/control.c#… [3] https://github.com/alsa-project/alsa-utils/blob/master/alsactl/clean.c#L55 Regards Takashi Sakamoto (6): test: ctl-elem-id: add test program for future APIs relevant to control element ID ctl: add API to check equality between a pair of control element IDs by numid ctl: add API to check equality between a pair of control element IDs by tuple ctl: add API to compare a pair of control element IDs by numid ctl: add API to compare a pair of control element IDs by one of algorithms according to tuple ctl: drop deprecated API to compare a pair of control element IDs include/control.h | 5 +- src/control/control.c | 135 ++++++++++++++---- test/lsb/Makefile.am | 6 +- test/lsb/ctl-elem-id.c | 301 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 418 insertions(+), 29 deletions(-) create mode 100644 test/lsb/ctl-elem-id.c -- 2.27.0

2 13

Fwd: Pioneer DJ DDJ-SR2 sound degradation
by Geraldo 21 Mar '21

21 Mar '21

Good news: my JACK transport has been rolling for more than 2 hours without sound degradation. I'll leave it overnight just to be sure sound degradation has been fixed and not just postponed. But yes, I think I managed to engage implicit feedback sync for the playback on my Pioneer DJ DDJ-SR2. Dr. Iwai, my initial assesment was ALSA not honoring .ep_attr on quirks-table.h was the culprit for the Incompatible EP Setup messages. Turns out that for implicit feedback sync the EP is opened twice, and the ep->opened refcount prevents that. Once I commented out ep->opened++; inside snd_usb_endpoint_open on endpoint.c the kernel was able to open the EP twice(one time for data, second time for sync). I could use some help in coding an elegant way of going around the ep->opened refcount instead of simpy disabling it! Once that particular problem is sorted out I'll be able to provide a proper PATCH for all users of Pioneer gear to test implicit feedback sync. Here's the relevant dyndbg info for the success case: [35084.131891] usb 3-2: 0:1: added playback implicit_fb sync_ep 82, iface 0:1 [35084.131905] usb 3-2: Creating new data endpoint #1 [35084.131911] usb 3-2: Creating new data endpoint #82 [35084.131917] xhci_hcd 0000:00:14.0: xhci_check_bandwidth called for udev 0000000052e95692 [35084.132102] usb 3-2: 0:1 Set sample rate 44100, clock 0 [35084.132116] xhci_hcd 0000:00:14.0: xhci_check_bandwidth called for udev 0000000052e95692 [35084.132163] usb 3-2: 0:1 Set sample rate 44100, clock 0 [35084.132592] usbcore: registered new interface driver snd-usb-audio [35096.805245] usb 3-2: Open EP 0x82, iface=0:1, idx=1 [35096.805254] usb 3-2: channels=6, rate=44100, format=S24_3LE, period_bytes=9216, periods=2, implicit_fb=0 [35096.805259] usb 3-2: Setting usb interface 0:0 for EP 0x82 [35096.805264] xhci_hcd 0000:00:14.0: xhci_check_bandwidth called for udev 0000000052e95692 [35096.805475] usb 3-2: Setting usb interface 0:1 for EP 0x82 [35096.805486] xhci_hcd 0000:00:14.0: add ep 0x1, slot id 32, new drop flags = 0x0, new add flags = 0x5 [35096.805492] xhci_hcd 0000:00:14.0: add ep 0x82, slot id 32, new drop flags = 0x0, new add flags = 0x25 [35096.805495] xhci_hcd 0000:00:14.0: xhci_check_bandwidth called for udev 0000000052e95692 [35096.805499] xhci_hcd 0000:00:14.0: Adding 2 ep ctxs, 6 now active. [35096.805503] xhci_hcd 0000:00:14.0: Recalculating BW for rootport 2 [35096.805505] xhci_hcd 0000:00:14.0: Final bandwidth: 282, Limit: 1607, Reserved: 322, Available: 62 percent [35096.805510] xhci_hcd 0000:00:14.0: // Ding dong! [35096.805652] xhci_hcd 0000:00:14.0: Successful Endpoint Configure command [35096.805895] usb 3-2: 0:1 Set sample rate 44100, clock 0 [35096.805902] usb 3-2: Setting params for data EP 0x82, pipe 0x14180 [35096.805910] usb 3-2: Set up 12 URBS, ret=0 [35096.805996] usb 3-2: Open EP 0x1, iface=0:1, idx=0 [35096.806000] usb 3-2: channels=4, rate=44100, format=S24_3LE, period_bytes=6144, periods=2, implicit_fb=1 [35096.806003] usb 3-2: Open EP 0x82, iface=0:1, idx=1 [35096.806005] usb 3-2: channels=6, rate=44100, format=S24_3LE, period_bytes=9216, periods=2, implicit_fb=0 [35096.806008] usb 3-2: 0:1 Set sample rate 44100, clock 0 [35096.806011] usb 3-2: Setting params for data EP 0x1, pipe 0xc100 [35096.806018] usb 3-2: Set up 12 URBS, ret=0 [35096.806021] usb 3-2: 0:1 Set sample rate 44100, clock 0 [35096.806025] usb 3-2: Setting params for data EP 0x82, pipe 0x14180 [35096.806029] usb 3-2: Set up 12 URBS, ret=0 [35096.806048] usb 3-2: Starting data EP 0x1 (running 0) [35096.806051] usb 3-2: No URB submission due to implicit fb sync [35096.806053] usb 3-2: Starting data EP 0x82 (running 0) [35096.806076] usb 3-2: 12 URBs submitted for EP 0x82 [35096.807337] usb 3-2: Starting data EP 0x82 (running 1) [35096.807343] usb 3-2: 0:1 Start Capture PCM [35096.807346] usb 3-2: 0:1 Start Playback PCM On Sun, Mar 21, 2021 at 5:22 AM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Sun, 21 Mar 2021 05:31:11 +0100, > Geraldo wrote: > > > > Until we manage to engage implicit feedback sync for Pioneer gear I'm > > afraid the clock will drift on DUPLEX mode. > > > > I reread the Pioneer related source code and implicit feedback sync was > > disabled because it gave Incompatible EP Setup messages. I should know, I > > was a reporter on that regression. > > > > Only those messages are fake. They arise because our ALSA code it's not > > honoring .ep_attr on quirks-table.h specifically in regards to > > USB_ENDPOINT_USAGE_IMPLICIT_FB. > > That's right. The parser reads from the original descriptor, not the > one from the quirk table. That is, the implicit fb quirk has to be > declared explicitly in another quirk table. And, the implicit fb is > skipped for those devices because they didn't work; see > sound/usb/implicit.c. > > > In fact the new improved implicit feedback sync code inside pcm.c that sets > > sync upon probe is failing to set implicit_fb to 1. I commented the > > conditional for testing and I told endpoint_compatible to always return > > true. > > > > Voilà, JACK started in duplex mode without nasty Incompatible EP Setup > > problems and supposedly with implicit feedback sync on. We are not setting > > bits 5..7 of bmAttributes, at least that's what I get from the missing > > Usage field in lsusb. > > > > My JACK transport has been rolling for 17 minutes and if it goes for two > > complete hours without distortion passing-thru vinyl on Mixxxx I'll provide > > the list with a more elegant PATCH and not this gambiarra as we say on > > Brazil. > > > > Hopefully their will be implicit feedback sync for Pioneer :-) > > I know there are quite a few users with Pioneer devices around here, > so let's hope that they can test the implicit feedback things and find > the culprit of the buggy behavior. Unfortunately it's quite hard to > debug such a thing remotely without the hardware from my side. > > > Takashi > > > > > > > > On Thu, Mar 18, 2021 at 7:41 PM Geraldo <geraldogabriel(a)gmail.com> wrote: > > > > > > Hi everyone, > > > > > > I'm running stable kernel 5.11.6 together with my Pioneer DJ DDJ-SR2 and > > JACK. Mixxx is running on top of JACK. > > > > > > > > > After I start JACK I have about 40 minutes of good sound quality from the > > inputs. After this period the sound will begin to degrade slowly until it > > is almost pure distortion. > > > > > > The issue is gone if I restart the JACK server, no need to reboot the > > hardware. The description I gave above sounds a lot like soundcard clock > > drift (which happens to those using more than one soundcard. You know) > > which motivates me to believe we missed a bug somewhere around the recent > > changes introduced in implicit feedback sync for Pioneer devices > > > > > > Dyndbg shows nothing useful I guess. No xruns reported. > > > > > > I've recorded a 1KHz tone and after 50+ minutes the sound degradation > > kicks in. Only problem is even after cutting and encoding to mp3 128kbps > > it's still a 10 megabytes file. I suppose I could post it to Soundcloud or > > Mixcloud if anyone is interested. > > > > > > Is anyone else experiencing these symptoms with Pioneer hardware? Note > > that the sound degradation kicks in after about an hour, more or less, and > > it's for the inputs only. Outputs continue to operate fine without the need > > to restart the JACK server every hour. > > > > > > Thanks everybody, > > > Geraldo > >

1 0

pcm: dshare - fix shared memory pointer check
by GitHub pull_request - opened 21 Mar '21

21 Mar '21

alsa-project/alsa-lib pull request #128 was opened from VanithaChannaiah: Currently shared memory pointer is initialized to 0 and set to -1 in some, but not in all error paths. In cleanup path of open the shm pointer is only compared to be non-NULL before dereferencing it which leads to SEGFAULT in case it was set to -1. This patch initializes pointer to -1 to have a unique identification for invalid pointer and also checks for pointer being not -1 on access in cleanup path. Signed-off-by: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/128 Patch URL : https://github.com/alsa-project/alsa-lib/pull/128.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Pioneer DJ DDJ-SR2 sound degradation
by Geraldo 21 Mar '21

21 Mar '21

Hi everyone, I'm running stable kernel 5.11.6 together with my Pioneer DJ DDJ-SR2 and JACK. Mixxx is running on top of JACK. After I start JACK I have about 40 minutes of good sound quality from the inputs. After this period the sound will begin to degrade slowly until it is almost pure distortion. The issue is gone if I restart the JACK server, no need to reboot the hardware. The description I gave above sounds a lot like soundcard clock drift (which happens to those using more than one soundcard. You know) which motivates me to believe we missed a bug somewhere around the recent changes introduced in implicit feedback sync for Pioneer devices Dyndbg shows nothing useful I guess. No xruns reported. I've recorded a 1KHz tone and after 50+ minutes the sound degradation kicks in. Only problem is even after cutting and encoding to mp3 128kbps it's still a 10 megabytes file. I suppose I could post it to Soundcloud or Mixcloud if anyone is interested. Is anyone else experiencing these symptoms with Pioneer hardware? Note that the sound degradation kicks in after about an hour, more or less, and it's for the inputs only. Outputs continue to operate fine without the need to restart the JACK server every hour. Thanks everybody, Geraldo

2 2

[PATCH 0/2] ALSA: hdsp and hdspm, don't disable device if not enabled
by Tong Zhang 21 Mar '21

21 Mar '21

This series fixes issues in hdsp and hdspm. The drivers in question want to disable a device that is not enabled on error path. Tong Zhang (2): ALSA: hdsp: don't disable if not enabled ALSA: hdspm: don't disable if not enabled sound/pci/rme9652/hdsp.c | 10 ++++++---- sound/pci/rme9652/hdspm.c | 10 ++++++---- 2 files changed, 12 insertions(+), 8 deletions(-) -- 2.25.1

1 2

[PATCH v2 1/2] ALSA: hda/realtek: fix a determine_headset_type issue for a Dell AIO
by Hui Wang 20 Mar '21

20 Mar '21

We found a recording issue on a Dell AIO, users plug a headset-mic and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. On this machine, the internal mic is not connected to the codec, the "Input Source" is headset mic by default. And when users plug a headset, the determine_headset_type() will be called immediately, the codec on this AIO is alc274, the delay time for this codec in the determine_headset_type() is only 80ms, the delay is too short to correctly determine the headset type, the fail rate is nearly 99% when users plug the headset with the normal speed. Other codecs set several hundred ms delay time, so here I change the delay time to 850ms for alc2x4 series, after this change, the fail rate is zero unless users plug the headset slowly on purpose. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8239e5efc12d..442e555de44c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5263,7 +5263,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0274: case 0x10ec0294: alc_process_coef_fw(codec, coef0274); - msleep(80); + msleep(850); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x00f0) == 0x00f0; break; -- 2.25.1

2 3

[PATCH v5 4/4] arm64: dts: rockchip: add rk817 codec to Odroid Go Advance
by Chris Morgan 19 Mar '21

19 Mar '21

Add the new rk817 codec driver to the Odroid Go Advance. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- .../boot/dts/rockchip/rk3326-odroid-go2.dts | 35 ++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/rockchip/rk3326-odroid-go2.dts b/arch/arm64/boot/dts/rockchip/rk3326-odroid-go2.dts index 97fb93e1cc00..e48dcf430efc 100644 --- a/arch/arm64/boot/dts/rockchip/rk3326-odroid-go2.dts +++ b/arch/arm64/boot/dts/rockchip/rk3326-odroid-go2.dts @@ -161,6 +161,29 @@ blue_led: led-0 { }; }; + rk817-sound { + compatible = "simple-audio-card"; + simple-audio-card,format = "i2s"; + simple-audio-card,name = "rockchip,rk817-codec"; + simple-audio-card,mclk-fs = <256>; + simple-audio-card,widgets = + "Microphone", "Mic Jack", + "Headphone", "Headphones", + "Speaker", "Speaker"; + simple-audio-card,routing = + "MICL", "Mic Jack", + "Headphones", "HPOL", + "Headphones", "HPOR", + "Speaker", "SPKO"; + simple-audio-card,hp-det-gpio = <&gpio2 RK_PC6 GPIO_ACTIVE_HIGH>; + simple-audio-card,cpu { + sound-dai = <&i2s1_2ch>; + }; + simple-audio-card,codec { + sound-dai = <&rk817>; + }; + }; + vccsys: vccsys { compatible = "regulator-fixed"; regulator-name = "vcc3v8_sys"; @@ -266,10 +289,13 @@ rk817: pmic@20 { interrupt-parent = <&gpio0>; interrupts = <RK_PB2 IRQ_TYPE_LEVEL_LOW>; pinctrl-names = "default"; - pinctrl-0 = <&pmic_int>; + pinctrl-0 = <&pmic_int>, <&i2s1_2ch_mclk>; wakeup-source; #clock-cells = <1>; clock-output-names = "rk808-clkout1", "xin32k"; + clocks = <&cru SCLK_I2S1_OUT>; + clock-names = "mclk"; + #sound-dai-cells = <0>; vcc1-supply = <&vccsys>; vcc2-supply = <&vccsys>; @@ -428,6 +454,11 @@ regulator-state-mem { }; }; }; + + rk817_codec: codec { + mic-in-differential; + status = "okay"; + }; }; }; @@ -439,6 +470,8 @@ &i2c1 { /* I2S 1 Channel Used */ &i2s1_2ch { + resets = <&cru SRST_I2S1>, <&cru SRST_I2S1_H>; + reset-names = "reset-m", "reset-h"; status = "okay"; }; -- 2.25.1

1 0

[PATCH v5 3/4] dt-bindings: Add Rockchip rk817 audio CODEC support
by Chris Morgan 19 Mar '21

19 Mar '21

Create dt-binding documentation to document rk817 codec. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- .../bindings/sound/rockchip,rk817-codec.yaml | 60 +++++++++++++++++++ 1 file changed, 60 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml new file mode 100644 index 000000000000..845886e2fdf4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-rk817.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk817 audio codec + +description: + The rk817 codec is an I2C codec integrated with every Rockchip + rk817 PMIC MFD. + +properties: + compatible: + enum: + - rockchip,rk817-codec + + "#sound-dai-cells": + const: 1 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + mic-in-differential: + description: the microphone is in differential mode. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - clocks + - clock-names + +additionalProperties: false + +examples: + - | +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>, <&i2s_8ch_mclk>; + #sound-dai-cells = <0>; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + + ........ + + rk817_codec: codec { + mic-in-differential; + status = "okay"; + }; + + ........ + +}; -- 2.25.1

1 0

[PATCH v5 1/4] mfd: Add Rockchip rk817 audio CODEC support
by Chris Morgan 19 Mar '21

19 Mar '21

Add rk817 codec support cell to rk808 mfd driver. Changes from v4: Move register definitions from rk817_codec.h to main rk808.h register definitions. Add volatile register for codec bits. Add default values for codec bits. Removed of_compatible. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- drivers/mfd/rk808.c | 89 +++++++++++++++++++++++++++++++++++++++ include/linux/mfd/rk808.h | 81 +++++++++++++++++++++++++++++++++++ 2 files changed, 170 insertions(+) diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..b92f4c6f503e 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -65,6 +65,9 @@ static bool rk817_is_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case RK817_SECONDS_REG ... RK817_WEEKS_REG: case RK817_RTC_STATUS_REG: +#ifdef CONFIG_SND_SOC_RK817 + case RK817_CODEC_DTOP_LPT_SRST: +#endif case RK817_INT_STS_REG0: case RK817_INT_STS_REG1: case RK817_INT_STS_REG2: @@ -163,6 +166,11 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { @@ -201,6 +209,87 @@ static const struct rk808_reg_data rk808_pre_init_reg[] = { static const struct rk808_reg_data rk817_pre_init_reg[] = { {RK817_RTC_CTRL_REG, RTC_STOP, RTC_STOP}, +#ifdef CONFIG_SND_SOC_RK817 + /* Codec specific registers */ + { RK817_CODEC_DTOP_VUCTL, MASK_ALL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, MASK_ALL, 0x06 }, + { RK817_CODEC_AADC_CFG0, MASK_ALL, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC2, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_NG, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_HPF, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AHP_ANTI0, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_ANTI1, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_CFG0, MASK_ALL, 0xe0 }, + { RK817_CODEC_AHP_CFG1, MASK_ALL, 0x1f }, + { RK817_CODEC_AHP_CP, MASK_ALL, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, MASK_ALL, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, MASK_ALL, 0x44 }, + { RK817_CODEC_APLL_CFG0, MASK_ALL, 0x04 }, + { RK817_CODEC_APLL_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_APLL_CFG2, MASK_ALL, 0x30 }, + { RK817_CODEC_APLL_CFG3, MASK_ALL, 0x19 }, + { RK817_CODEC_APLL_CFG4, MASK_ALL, 0x65 }, + { RK817_CODEC_APLL_CFG5, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_CKM, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_RSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, MASK_ALL, 0x00 }, +#endif {RK817_GPIO_INT_CFG, RK817_INT_POL_MSK, RK817_INT_POL_L}, {RK817_SYS_CFG(1), RK817_HOTDIE_TEMP_MSK | RK817_TSD_TEMP_MSK, RK817_HOTDIE_105 | RK817_TSD_140}, diff --git a/include/linux/mfd/rk808.h b/include/linux/mfd/rk808.h index e07f6e61cd38..a96e6d43ca06 100644 --- a/include/linux/mfd/rk808.h +++ b/include/linux/mfd/rk808.h @@ -437,6 +437,87 @@ enum rk809_reg_id { #define RK817_RTC_COMP_LSB_REG 0x10 #define RK817_RTC_COMP_MSB_REG 0x11 +/* RK817 Codec Registers */ +#define RK817_CODEC_DTOP_VUCTL 0x12 +#define RK817_CODEC_DTOP_VUCTIME 0x13 +#define RK817_CODEC_DTOP_LPT_SRST 0x14 +#define RK817_CODEC_DTOP_DIGEN_CLKE 0x15 +#define RK817_CODEC_AREF_RTCFG0 0x16 +#define RK817_CODEC_AREF_RTCFG1 0x17 +#define RK817_CODEC_AADC_CFG0 0x18 +#define RK817_CODEC_AADC_CFG1 0x19 +#define RK817_CODEC_DADC_VOLL 0x1a +#define RK817_CODEC_DADC_VOLR 0x1b +#define RK817_CODEC_DADC_SR_ACL0 0x1e +#define RK817_CODEC_DADC_ALC1 0x1f +#define RK817_CODEC_DADC_ALC2 0x20 +#define RK817_CODEC_DADC_NG 0x21 +#define RK817_CODEC_DADC_HPF 0x22 +#define RK817_CODEC_DADC_RVOLL 0x23 +#define RK817_CODEC_DADC_RVOLR 0x24 +#define RK817_CODEC_AMIC_CFG0 0x27 +#define RK817_CODEC_AMIC_CFG1 0x28 +#define RK817_CODEC_DMIC_PGA_GAIN 0x29 +#define RK817_CODEC_DMIC_LMT1 0x2a +#define RK817_CODEC_DMIC_LMT2 0x2b +#define RK817_CODEC_DMIC_NG1 0x2c +#define RK817_CODEC_DMIC_NG2 0x2d +#define RK817_CODEC_ADAC_CFG0 0x2e +#define RK817_CODEC_ADAC_CFG1 0x2f +#define RK817_CODEC_DDAC_POPD_DACST 0x30 +#define RK817_CODEC_DDAC_VOLL 0x31 +#define RK817_CODEC_DDAC_VOLR 0x32 +#define RK817_CODEC_DDAC_SR_LMT0 0x35 +#define RK817_CODEC_DDAC_LMT1 0x36 +#define RK817_CODEC_DDAC_LMT2 0x37 +#define RK817_CODEC_DDAC_MUTE_MIXCTL 0x38 +#define RK817_CODEC_DDAC_RVOLL 0x39 +#define RK817_CODEC_DDAC_RVOLR 0x3a +#define RK817_CODEC_AHP_ANTI0 0x3b +#define RK817_CODEC_AHP_ANTI1 0x3c +#define RK817_CODEC_AHP_CFG0 0x3d +#define RK817_CODEC_AHP_CFG1 0x3e +#define RK817_CODEC_AHP_CP 0x3f +#define RK817_CODEC_ACLASSD_CFG1 0x40 +#define RK817_CODEC_ACLASSD_CFG2 0x41 +#define RK817_CODEC_APLL_CFG0 0x42 +#define RK817_CODEC_APLL_CFG1 0x43 +#define RK817_CODEC_APLL_CFG2 0x44 +#define RK817_CODEC_APLL_CFG3 0x45 +#define RK817_CODEC_APLL_CFG4 0x46 +#define RK817_CODEC_APLL_CFG5 0x47 +#define RK817_CODEC_DI2S_CKM 0x48 +#define RK817_CODEC_DI2S_RSD 0x49 +#define RK817_CODEC_DI2S_RXCR1 0x4a +#define RK817_CODEC_DI2S_RXCR2 0x4b +#define RK817_CODEC_DI2S_RXCMD_TSD 0x4c +#define RK817_CODEC_DI2S_TXCR1 0x4d +#define RK817_CODEC_DI2S_TXCR2 0x4e +#define RK817_CODEC_DI2S_TXCR3_TXCMD 0x4f + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + #define RK817_POWER_EN_REG(i) (0xb1 + (i)) #define RK817_POWER_SLP_EN_REG(i) (0xb5 + (i)) -- 2.25.1

1 0

[PATCH -next] ASoC: rt1019: make symbol 'rt1019_i2c_driver' static
by 'w00385741 19 Mar '21

19 Mar '21

From: Wei Yongjun <weiyongjun1(a)huawei.com> The sparse tool complains as follows: sound/soc/codecs/rt1019.c:927:19: warning: symbol 'rt1019_i2c_driver' was not declared. Should it be static? This symbol is not used outside of rt1019.c, so this commit marks it static. Fixes: 7ec79d3850d0 ("ASoC: rt1019: add rt1019 amplifier driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt1019.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index fb275686a00f..5138f028d9f2 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -924,7 +924,7 @@ static int rt1019_i2c_probe(struct i2c_client *i2c, &soc_component_dev_rt1019, rt1019_dai, ARRAY_SIZE(rt1019_dai)); } -struct i2c_driver rt1019_i2c_driver = { +static struct i2c_driver rt1019_i2c_driver = { .driver = { .name = "rt1019", .of_match_table = of_match_ptr(rt1019_of_match),

2 1

[PATCH 00/14] ASoC: tidyup error message timing
by Kuninori Morimoto 19 Mar '21

19 Mar '21

Hi Mark Indicating error message when failed case is very useful for debuging. In many case, it uses below style. int function(...) { ... return ret; } int caller(...) { ... ret = function(...); if (ret < 0) dev_err(...) ... } This is not so bad, but in this style *each caller* needs to indicate duplicate same error message, and some caller is forgetting to do it. And caller can't indicate detail function() error information. I know many people have many opinion, but if function() indicates error message, we can get same and detail information without forgot, and it is better. This patch-set tidyup to do it. int function(...) { ... if (ret < 0) dev_err(...) return ret; } int caller(...) { ... ret = function(...); ... } Kuninori Morimoto (14): ASoC: soc-pcm: indicate error message at soc_pcm_open() ASoC: soc-pcm: indicate error message at soc_pcm_hw_params() ASoC: soc-pcm: indicate error message at soc_pcm_prepare() ASoC: soc-pcm: indicate error message at dpcm_path_get() ASoC: soc-pcm: indicate error message at dpcm_be_dai_trigger() ASoC: soc-pcm: indicate error message at dpcm_apply_symmetry() ASoC: soc-pcm: indicate error message at dpcm_run_update_startup/shutdown() ASoC: soc-pcm: indicate error message at dpcm_fe/be_dai_startup() ASoC: soc-pcm: indicate error message at dpcm_fe/be_dai_hw_params() ASoC: soc-pcm: indicate error message at dpcm_fe/be_dai_prepare() ASoC: soc-pcm: don't indicate error message for soc_pcm_hw_free() ASoC: soc-pcm: don't indicate error message for dpcm_be_dai_hw_free() ASoC: don't indicate error message for snd_soc_[pcm_]dai_xxx() ASoC: don't indicate error message for snd_soc_[pcm_]component_xxx() include/sound/soc-dpcm.h | 2 +- sound/soc/soc-compress.c | 9 +- sound/soc/soc-core.c | 22 +---- sound/soc/soc-dapm.c | 24 ++--- sound/soc/soc-pcm.c | 197 +++++++++++++++++++-------------------- 5 files changed, 108 insertions(+), 146 deletions(-) -- 2.25.1

2 15

[PATCH v2 0/3] Fix SND_HDA_PREALLOC issue
by Amadeusz Sławiński 19 Mar '21

19 Mar '21

For context it started with user reporting failures when running arecord without any error or warning in dmesg (after fixing some configuration problems thet they had). https://bugzilla.kernel.org/show_bug.cgi?id=201251#c279 After spending time investigating the issue it was narrowed to quite big setting of CONFIG_SND_HDA_PREALLOC_SIZE (4096). When looking at code https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… there is a limit of memory per card: max_alloc_per_card = 32UL * 1024UL * 1024UL When SND_HDA_PREALLOC_SIZE is set to 4096 it only has memory to alloc for 8 frontends, while Skylake HDA card has 10 of them (6 playback and 4 capture), so preallocated memory is exhausted while probing. In consequence 2 of FEs end without allocated memory. It can be workarounded on user side with setting SND_HDA_PREALLOC_SIZE to lower value, other is changing memory limit per card. However in order to not waste user memory, change maximum allocation size on HDA controller to 4MB and force automatical memory allocation insted of preallocated one. First patch adds prints, so similar issues can be easily identified in the future. Second changes maximum size of hda buffer to reasonable value of 4MB. And last one reverts patch which allowed setting prealloc size on X86 platforms. Amadeusz Sławiński (3): ALSA: pcm: Add debug print on memory allocation failure ALSA: hda: Change AZX_MAX_BUF_SIZE from 1GB to 4MB ALSA: hda: Revert "ALSA: hda: Allow setting preallocation again for x86" include/sound/hda_register.h | 8 ++++++-- sound/core/pcm_memory.c | 8 ++++++++ sound/hda/Kconfig | 7 +++---- 3 files changed, 17 insertions(+), 6 deletions(-) -- 2.25.1

2 4

[PATCH v4 1/3] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 19 Mar '21

19 Mar '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Changes from v3: Created separate set_pll() call. Created user visible gain control for microphone. Check for return value of clk_prepare_enable(). Remove duplicate clk_prepare_enable(). Moved documentation to a separate commit. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 706 +++++++++++++++++++++++++++++++++ sound/soc/codecs/rk817_codec.h | 94 +++++ 4 files changed, 805 insertions(+) create mode 100644 sound/soc/codecs/rk817_codec.c create mode 100644 sound/soc/codecs/rk817_codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..d835145ad65e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1059,6 +1060,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..045acf05da5c --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,706 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "rk817_codec.h" + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct rk808 *rk817; + struct clk *mclk; + unsigned int stereo_sysclk; + bool mic_in_differential; +}; + +static const struct reg_default rk817_reg_defaults[] = { + { RK817_CODEC_DTOP_VUCTL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, 0x06 }, + { RK817_CODEC_AADC_CFG0, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, 0x00 }, + { RK817_CODEC_DADC_VOLL, 0x00 }, + { RK817_CODEC_DADC_VOLR, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, 0x00 }, + { RK817_CODEC_DADC_ALC1, 0x00 }, + { RK817_CODEC_DADC_ALC2, 0x00 }, + { RK817_CODEC_DADC_NG, 0x00 }, + { RK817_CODEC_DADC_HPF, 0x00 }, + { RK817_CODEC_DADC_RVOLL, 0xff }, + { RK817_CODEC_DADC_RVOLR, 0xff }, + { RK817_CODEC_AMIC_CFG0, 0x70 }, + { RK817_CODEC_AMIC_CFG1, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, 0x66 }, + { RK817_CODEC_DMIC_LMT1, 0x00 }, + { RK817_CODEC_DMIC_LMT2, 0x00 }, + { RK817_CODEC_DMIC_NG1, 0x00 }, + { RK817_CODEC_DMIC_NG2, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, 0x00 }, + { RK817_CODEC_ADAC_CFG1, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, 0x82 }, + { RK817_CODEC_DDAC_VOLL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, 0x00 }, + { RK817_CODEC_DDAC_LMT1, 0x00 }, + { RK817_CODEC_DDAC_LMT2, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, 0xff }, + { RK817_CODEC_AHP_ANTI0, 0x00 }, + { RK817_CODEC_AHP_ANTI1, 0x00 }, + { RK817_CODEC_AHP_CFG0, 0xe0 }, + { RK817_CODEC_AHP_CFG1, 0x1f }, + { RK817_CODEC_AHP_CP, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, 0x44 }, + { RK817_CODEC_APLL_CFG0, 0x04 }, + { RK817_CODEC_APLL_CFG1, 0x00 }, + { RK817_CODEC_APLL_CFG2, 0x30 }, + { RK817_CODEC_APLL_CFG3, 0x19 }, + { RK817_CODEC_APLL_CFG4, 0x65 }, + { RK817_CODEC_APLL_CFG5, 0x01 }, + { RK817_CODEC_DI2S_CKM, 0x01 }, + { RK817_CODEC_DI2S_RSD, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, 0x00 }, +}; + +static bool rk817_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_LPT_SRST: + return true; + default: + return false; + } +} + +static bool rk817_codec_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_VUCTL: + case RK817_CODEC_DTOP_VUCTIME: + case RK817_CODEC_DTOP_LPT_SRST: + case RK817_CODEC_DTOP_DIGEN_CLKE: + case RK817_CODEC_AREF_RTCFG0: + case RK817_CODEC_AREF_RTCFG1: + case RK817_CODEC_AADC_CFG0: + case RK817_CODEC_AADC_CFG1: + case RK817_CODEC_DADC_VOLL: + case RK817_CODEC_DADC_VOLR: + case RK817_CODEC_DADC_SR_ACL0: + case RK817_CODEC_DADC_ALC1: + case RK817_CODEC_DADC_ALC2: + case RK817_CODEC_DADC_NG: + case RK817_CODEC_DADC_HPF: + case RK817_CODEC_DADC_RVOLL: + case RK817_CODEC_DADC_RVOLR: + case RK817_CODEC_AMIC_CFG0: + case RK817_CODEC_AMIC_CFG1: + case RK817_CODEC_DMIC_PGA_GAIN: + case RK817_CODEC_DMIC_LMT1: + case RK817_CODEC_DMIC_LMT2: + case RK817_CODEC_DMIC_NG1: + case RK817_CODEC_DMIC_NG2: + case RK817_CODEC_ADAC_CFG0: + case RK817_CODEC_ADAC_CFG1: + case RK817_CODEC_DDAC_POPD_DACST: + case RK817_CODEC_DDAC_VOLL: + case RK817_CODEC_DDAC_VOLR: + case RK817_CODEC_DDAC_SR_LMT0: + case RK817_CODEC_DDAC_LMT1: + case RK817_CODEC_DDAC_LMT2: + case RK817_CODEC_DDAC_MUTE_MIXCTL: + case RK817_CODEC_DDAC_RVOLL: + case RK817_CODEC_DDAC_RVOLR: + case RK817_CODEC_AHP_ANTI0: + case RK817_CODEC_AHP_ANTI1: + case RK817_CODEC_AHP_CFG0: + case RK817_CODEC_AHP_CFG1: + case RK817_CODEC_AHP_CP: + case RK817_CODEC_ACLASSD_CFG1: + case RK817_CODEC_ACLASSD_CFG2: + case RK817_CODEC_APLL_CFG0: + case RK817_CODEC_APLL_CFG1: + case RK817_CODEC_APLL_CFG2: + case RK817_CODEC_APLL_CFG3: + case RK817_CODEC_APLL_CFG4: + case RK817_CODEC_APLL_CFG5: + case RK817_CODEC_DI2S_CKM: + case RK817_CODEC_DI2S_RSD: + case RK817_CODEC_DI2S_RXCR1: + case RK817_CODEC_DI2S_RXCR2: + case RK817_CODEC_DI2S_RXCMD_TSD: + case RK817_CODEC_DI2S_TXCR1: + case RK817_CODEC_DI2S_TXCR2: + case RK817_CODEC_DI2S_TXCR3_TXCMD: + return true; + default: + return false; + } +} + +/* + * This sets the codec up with the values defined in the default implementation including the APLL + * from the Rockchip vendor kernel. I do not know if these values are universal despite differing + * from the default values defined above and taken from the datasheet, or implementation specific. + * I don't have another implementation to compare from the Rockchip sources. Hard-coding for now. + */ + +static int rk817_init(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DDAC_SR_LMT0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DADC_SR_ACL0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_VUCTIME, 0xf4); + if (rk817->mic_in_differential) { + snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, + MIC_DIFF_EN); + }; + return 0; +} + +static int rk817_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + + return 0; +}; + +/* + * DDAC/DADC L/R volume setting + * 0db~-95db, 0.375db/step, for example: + * 0x00: 0dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, 0); + +/* + * PGA GAIN L/R volume setting + * 27db~-18db, 3db/step, for example: + * 0x0: -18dB + * 0xf: 27dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_gain_tlv, -1800, 2700); + +static const struct snd_kcontrol_new rk817_volume_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", RK817_CODEC_DADC_VOLL, + RK817_CODEC_DADC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_TLV("Mic Capture Gain", RK817_CODEC_DMIC_PGA_GAIN, 4, 0, 0xf, 0, + rk817_gain_tlv), +}; + +static const char *dac_mux_text[] = { + "HP", + "SPK", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(dac_enum, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = + SOC_DAPM_ENUM("Playback Mux", dac_enum); + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + + /* capture/playback common */ + SND_SOC_DAPM_SUPPLY("LDO Regulator", RK817_CODEC_AREF_RTCFG1, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBIAS Block", RK817_CODEC_AREF_RTCFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VAvg Buffer", RK817_CODEC_AREF_RTCFG1, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL Power", RK817_CODEC_APLL_CFG5, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX1 Transfer Start", RK817_CODEC_DI2S_RXCMD_TSD, 5, 0, NULL, 0), + + /* capture path common */ + SND_SOC_DAPM_SUPPLY("ADC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC Power On", RK817_CODEC_AMIC_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Transfer Start", RK817_CODEC_DI2S_TXCR3_TXCMD, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Right Justified", RK817_CODEC_DI2S_TXCR3_TXCMD, 3, 0, NULL, 0), + + /* capture path L */ + SND_SOC_DAPM_ADC("ADC L", "Capture", RK817_CODEC_AADC_CFG0, 7, 1), + SND_SOC_DAPM_SUPPLY("PGA L Power On", RK817_CODEC_AMIC_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L2", RK817_CODEC_AMIC_CFG0, 2, 0, NULL, 0), + + /* capture path R */ + SND_SOC_DAPM_ADC("ADC R", "Capture", RK817_CODEC_AADC_CFG0, 6, 1), + SND_SOC_DAPM_SUPPLY("PGA R Power On", RK817_CODEC_AMIC_CFG0, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R2", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + + /* playback path common */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Bias", RK817_CODEC_ADAC_CFG1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mute Off", RK817_CODEC_DDAC_MUTE_MIXCTL, 0, 1, NULL, 0), + + /* playback path speaker */ + SND_SOC_DAPM_SUPPLY("Class D Mode", RK817_CODEC_DDAC_MUTE_MIXCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("High Pass Filter", RK817_CODEC_DDAC_MUTE_MIXCTL, 7, 0, NULL, 0), + SND_SOC_DAPM_DAC("SPK DAC", "Playback", RK817_CODEC_ADAC_CFG1, 2, 1), + SND_SOC_DAPM_SUPPLY("Enable Class D", RK817_CODEC_ACLASSD_CFG1, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Disable Class D Mute Ramp", RK817_CODEC_ACLASSD_CFG1, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 1", RK817_CODEC_ACLASSD_CFG1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 2", RK817_CODEC_ACLASSD_CFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 2", RK817_CODEC_ACLASSD_CFG2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 3", RK817_CODEC_ACLASSD_CFG2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 2", RK817_CODEC_ACLASSD_CFG2, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 3", RK817_CODEC_ACLASSD_CFG2, 0, 0, NULL, 0), + + /* playback path headphones */ + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", RK817_CODEC_AHP_CP, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone CP Discharge LDO", RK817_CODEC_AHP_CP, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone OStage", RK817_CODEC_AHP_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Pre Amp", RK817_CODEC_AHP_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", "Playback", RK817_CODEC_ADAC_CFG1, 1, 1), + SND_SOC_DAPM_DAC("DAC R", "Playback", RK817_CODEC_ADAC_CFG1, 0, 1), + + /* Mux for input/output path selection */ + SND_SOC_DAPM_MUX("Playback Mux", SND_SOC_NOPM, 1, 0, &dac_mux), + + /* Pins for Simple Card Bindings */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKO"), +}; + +static const struct snd_soc_dapm_route rk817_dapm_routes[] = { + + /* capture path */ + /* left mic */ + {"ADC L", NULL, "LDO Regulator"}, + {"ADC L", NULL, "IBIAS Block"}, + {"ADC L", NULL, "VAvg Buffer"}, + {"ADC L", NULL, "PLL Power"}, + {"ADC L", NULL, "ADC Clock"}, + {"ADC L", NULL, "I2S TX Clock"}, + {"ADC L", NULL, "ADC Channel Enable"}, + {"ADC L", NULL, "I2S TX Channel Enable"}, + {"ADC L", NULL, "I2S TX1 Transfer Start"}, + {"MICL", NULL, "MIC Power On"}, + {"MICL", NULL, "PGA L Power On"}, + {"MICL", NULL, "Mic Boost L1"}, + {"MICL", NULL, "Mic Boost L2"}, + {"MICL", NULL, "I2S TX3 Transfer Start"}, + {"MICL", NULL, "I2S TX3 Right Justified"}, + {"ADC L", NULL, "MICL"}, + + /* right mic */ + {"ADC R", NULL, "LDO Regulator"}, + {"ADC R", NULL, "IBIAS Block"}, + {"ADC R", NULL, "VAvg Buffer"}, + {"ADC R", NULL, "PLL Power"}, + {"ADC R", NULL, "ADC Clock"}, + {"ADC R", NULL, "I2S TX Clock"}, + {"ADC R", NULL, "ADC Channel Enable"}, + {"ADC R", NULL, "I2S TX Channel Enable"}, + {"ADC R", NULL, "I2S TX1 Transfer Start"}, + {"MICR", NULL, "MIC Power On"}, + {"MICR", NULL, "PGA R Power On"}, + {"MICR", NULL, "Mic Boost R1"}, + {"MICR", NULL, "Mic Boost R2"}, + {"MICR", NULL, "I2S TX3 Transfer Start"}, + {"MICR", NULL, "I2S TX3 Right Justified"}, + {"ADC R", NULL, "MICR"}, + + /* playback path */ + /* speaker path */ + {"SPK DAC", NULL, "LDO Regulator"}, + {"SPK DAC", NULL, "IBIAS Block"}, + {"SPK DAC", NULL, "VAvg Buffer"}, + {"SPK DAC", NULL, "PLL Power"}, + {"SPK DAC", NULL, "I2S TX1 Transfer Start"}, + {"SPK DAC", NULL, "DAC Clock"}, + {"SPK DAC", NULL, "I2S RX Clock"}, + {"SPK DAC", NULL, "DAC Channel Enable"}, + {"SPK DAC", NULL, "I2S RX Channel Enable"}, + {"SPK DAC", NULL, "Class D Mode"}, + {"SPK DAC", NULL, "DAC Bias"}, + {"SPK DAC", NULL, "DAC Mute Off"}, + {"SPK DAC", NULL, "Enable Class D"}, + {"SPK DAC", NULL, "Disable Class D Mute Ramp"}, + {"SPK DAC", NULL, "Class D Mute Rate 1"}, + {"SPK DAC", NULL, "Class D Mute Rate 2"}, + {"SPK DAC", NULL, "Class D OCPP 2"}, + {"SPK DAC", NULL, "Class D OCPP 3"}, + {"SPK DAC", NULL, "Class D OCPN 2"}, + {"SPK DAC", NULL, "Class D OCPN 3"}, + {"SPK DAC", NULL, "High Pass Filter"}, + + /* headphone path L */ + {"DAC L", NULL, "LDO Regulator"}, + {"DAC L", NULL, "IBIAS Block"}, + {"DAC L", NULL, "VAvg Buffer"}, + {"DAC L", NULL, "PLL Power"}, + {"DAC L", NULL, "I2S TX1 Transfer Start"}, + {"DAC L", NULL, "DAC Clock"}, + {"DAC L", NULL, "I2S RX Clock"}, + {"DAC L", NULL, "DAC Channel Enable"}, + {"DAC L", NULL, "I2S RX Channel Enable"}, + {"DAC L", NULL, "DAC Bias"}, + {"DAC L", NULL, "DAC Mute Off"}, + {"DAC L", NULL, "Headphone Charge Pump"}, + {"DAC L", NULL, "Headphone CP Discharge LDO"}, + {"DAC L", NULL, "Headphone OStage"}, + {"DAC L", NULL, "Headphone Pre Amp"}, + + /* headphone path R */ + {"DAC R", NULL, "LDO Regulator"}, + {"DAC R", NULL, "IBIAS Block"}, + {"DAC R", NULL, "VAvg Buffer"}, + {"DAC R", NULL, "PLL Power"}, + {"DAC R", NULL, "I2S TX1 Transfer Start"}, + {"DAC R", NULL, "DAC Clock"}, + {"DAC R", NULL, "I2S RX Clock"}, + {"DAC R", NULL, "DAC Channel Enable"}, + {"DAC R", NULL, "I2S RX Channel Enable"}, + {"DAC R", NULL, "DAC Bias"}, + {"DAC R", NULL, "DAC Mute Off"}, + {"DAC R", NULL, "Headphone Charge Pump"}, + {"DAC R", NULL, "Headphone CP Discharge LDO"}, + {"DAC R", NULL, "Headphone OStage"}, + {"DAC R", NULL, "Headphone Pre Amp"}, + + /* mux path for output selection */ + {"Playback Mux", "HP", "DAC L"}, + {"Playback Mux", "HP", "DAC R"}, + {"Playback Mux", "SPK", "SPK DAC"}, + {"SPKO", NULL, "Playback Mux"}, + {"HPOL", NULL, "Playback Mux"}, + {"HPOR", NULL, "Playback Mux"}, +}; + +static int rk817_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_init_regmap(component, rk817->regmap); + rk817->component = component; + + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + + rk817_init(component); + + /* setting initial pll values so that we can continue to leverage simple-audio-card. + * The values aren't important since no parameters are used. + */ + + snd_soc_component_set_pll(component, 0, 0, 0, 0); + + return 0; +} + +static void rk817_remove(struct snd_soc_component *component) +{ + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_volume_controls, + .num_controls = ARRAY_SIZE(rk817_volume_controls), + .dapm_routes = rk817_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk817_dapm_routes), + .dapm_widgets = rk817_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk817_dapm_widgets), + .set_pll = rk817_set_component_pll, +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static const struct regmap_config rk817_codec_regmap_config = { + .name = "rk817-codec", + .reg_bits = 8, + .val_bits = 8, + .reg_stride = 1, + .max_register = 0x4f, + .cache_type = REGCACHE_NONE, + .volatile_reg = rk817_volatile_register, + .writeable_reg = rk817_codec_register, + .readable_reg = rk817_codec_register, + .reg_defaults = rk817_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk817_reg_defaults), +}; + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk817 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk817) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->regmap = devm_regmap_init_i2c(rk817->i2c, + &rk817_codec_regmap_config); + if (IS_ERR(rk817_codec_data->regmap)) { + ret = PTR_ERR(rk817_codec_data->regmap); + dev_err(&pdev->dev, "failed to allocate register map: %d\n", + ret); + goto err_; + } + + rk817_codec_data->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = clk_prepare_enable(rk817_codec_data->mclk); + if (ret < 0) { + dev_err(&pdev->dev, "%s() clock prepare error %d\n", + __func__, ret); + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static int rk817_platform_remove(struct platform_device *pdev) +{ + struct rk817_codec_priv *rk817 = platform_get_drvdata(pdev); + + clk_disable_unprepare(rk817->mclk); + + return 0; +} + +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + }, + .probe = rk817_platform_probe, + .remove = rk817_platform_remove, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk817_codec.h b/sound/soc/codecs/rk817_codec.h new file mode 100644 index 000000000000..f3bc59d498b4 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.h @@ -0,0 +1,94 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * rk817 ALSA SoC Audio driver + * + Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + */ + +#ifndef __RK817_CODEC_H__ +#define __RK817_CODEC_H__ + +/* codec register */ +#define RK817_CODEC_BASE 0x0000 + +#define RK817_CODEC_DTOP_VUCTL (RK817_CODEC_BASE + 0x12) +#define RK817_CODEC_DTOP_VUCTIME (RK817_CODEC_BASE + 0x13) +#define RK817_CODEC_DTOP_LPT_SRST (RK817_CODEC_BASE + 0x14) +#define RK817_CODEC_DTOP_DIGEN_CLKE (RK817_CODEC_BASE + 0x15) +#define RK817_CODEC_AREF_RTCFG0 (RK817_CODEC_BASE + 0x16) +#define RK817_CODEC_AREF_RTCFG1 (RK817_CODEC_BASE + 0x17) +#define RK817_CODEC_AADC_CFG0 (RK817_CODEC_BASE + 0x18) +#define RK817_CODEC_AADC_CFG1 (RK817_CODEC_BASE + 0x19) +#define RK817_CODEC_DADC_VOLL (RK817_CODEC_BASE + 0x1a) +#define RK817_CODEC_DADC_VOLR (RK817_CODEC_BASE + 0x1b) +#define RK817_CODEC_DADC_SR_ACL0 (RK817_CODEC_BASE + 0x1e) +#define RK817_CODEC_DADC_ALC1 (RK817_CODEC_BASE + 0x1f) +#define RK817_CODEC_DADC_ALC2 (RK817_CODEC_BASE + 0x20) +#define RK817_CODEC_DADC_NG (RK817_CODEC_BASE + 0x21) +#define RK817_CODEC_DADC_HPF (RK817_CODEC_BASE + 0x22) +#define RK817_CODEC_DADC_RVOLL (RK817_CODEC_BASE + 0x23) +#define RK817_CODEC_DADC_RVOLR (RK817_CODEC_BASE + 0x24) +#define RK817_CODEC_AMIC_CFG0 (RK817_CODEC_BASE + 0x27) +#define RK817_CODEC_AMIC_CFG1 (RK817_CODEC_BASE + 0x28) +#define RK817_CODEC_DMIC_PGA_GAIN (RK817_CODEC_BASE + 0x29) +#define RK817_CODEC_DMIC_LMT1 (RK817_CODEC_BASE + 0x2a) +#define RK817_CODEC_DMIC_LMT2 (RK817_CODEC_BASE + 0x2b) +#define RK817_CODEC_DMIC_NG1 (RK817_CODEC_BASE + 0x2c) +#define RK817_CODEC_DMIC_NG2 (RK817_CODEC_BASE + 0x2d) +#define RK817_CODEC_ADAC_CFG0 (RK817_CODEC_BASE + 0x2e) +#define RK817_CODEC_ADAC_CFG1 (RK817_CODEC_BASE + 0x2f) +#define RK817_CODEC_DDAC_POPD_DACST (RK817_CODEC_BASE + 0x30) +#define RK817_CODEC_DDAC_VOLL (RK817_CODEC_BASE + 0x31) +#define RK817_CODEC_DDAC_VOLR (RK817_CODEC_BASE + 0x32) +#define RK817_CODEC_DDAC_SR_LMT0 (RK817_CODEC_BASE + 0x35) +#define RK817_CODEC_DDAC_LMT1 (RK817_CODEC_BASE + 0x36) +#define RK817_CODEC_DDAC_LMT2 (RK817_CODEC_BASE + 0x37) +#define RK817_CODEC_DDAC_MUTE_MIXCTL (RK817_CODEC_BASE + 0x38) +#define RK817_CODEC_DDAC_RVOLL (RK817_CODEC_BASE + 0x39) +#define RK817_CODEC_DDAC_RVOLR (RK817_CODEC_BASE + 0x3a) +#define RK817_CODEC_AHP_ANTI0 (RK817_CODEC_BASE + 0x3b) +#define RK817_CODEC_AHP_ANTI1 (RK817_CODEC_BASE + 0x3c) +#define RK817_CODEC_AHP_CFG0 (RK817_CODEC_BASE + 0x3d) +#define RK817_CODEC_AHP_CFG1 (RK817_CODEC_BASE + 0x3e) +#define RK817_CODEC_AHP_CP (RK817_CODEC_BASE + 0x3f) +#define RK817_CODEC_ACLASSD_CFG1 (RK817_CODEC_BASE + 0x40) +#define RK817_CODEC_ACLASSD_CFG2 (RK817_CODEC_BASE + 0x41) +#define RK817_CODEC_APLL_CFG0 (RK817_CODEC_BASE + 0x42) +#define RK817_CODEC_APLL_CFG1 (RK817_CODEC_BASE + 0x43) +#define RK817_CODEC_APLL_CFG2 (RK817_CODEC_BASE + 0x44) +#define RK817_CODEC_APLL_CFG3 (RK817_CODEC_BASE + 0x45) +#define RK817_CODEC_APLL_CFG4 (RK817_CODEC_BASE + 0x46) +#define RK817_CODEC_APLL_CFG5 (RK817_CODEC_BASE + 0x47) +#define RK817_CODEC_DI2S_CKM (RK817_CODEC_BASE + 0x48) +#define RK817_CODEC_DI2S_RSD (RK817_CODEC_BASE + 0x49) +#define RK817_CODEC_DI2S_RXCR1 (RK817_CODEC_BASE + 0x4a) +#define RK817_CODEC_DI2S_RXCR2 (RK817_CODEC_BASE + 0x4b) +#define RK817_CODEC_DI2S_RXCMD_TSD (RK817_CODEC_BASE + 0x4c) +#define RK817_CODEC_DI2S_TXCR1 (RK817_CODEC_BASE + 0x4d) +#define RK817_CODEC_DI2S_TXCR2 (RK817_CODEC_BASE + 0x4e) +#define RK817_CODEC_DI2S_TXCR3_TXCMD (RK817_CODEC_BASE + 0x4f) + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + +#endif /* __RK817_CODEC_H__ */ -- 2.25.1

2 7

[PATCH 1/2] ASoC: rt715: add main capture switch and main capture volume
by Jack Yu 19 Mar '21

19 Mar '21

Add main capture switch and main capture volume control. Main capture control has its own channel value respectivelly. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 392 ++++++++++++++++++++++++++++++++------- 1 file changed, 323 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index f419d001a3fb..c461a4c23fd3 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -80,85 +80,102 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; - unsigned int read_ll, read_rl; - int i; - - /* Can't use update bit function, so read the original value first */ - addr_h = mc->reg; - addr_l = mc->rreg; - if (mc->shift == RT715_DIR_OUT_SFT) /* output */ - val_h = 0x80; - else /* input */ - val_h = 0x0; - - rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); - - /* L Channel */ - if (mc->invert) { - /* for mute */ - val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; - /* keep gain */ - read_ll = read_ll & 0x7f; - val_ll |= read_ll; - } else { - /* for gain */ - val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); - if (val_ll > mc->max) - val_ll = mc->max; - /* keep mute status */ - read_ll = read_ll & 0x80; - val_ll |= read_ll; - } - - /* R Channel */ - if (mc->invert) { - regmap_write(rt715->regmap, - RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); - /* for mute */ - val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; - /* keep gain */ - read_rl = read_rl & 0x7f; - val_lr |= read_rl; - } else { - /* for gain */ - val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); - if (val_lr > mc->max) - val_lr = mc->max; - /* keep mute status */ - read_rl = read_rl & 0x80; - val_lr |= read_rl; - } + unsigned int read_ll, read_rl, i, j, loop_cnt; - for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (strstr(ucontrol->id.name, "Main Capture Switch") || + strstr(ucontrol->id.name, "Main Capture Volume")) + loop_cnt = 4; + else + loop_cnt = 1; - if (val_ll == val_lr) { - /* Set both L/R channels at the same time */ - val_h = (1 << mc->shift) | (3 << 4); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_ll)); + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + if (loop_cnt == 1) { + addr_h = mc->reg; + addr_l = mc->rreg; } else { - /* Lch*/ - val_h = (1 << mc->shift) | (1 << 5); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - /* Rch */ - val_h = (1 << mc->shift) | (1 << 4); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_lr)); + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; } - /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ val_h = 0x80; else /* input */ val_h = 0x0; - rt715_get_gain(rt715, addr_h, addr_l, val_h, - &read_rl, &read_ll); - if (read_rl == val_lr && read_ll == val_ll) - break; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + } + + /* R Channel */ + if (mc->invert) { + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } } /* D0:power on state, D3: power saving mode */ if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) @@ -201,9 +218,227 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt715_set_main_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + unsigned int k_shift = RT715_DIR_IN_SFT; + + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for mute */ + val_ll = (!ucontrol->value.integer.value[j * 2]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + + /* R Channel */ + /* for mute */ + val_lr = (!ucontrol->value.integer.value[j * 2 + 1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_main_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = !(read_ll & 0x80); + ucontrol->value.integer.value[i * 2 + 1] = !(read_rl & 0x80); + } + + return 0; +} + +static int rt715_set_main_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + unsigned int k_shift = RT715_DIR_IN_SFT, k_max = 0x3f; + + for (j = 0; j < loop_cnt; j++) { + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for gain */ + val_ll = ((ucontrol->value.integer.value[j * 2]) & 0x7f); + if (val_ll > k_max) + val_ll = k_max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + + /* R Channel */ + /* for gain */ + val_lr = ((ucontrol->value.integer.value[j * 2 + 1]) & 0x7f); + if (val_lr > k_max) + val_lr = k_max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_main_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = read_ll & 0x7f; + ucontrol->value.integer.value[i * 2 + 1] = read_rl & 0x7f; + } + + return 0; +} + static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); +static int rt715_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int rt715_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x3f; + return 0; +} + #define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -212,6 +447,21 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define RT715_MAIN_SWITCH_EXT(xname, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt715_switch_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + +#define RT715_MAIN_VOL_EXT_TLV(xname, xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt715_vol_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, @@ -226,6 +476,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, rt715_set_amp_gain_get, rt715_set_amp_gain_put), + RT715_MAIN_SWITCH_EXT("Main Capture Switch", + rt715_set_main_switch_get, rt715_set_main_switch_put), /* Volume Control */ SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, @@ -243,6 +495,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, in_vol_tlv), + RT715_MAIN_VOL_EXT_TLV("Main Capture Volume", + rt715_set_main_vol_get, rt715_set_main_vol_put, in_vol_tlv), /* MIC Boost Control */ SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, -- 2.29.0

3 3

aplay: avoid any further PCM writing if in abort
by GitHub pull_request - opened 19 Mar '21

19 Mar '21

alsa-project/alsa-utils pull request #81 was opened from aditpape: Terminating stream with CTRL-C will set in_aborting flag which is used to leave any write/read loop on the ALSA device. After ending the read/write loop aplay tries to drain the stream which is not required and can also lead to malfunctions: -If user interrupts a blocked/non responsive PCM (e.g. usb uac2 gadget which does not consume data due to stream stopped by host) it will successfully terminate the write loop but will hang again in drain call. This would require to hit CTRL-C again to unblock which should be avoided. Aplay currently anyhow allows signal handler to get invoked only once. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/81 Patch URL : https://github.com/alsa-project/alsa-utils/pull/81.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda: Fix spelling mistakes
by huangjianghui 19 Mar '21

19 Mar '21

Signed-off-by: huangjianghui <huangjianghui(a)uniontech.com> --- sound/hda/hdac_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index a6ed3dc35f7e..1eb8563db2df 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -618,7 +618,7 @@ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger); /** - * snd_hdac_stream_sync - sync with start/strop trigger operation + * snd_hdac_stream_sync - sync with start/stop trigger operation * @azx_dev: HD-audio core stream (master stream) * @start: true = start, false = stop * @streams: bit flags of streams to sync -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: add more delay time before determine_headset_type
by Hui Wang 19 Mar '21

19 Mar '21

We found a recording issue on the headset-mic recently, sometimes users plug a headset and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. In the past, determine_headset_type() worked well because the internal mic is connected to the codec, so the "Input Source" or "Capture Source" is internal mic by default when users plug a headset, the determine_headset_type() will not be called unless users select headset-mic from UI, when users select headset-mic, the plugging action already finished and the headset is completely plugged into the jack, so determine_headset_type() could return a correct type. But more and more machines connect the internal mic to the PCH now, and the "Input Source" is headset mic by default, when users plug a headset, the determine_headset_type() will be called immediately, if the headset is not completely plugged in, it will return a wrong type. Here add 2s delay before calling determine_headset_type(), and since there is a pop-up dialogue when users do plugging action, to avoid freezing the UI, use the deleyed_work to call that function. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++------- 1 file changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b47504fa8dfd..1f6fc8addf3e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -127,6 +127,8 @@ struct alc_spec { unsigned int coef0; struct input_dev *kb_dev; u8 alc_mute_keycode_map[1]; + struct hda_codec *codec; + struct delayed_work headset_set_work; }; /* @@ -1160,6 +1162,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) kfree(spec); return err; } + spec->codec = codec; return 0; } @@ -5363,6 +5366,21 @@ static void alc_determine_headset_type(struct hda_codec *codec) spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; } +static void alc_headset_check_and_set(struct work_struct *work) +{ + struct alc_spec *spec = container_of(work, struct alc_spec, + headset_set_work.work); + struct hda_codec *codec = spec->codec; + + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; +} + static void alc_update_headset_mode(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5386,6 +5404,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) return; } + cancel_delayed_work_sync(&spec->headset_set_work); switch (new_headset_mode) { case ALC_HEADSET_MODE_UNPLUGGED: alc_headset_mode_unplugged(codec); @@ -5394,13 +5413,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) spec->gen.hp_jack_present = false; break; case ALC_HEADSET_MODE_HEADSET: - if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) - alc_determine_headset_type(codec); - if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) - alc_headset_mode_ctia(codec); - else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) - alc_headset_mode_omtp(codec); - spec->gen.hp_jack_present = true; + schedule_delayed_work(&spec->headset_set_work, msecs_to_jiffies(2000)); break; case ALC_HEADSET_MODE_MIC: alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); @@ -5466,6 +5479,7 @@ static void alc_fixup_headset_mode(struct hda_codec *codec, spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; break; case HDA_FIXUP_ACT_PROBE: + INIT_DELAYED_WORK(&spec->headset_set_work, alc_headset_check_and_set); alc_probe_headset_mode(codec); break; case HDA_FIXUP_ACT_INIT: @@ -5475,6 +5489,9 @@ static void alc_fixup_headset_mode(struct hda_codec *codec, } alc_update_headset_mode(codec); break; + case HDA_FIXUP_ACT_FREE: + cancel_delayed_work_sync(&spec->headset_set_work); + break; } } -- 2.25.1

2 3

Very unstable and broken after audio device switching, reboots, multimedia app usage etc etc etc
by GitHub issues - edited 19 Mar '21

19 Mar '21

alsa-project/alsa-lib issue #127 was edited from oliiix: To me this seems to be a huge issue for years now. Running Ubuntu 20.10 very latest. I'm using an Astro A50 and two HDMI Displays for playing audio plus there's two more outputs I'm not really using. But as soon as the computer ran a little long (yes, that's actually enough to cause problems, probably due to certain apps like Unreal engine and games!), came out of sleep or I switch playback devices (which I'm doing very often and that shouldnt be ANY issue), things get seriously mixed up. Suddenly the routing goes all wrong (i select one output device, another plays instead, or just no sound at all). Sometimes "no sound" problem gets fixed by just switching devices about 10-20! times until sound is back (it either works or doesn't but that it starts working again after ten times switching devices is very weird behavior). Reinstalling as well as resetting alsa doesn't do anything to remedy no sound, restarting computer doesn't fix sound problems, only switching devices ten times does! Alsa Mixer doesn't show any controls for my A50, yet it often works even if some profiles are to low volume and the one that has good volume there I can't hear audio that comes from behind me (I have the no sound issue also on my HDMI outputs, just saying). Pavucontrol on the other hand lists my A50 but doesn't show one of my HDMI displays. I mean what is going wrong here (for years)??? I mean I found forum posts about these issues that reach back years in time -.- When will we have proper working audio where audio device switching on linux is possible without any problems and the sound doesn't just "give up" after some time? maybe someone should create one all including audio software system structure instead of several packages that are unable to work together properly... can't be that two apps (pavucontrol and pulseaudio) that work on top on alsa show different devices and settings to begin with Issue URL : https://github.com/alsa-project/alsa-lib/issues/127 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Very unstable and broken after audio device switching, reboots, multimedia app usage etc etc etc
by GitHub issues - opened 19 Mar '21

19 Mar '21

alsa-project/alsa-lib issue #127 was opened from oliiix: To me this seems to be a huge issue for years now. Running Ubuntu 20.10 very latest. I'm using an Astro A50 and two HDMI Displays for playing audio plus there's two more outputs I'm not really using. But as soon as the computer ran a little long (yes, that's actually enough to cause problems, probably due to certain apps like Unreal engine and games!), came out of sleep or I switch playback devices (which I'm doing very often and that shouldnt be ANY issue), things get seriously mixed up. Suddenly the routing goes all wrong (i select one output device, another plays instead, or just no sound at all). Sometimes "no sound" problem gets fixed by just switching devices about 10-20! times until sound is back (it either works or doesn't but that it starts working again after ten times switching devices is very weird behavior). Reinstalling as well as resetting alsa doesn't do anything to remedy no sound, restarting computer doesn't fix sound problems, only switching devices ten times does! Alsa Mixer doesn't show any controls for my A50, yet it often works even if some profiles are to low volume and the one that has good volume there I can't hear audio that comes from behind me (I have the no sound issue also on my HDMI outputs, just saying). Pavucontrol on the other hand lists my A50 but doesn't show one of my HDMI displays. I mean what is going wrong here (for years)??? I mean I found forum posts about these issues that reach back years in time -.- When will we have proper working audio where audio device switching on linux is possible without any problems and the sound doesn't just "give up" after some time? Issue URL : https://github.com/alsa-project/alsa-lib/issues/127 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: fsl_sai: remove reset code from dai_probe
by Shengjiu Wang 19 Mar '21

19 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> SAI software reset is done in runtime resume, there is no need to do it in fsl_sai_dai_probe. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 6ef2ce348232..9e7893f91882 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -711,13 +711,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); unsigned int ofs = sai->soc_data->reg_offset; - /* Software Reset for both Tx and Rx */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); - /* Clear SR bit to finish the reset */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); - regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs), FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX); -- 2.27.0

4 5

[PATCH] docs: devicetree: bindings: sound: Fix a typo in the file rt5682.txt
by Bhaskar Chowdhury 18 Mar '21

18 Mar '21

s/drving/driving/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 9c5fadb6ac82..cd8c53d8497e 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -44,7 +44,7 @@ Optional properties: - realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of the particular DMIC. -- realtek,dmic-clk-driving-high : Set the high drving of the DMIC clock out. +- realtek,dmic-clk-driving-high : Set the high driving of the DMIC clock out. Pins on the device (for linking into audio routes) for RT5682: -- 2.26.2

3 3

[PATCH v3] ASoC: Intel: sof_rt5682: Add ALC1015Q-VB speaker amp support
by Brent Lu 18 Mar '21

18 Mar '21

This patch adds jsl_rt5682_rt1015p which supports the RT5682 headset codec and ALC1015Q-VB speaker amplifier combination on JasperLake platform. This driver also supports ALC1015Q-CG if running in auto-mode. Following table shows the audio interface support of the two amplifiers. | ALC1015Q-CG | ALC1015Q-VB ===================================== I2C | Yes | No Auto-mode | 48K, 64fs | 16k, 32fs | 48k, 32fs | 48k, 64fs Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_realtek_common.c | 105 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 19 +++- .../intel/common/soc-acpi-intel-jsl-match.c | 13 +++ 5 files changed, 143 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index d1d28129a32b..58379393b8e4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -457,6 +457,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_MAX98373_I2C select SND_SOC_RT1011 select SND_SOC_RT1015 + select SND_SOC_RT1015P select SND_SOC_RT5682_I2C select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index f3cf73c620ba..2ec34f8df9e1 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -7,6 +7,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -136,3 +137,107 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1011_codec_confs; card->num_configs = ARRAY_SIZE(rt1011_codec_confs); } + +/* + * rt1015: i2c mode driver for ALC1015 and ALC1015Q + * rt1015p: auto-mode driver for ALC1015, ALC1015Q, and ALC1015Q-VB + * + * For stereo output, there are always two amplifiers on the board. + * However, the ACPI implements only one device instance (UID=0) if they + * are sharing the same enable pin. The code will detect the number of + * device instance and use corresponding DAPM structures for + * initialization. + */ +static const struct snd_soc_dapm_route rt1015p_1dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Speaker" }, + { "Right Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route rt1015p_2dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left Speaker" }, + { "Right Spk", NULL, "Right Speaker" }, +}; + +static struct snd_soc_codec_conf rt1015p_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = { + { + .name = RT1015P_DEV0_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, + { + .name = RT1015P_DEV1_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, +}; + +static int rt1015p_get_num_codecs(void) +{ + static int dev_num; + + if (dev_num) + return dev_num; + + if (!acpi_dev_present("RTL1015", "1", -1)) + dev_num = 1; + else + dev_num = 2; + + return dev_num; +} + +static int rt1015p_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* reserved for debugging purpose */ + + return 0; +} + +static const struct snd_soc_ops rt1015p_ops = { + .hw_params = rt1015p_hw_params, +}; + +static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + if (rt1015p_get_num_codecs() == 1) + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes, + ARRAY_SIZE(rt1015p_1dev_dapm_routes)); + else + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_2dev_dapm_routes, + ARRAY_SIZE(rt1015p_2dev_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1015p_dai_link_components; + link->num_codecs = rt1015p_get_num_codecs(); + link->init = rt1015p_init; + link->ops = &rt1015p_ops; +} + +void sof_rt1015p_codec_conf(struct snd_soc_card *card) +{ + if (rt1015p_get_num_codecs() == 1) + return; + + card->codec_conf = rt1015p_codec_confs; + card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 87cb3812b926..cb0b49b2855c 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -21,4 +21,11 @@ void sof_rt1011_dai_link(struct snd_soc_dai_link *link); void sof_rt1011_codec_conf(struct snd_soc_card *card); +#define RT1015P_CODEC_DAI "HiFi" +#define RT1015P_DEV0_NAME "RTL1015:00" +#define RT1015P_DEV1_NAME "RTL1015:01" + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); +void sof_rt1015p_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 55505e207bc0..f4b898c1719f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -45,8 +45,9 @@ #define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) #define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) #define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -723,6 +724,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(rt1015_components); links[id].init = speaker_codec_init_lr; links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; @@ -851,6 +854,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_max98373_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) sof_rt1011_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) + sof_rt1015p_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -940,6 +945,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015p", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; @@ -966,3 +980,4 @@ MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); MODULE_ALIAS("platform:tgl_rt1011_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015p"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 52238db0bcb5..73fe4f89a82d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs rt1015_spk = { .codecs = {"10EC1015"} }; +static struct snd_soc_acpi_codecs rt1015p_spk = { + .num_codecs = 1, + .codecs = {"RTL1015"} +}; + static struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &rt1015_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015p", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10EC5682", .drv_name = "jsl_rt5682_max98360a", -- 2.17.1

6 7

[PATCH 00/23] ASoC: codecs: remove cppcheck warnings
by Pierre-Louis Bossart 18 Mar '21

18 Mar '21

Lots of small fixes in various codec drivers that should have no functional impact. Pierre-Louis Bossart (23): ASoC: ab8500-codec: remove useless structure ASoC: ad1836: remove useless return ASoC: adau1977: remove useless return ASoC: cros_ec_codec: remove null pointer dereference warning ASoC: cx2070x: remove useless assignment ASoC: cx2070x: remove duplicate else branch ASoC: da7219-aad: remove useless initialization ASoC: hdac_hdmi: remove useless initializations ASoC: hdac_hdmi: align function arguments ASoC: hdmi-codec: remove useless initialization ASoC: hdmi-codec: remove unused spk_mask member ASoC: max98090: remove useless assignment ASoC: mt6358: remove useless initializations ASoC: mt6359: remove useless assignment ASoC: nau8825: remove useless assignment ASoC: pcm1681: remove useless assignment ASoC: sigmadsp: align function prototype ASoC: sti-sas: remove unused struct members ASoC: tas2562: remove useless assignment ASoC: tas2562: remove warning on return value ASoC: tas2770: remove useless initialization ASoC: tlv320dac33: clarify expression ASoC: tscs454: remove useless test on PLL disable sound/soc/codecs/ab8500-codec.c | 7 ------- sound/soc/codecs/ad1836.c | 2 -- sound/soc/codecs/adau1977.c | 2 -- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/cx2072x.c | 11 ++++------- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 14 +++++++------- sound/soc/codecs/hdac_hdmi.h | 2 +- sound/soc/codecs/hdmi-codec.c | 3 +-- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mt6358.c | 4 ++-- sound/soc/codecs/mt6359.c | 4 ++-- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/sigmadsp.h | 2 +- sound/soc/codecs/sti-sas.c | 3 --- sound/soc/codecs/tas2562.c | 3 +-- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tscs454.c | 7 ++++++- 20 files changed, 32 insertions(+), 46 deletions(-) -- 2.25.1

6 30

[PATCH] ASoC: Intel: Skylake: skl-topology: fix -frame-larger-than
by Nick Desaulniers 18 Mar '21

18 Mar '21

Fixes: sound/soc/intel/skylake/skl-topology.c:3613:13: warning: stack frame size of 1304 bytes in function 'skl_tplg_complete' [-Wframe-larger-than=] struct snd_ctl_elem_value is 1224 bytes in my configuration. Heap allocate it, then free it within the current frame. Signed-off-by: Nick Desaulniers <nick.desaulniers(a)gmail.com> --- sound/soc/intel/skylake/skl-topology.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd59292..cdc916c91301 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3613,10 +3613,15 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, static void skl_tplg_complete(struct snd_soc_component *component) { struct snd_soc_dobj *dobj; - struct snd_soc_acpi_mach *mach = - dev_get_platdata(component->card->dev); + struct snd_soc_acpi_mach *mach; + struct snd_ctl_elem_value *val; int i; + val = kzalloc(sizeof(*val), GFP_KERNEL); + if (!val) + return; + + mach = dev_get_platdata(component->card->dev); list_for_each_entry(dobj, &component->dobj_list, list) { struct snd_kcontrol *kcontrol = dobj->control.kcontrol; struct soc_enum *se = @@ -3631,14 +3636,13 @@ static void skl_tplg_complete(struct snd_soc_component *component) sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; - if (strstr(texts[i], chan_text)) { - val.value.enumerated.item[0] = i; - kcontrol->put(kcontrol, &val); + val->value.enumerated.item[0] = i; + kcontrol->put(kcontrol, val); } } } + kfree(val); } static struct snd_soc_tplg_ops skl_tplg_ops = { base-commit: 65f0d2414b7079556fbbcc070b3d1c9f9587606d prerequisite-patch-id: 4d05aad8c2b50c0c3b4447dd842abe8b1b840927 -- 2.25.1

3 5

[PATCH v5 00/17] Fix reset controls and RPM of NVIDIA Tegra ASoC drivers
by Dmitry Osipenko 18 Mar '21

18 Mar '21

Hi, This series adds missing hardware reset controls to I2S and AC97 drivers, corrects runtime PM usage and drivers probe/remove order. Currently drivers happen to work properly because reset is implicitly deasserted by tegra-clk driver, but clk driver shouldn't touch the resets and we need to fix it because this breaks other Tegra drivers. Previously we fixed the resets of the AHUB and HDMI codec drivers, but turned out that we missed the I2C and AC97 drivers. Thanks to Paul Fertser for testing the pending clk patches and finding that audio got broken on Tegra20 AC100 netbook because of the missing I2S reset. Changelog: v5: - After taking another look at the drivers I noticed couple more things that could be improved. These new patches correct runtime PM and probe/remove order of the drivers: ASoC: tegra20: spdif: Correct driver removal order ASoC: tegra20: spdif: Remove handing of disabled runtime PM ASoC: tegra20: i2s: Add system level suspend-resume callbacks ASoC: tegra20: i2s: Correct driver removal order ASoC: tegra20: i2s: Use devm_clk_get() ASoC: tegra20: i2s: Remove handing of disabled runtime PM ASoC: tegra30: i2s: Correct driver removal order ASoC: tegra30: i2s: Use devm_clk_get() ASoC: tegra30: i2s: Remove handing of disabled runtime PM ASoC: tegra30: ahub: Reset global variable ASoC: tegra30: ahub: Correct suspend-resume callbacks ASoC: tegra30: ahub: Remove handing of disabled runtime PM v4: - Added missing prototype for reset_control_bulk_put(). v3: - Fixed reset stubs for !CONFIG_RESET_CONTROLLER. v2: - After some more testing I found that I2S control logic doesn't require I2S clock to be enabled for resetting. Hence it's fine to have I2S to be reset by parent AHUB driver, so I dropped "tegra30: i2s: Add reset control" patch. - While I was double-checking resets on Tegra30, I found that that Tegra30 I2S driver has a broken runtime PM which doesn't restore hardware state on resume and it's lost after AHUB RPM-resume. Thus, added this new patch "tegra30: i2s: Restore hardware state on runtime PM resume". - Added new patches which switch AHUB driver to use reset-bulk API. I took the RFC patch from Philipp Zabel, fixed it and added devm_reset_control_bulk_optional_get_exclusive_released() that will be useful for further Tegra GPU patches. This is a minor improvement which makes code cleaner. Dmitry Osipenko (16): ASoC: tegra20: ac97: Add reset control ASoC: tegra20: i2s: Add reset control ASoC: tegra30: i2s: Restore hardware state on runtime PM resume ASoC: tegra30: ahub: Switch to use reset-bulk API ASoC: tegra20: spdif: Correct driver removal order ASoC: tegra20: spdif: Remove handing of disabled runtime PM ASoC: tegra20: i2s: Add system level suspend-resume callbacks ASoC: tegra20: i2s: Correct driver removal order ASoC: tegra20: i2s: Use devm_clk_get() ASoC: tegra20: i2s: Remove handing of disabled runtime PM ASoC: tegra30: i2s: Correct driver removal order ASoC: tegra30: i2s: Use devm_clk_get() ASoC: tegra30: i2s: Remove handing of disabled runtime PM ASoC: tegra30: ahub: Reset global variable ASoC: tegra30: ahub: Correct suspend-resume callbacks ASoC: tegra30: ahub: Remove handing of disabled runtime PM Philipp Zabel (1): reset: Add reset_control_bulk API drivers/reset/core.c | 215 ++++++++++++++++++++++ include/linux/reset.h | 315 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 21 +++ sound/soc/tegra/tegra20_ac97.h | 1 + sound/soc/tegra/tegra20_i2s.c | 60 +++--- sound/soc/tegra/tegra20_i2s.h | 1 + sound/soc/tegra/tegra20_spdif.c | 16 +- sound/soc/tegra/tegra30_ahub.c | 168 ++++++----------- sound/soc/tegra/tegra30_ahub.h | 5 +- sound/soc/tegra/tegra30_i2s.c | 65 ++----- 10 files changed, 667 insertions(+), 200 deletions(-) -- 2.30.2

2 21

[PATCH alsa-lib] conf: Fix invalid free at parse_args()
by Takashi Iwai 18 Mar '21

18 Mar '21

The previous fix for memory leaks introduced a few regression. The major one is the assert hit in the error path reaching with NULL or uninitialized sub object. Also, in other code paths, it's possible that an already released sub object gets freed again. Fix those bugs by initializing the sub object properly and add a NULL check before calling snd_config_delete(). Fixes: ad5f255b4767 ("conf: fix memory leak on the error path in parse_args()") Reported-and-tested-by: Mark Hills <mark(a)xwax.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/src/conf.c b/src/conf.c index 14b14b597e16..1bcd65c87b94 100644 --- a/src/conf.c +++ b/src/conf.c @@ -5080,6 +5080,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) const char *new = str; const char *tmp; char *val = NULL; + + sub = NULL; err = parse_arg(&new, &varlen, &val); if (err < 0) goto _err; @@ -5104,6 +5106,7 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) err = snd_config_search(subs, var, &sub); if (err >= 0) snd_config_delete(sub); + sub = NULL; err = snd_config_search(def, "type", &typ); if (err < 0) { _invalid_type: @@ -5169,7 +5172,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) err = snd_config_add(subs, sub); if (err < 0) { _err: - snd_config_delete(sub); + if (sub) + snd_config_delete(sub); free(val); return err; } -- 2.26.2

1 0

ASoC topology loading vs card bind
by Srinivas Kandagatla 18 Mar '21

18 Mar '21

Hi All, Am trying to load a FrontEnd/pcm dai definition from ASoC topology however I hit a catch 22 situation here. Topology is loaded as part of component probe() but component probe is only called as part of sound-card bind(). Any pointers on how is this supposed to work? Thanks, srini

4 6

[PATCH 0/3] Fix SND_HDA_PREALLOC issue
by Amadeusz Sławiński 18 Mar '21

18 Mar '21

For context it started with user reporting failures when running arecord without any error or warning in dmesg (after fixing some configuration problems thet they had). https://bugzilla.kernel.org/show_bug.cgi?id=201251#c279 After spending time investigating the issue it was narrowed to quite big setting of CONFIG_SND_HDA_PREALLOC_SIZE (4096). When looking at code https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… there is a limit of memory per card: max_alloc_per_card = 32UL * 1024UL * 1024UL When SND_HDA_PREALLOC_SIZE is set to 4096 it only has memory to alloc for 8 frontends, while Skylake HDA card has 10 of them (6 playback and 4 capture), so preallocated memory is exhausted while probing. In consequence 2 of FEs end without allocated memory. It can be workarounded on user side with setting SND_HDA_PREALLOC_SIZE to lower value, other is changing memory limit per card. However in order to not waste user memory, change maximum allocation size on HDA controller to 4MB and force automatical memory allocation insted of preallocated one. First patch adds prints, so similar issues can be easily identified in the future. Second changes maximum size of hda buffer to reasonable value of 4MB. And last one reverts patch which allowed setting prealloc size on X86 platforms. Amadeusz Sławiński (3): ALSA: pcm: Add debug print on memory allocation failure ALSA: hda: Change AZX_MAX_BUF_SIZE from 1GB to 4MB ALSA: hda: Revert "ALSA: hda: Allow setting preallocation again for x86" include/sound/hda_register.h | 4 ++-- sound/core/pcm_memory.c | 8 ++++++++ sound/hda/Kconfig | 7 +++---- 3 files changed, 13 insertions(+), 6 deletions(-) -- 2.25.1

3 7

[PATCH] ALSA: usb-audio: Fix unintentional sign extension issue
by Colin King 18 Mar '21

18 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> The shifting of the u8 integer device by 24 bits to the left will be promoted to a 32 bit signed int and then sign-extended to a 64 bit unsigned long. In the event that the top bit of device is set then all then all the upper 32 bits of the unsigned long will end up as also being set because of the sign-extension. Fix this by casting device to an unsigned long before the shift. Addresses-Coverity: ("Unintended sign extension") Fixes: a07df82c7990 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 5b77e8140ba6..f13cdfd8bd20 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2926,7 +2926,7 @@ static int snd_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_v u8 group = (private_value & SND_DJM_GROUP_MASK) >> SND_DJM_GROUP_SHIFT; u16 value = elem->value.enumerated.item[0]; - kctl->private_value = ((device << SND_DJM_DEVICE_SHIFT) | + kctl->private_value = (((unsigned long)device << SND_DJM_DEVICE_SHIFT) | (group << SND_DJM_GROUP_SHIFT) | value); @@ -2964,7 +2964,7 @@ static int snd_djm_controls_create(struct usb_mixer_interface *mixer, value = device->controls[i].default_value; knew.name = device->controls[i].name; knew.private_value = ( - (device_idx << SND_DJM_DEVICE_SHIFT) | + ((unsigned long)device_idx << SND_DJM_DEVICE_SHIFT) | (i << SND_DJM_GROUP_SHIFT) | value); err = snd_djm_controls_update(mixer, device_idx, i, value); -- 2.30.2

2 1

[GIT PULL] ASoC fixes for v5.12-rc2
by Mark Brown 18 Mar '21

18 Mar '21

The following changes since commit a38fd8748464831584a19438cbb3082b5a2dab15: Linux 5.12-rc2 (2021-03-05 17:33:41 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.12-rc2 for you to fetch changes up to f9dc51cc6676e0360dfcb1836b3d426a3ccf3093: Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar(a)nvidia.com>: (2021-03-16 17:55:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.12 Quite a lot of mostly platform specific fixes here, the only one which is generic is a fix for regressions on devices with more complex clocking support with simple-card. There's also a few new device IDs and platform quirks. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: fsl_ssi: Fix TDM slot setup for I2S mode Bard Liao (1): ASoC: rt711: add snd_soc_component remove callback Benjamin Rood (1): ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe Colin Ian King (1): ASoC: codecs: lpass-rx-macro: Fix uninitialized variable ec_tx Hans de Goede (8): ASoC: rt5670: Remove 'OUT Channel Switch' control ASoC: rt5670: Remove 'HP Playback Switch' control ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer settings ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: es8316: Simplify adc_pga_gain_tlv table Jack Yu (2): ASoC: rt1015: fix i2c communication error ASoC: rt1015: enable BCLK detection after calibration Jiaxin Yu (1): ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge Jon Hunter (1): ASoC: soc-core: Prevent warning if no DMI table is present Jonathan Marek (2): ASoC: codecs: lpass-va-macro: mute/unmute all active decimators ASoC: codecs: lpass-wsa-macro: fix RX MIX input controls Lucas Tanure (5): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset Mark Brown (5): Merge series "ASoC: rt5670: Various kcontrol fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "AsoC: rt5640/rt5651: Volume control fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal(a)opensource.cirrus.com>: Merge series "ASoC: sdm845: array out of bound issues" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar(a)nvidia.com>: Meng Li (1): spi: cadence: set cqspi to the driver_data field of struct device Pan Xiuli (1): ASoC: SOF: intel: fix wrong poll bits in dsp power down Peter Robinson (1): ASoC: remove remnants of sirf prima/atlas audio codec Pierre-Louis Bossart (1): ASoC: SOF: Intel: unregister DMIC device on probe error Sameer Pujar (2): ASoC: simple-card-utils: Do not handle device clock ASoC: rt5659: Update MCLK rate in set_sysclk() Shengjiu Wang (3): ASoC: ak4458: Add MODULE_DEVICE_TABLE ASoC: ak5558: Add MODULE_DEVICE_TABLE ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms Srinivas Kandagatla (3): ASoC: qcom: sdm845: Fix array out of bounds access ASoC: qcom: sdm845: Fix array out of range on rx slim channels ASoC: codecs: wcd934x: add a sanity check in set channel map Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-cpu: Fix lpass dai ids parse .../devicetree/bindings/sound/fsl,spdif.yaml | 4 + drivers/spi/spi-cadence-quadspi.c | 1 + sound/soc/codecs/Kconfig | 5 - sound/soc/codecs/ak4458.c | 1 + sound/soc/codecs/ak5558.c | 1 + sound/soc/codecs/cs42l42.c | 112 ++++++------------- sound/soc/codecs/cs42l42.h | 13 ++- sound/soc/codecs/es8316.c | 9 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 28 +++-- sound/soc/codecs/lpass-wsa-macro.c | 20 ++-- sound/soc/codecs/rt1015.c | 2 + sound/soc/codecs/rt5640.c | 4 +- sound/soc/codecs/rt5651.c | 4 +- sound/soc/codecs/rt5659.c | 5 + sound/soc/codecs/rt5670.c | 110 +++++++++++++++--- sound/soc/codecs/rt5670.h | 9 +- sound/soc/codecs/rt711.c | 8 ++ sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sirf-audio-codec.h | 124 --------------------- sound/soc/codecs/wcd934x.c | 6 + sound/soc/fsl/fsl_ssi.c | 6 +- sound/soc/generic/simple-card-utils.c | 13 ++- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 +- sound/soc/mediatek/mt8192/mt8192-reg.h | 8 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/sdm845.c | 6 +- sound/soc/soc-core.c | 4 + sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda.c | 1 + 31 files changed, 232 insertions(+), 286 deletions(-) delete mode 100644 sound/soc/codecs/sirf-audio-codec.h

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[PATCH v4 2/3] dt-bindings: Add Rockchip rk817 audio CODEC support
by Chris Morgan 18 Mar '21

18 Mar '21

Create dt-binding documentation to document rk817 codec. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- .../bindings/sound/rockchip,rk817-codec.yaml | 63 +++++++++++++++++++ 1 file changed, 63 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml new file mode 100644 index 000000000000..986cebcd0832 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-rk817.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk817 audio codec + +description: + The rk817 codec is an I2C codec integrated with every Rockchip + rk817 PMIC MFD. + +properties: + compatible: + enum: + - rockchip,rk817-codec + + "#sound-dai-cells": + const: 1 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + mic-in-differential: + description: the microphone is in differential mode. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - clocks + - clock-names + +additionalProperties: false + +examples: + - | +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>; + + ........ + + rk817_codec: codec { + #sound-dai-cells = <0>; + compatible = "rockchip,rk817-codec"; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s_8ch_mclk>; + mic-in-differential; + status = "okay"; + }; + + ........ + +}; -- 2.25.1

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Re: [PATCH] ASoC: rt715: add main capture switch and main capture volume
by Mark Brown 18 Mar '21

18 Mar '21

On Thu, Mar 18, 2021 at 12:19:18PM +0000, Jack Yu wrote: > > This change is literally just removing some brackets that don't do anything, the > > regmap API has no impact here. > Sorry, I misunderstand your reply, and yes, this modification just remove some brackets. > Should I send it with separate patch? Yes, sure.

1 0

Re: [PATCH] conf: fix memory leak on the error path in parse_args()
by Mark Hills 18 Mar '21

18 Mar '21

Having a little trouble which bisected to this patch. First noticed it's causing Chromium to crash out one of its subprocesses (stack trace below) Can actually be replicated with a simple "aplay -L": aplay: conf.c:2207: snd_config_delete: Assertion `config' failed. Aborted (core dumped) Program received signal SIGABRT, Aborted. 0x00007ffff7984c7b in raise () from /lib64/libc.so.6 (gdb) bt #0 0x00007ffff7984c7b in raise () from /lib64/libc.so.6 #1 0x00007ffff7965548 in abort () from /lib64/libc.so.6 #2 0x00007ffff796542f in __assert_fail_base.cold () from /lib64/libc.so.6 #3 0x00007ffff7975fc2 in __assert_fail () from /lib64/libc.so.6 #4 0x00007ffff7cce7e2 in snd_config_delete (config=<optimized out>) at conf.c:2207 #5 0x00007ffff7cd004d in parse_args (subs=0x6c66f0, str=<optimized out>, str@entry=0x641fa5 "CARD=Layla3G,DEV=0", defs=<optimized out>) at conf.c:5172 #6 0x00007ffff7cd0c31 in snd_config_expand (config=0x6594a0, root=root@entry=0x639330, args=args@entry=0x641fa5 "CARD=Layla3G,DEV=0", private_data=private_data@entry=0x0, result=result@entry=0x7fffffffdf80) at conf.c:5227 #7 0x00007ffff7cd13e2 in snd_config_search_definition (config=config@entry=0x639330, base=base@entry=0x40cf74 "pcm", name=name@entry=0x641fa0 "plug:CARD=Layla3G,DEV=0", result=result@entry=0x7fffffffdf80) at conf.c:5312 #8 0x00007ffff7cd8154 in try_config (config=config@entry=0x639330, list=list@entry=0x7fffffffe260, base=0x40cf74 "pcm", name=<optimized out>) at namehint.c:269 #9 0x00007ffff7cd8e8a in add_card (config=<optimized out>, rw_config=0x639330, list=list@entry=0x7fffffffe260, card=0) at namehint.c:488 #10 0x00007ffff7cd9410 in snd_device_name_hint (card=0, card@entry=-1, iface=<optimized out>, iface@entry=0x40cf74 "pcm", hints=hints@entry=0x7fffffffe308) at namehint.c:639 #11 0x000000000040442b in pcm_list () at aplay.c:339 #12 0x000000000040a638 in main (argc=2, argv=0x7fffffffe5c8) at aplay.c:824 I saw "namehint" in the stack; I'm using this fragment of asoundrc (it's the only way to get chosen ALSA devices available in Chrome) but it seems to be unrelated; commenting it out and the assertion still fails. I haven't had time to look any deeper. namehint.pcm [ "layla1|Layla: Microphone input 1" "layla2|Layla: Microphone input 2" "layla12|Layla: Microphone inputs 1,2" "layla34|Layla: Line inputs 3,4 (DJ mixer)" "layla|Layla: Analogue inputs 1,2,3,4" "plug:shure|Shure X2U: Microphone input" "plug:dj1|Audio 8 DJ: Input 1 (Left deck)" "plug:dj2|Audio 8 DJ: Input 2 (Right deck)" "plug:local|Local audio bus" ] -- Mark >From ad5f255b4767d52f7aaafd3eb348c680e990f0b8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Wed, 10 Mar 2021 18:10:57 +0100 Subject: [PATCH] conf: fix memory leak on the error path in parse_args() Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- src/conf.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf.c b/src/conf.c index 38eefbf8..14b14b59 100644 --- a/src/conf.c +++ b/src/conf.c @@ -5169,6 +5169,7 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) err = snd_config_add(subs, sub); if (err < 0) { _err: + snd_config_delete(sub); free(val); return err; } -- 2.17.5 chrome: conf.c:2207: snd_config_delete: Assertion `config' failed. Received signal 6 #0 0x55c2535f35e9 base::debug::CollectStackTrace() #1 0x55c25355e723 base::debug::StackTrace::StackTrace() #2 0x55c2535f3190 base::debug::(anonymous namespace)::StackDumpSignalHandler() #3 0x7f35b16e6690 (/lib64/libpthread-2.30.so+0x1368f) #4 0x7f35ac583c7b __GI_raise #5 0x7f35ac564548 __GI_abort #6 0x7f35ac56442f __assert_fail_base.cold #7 0x7f35ac574fc2 __GI___assert_fail #8 0x7f35acda27e2 snd_config_delete #9 0x7f35acda404d parse_args #10 0x7f35acda4c31 snd_config_expand #11 0x7f35acda53e2 snd_config_search_definition #12 0x7f35acdac154 try_config #13 0x7f35acdace8a add_card #14 0x7f35acdad13c snd_device_name_hint #15 0x55c250eb3b13 media::AudioManagerAlsa::HasAnyAlsaAudioDevice() #16 0x55c250ea9445 media::AudioSystemHelper::ComputeOutputParameters() #17 0x55c250ea93e0 media::AudioSystemHelper::GetOutputStreamParameters() #18 0x55c251d38db3 audio::SystemInfo::GetOutputStreamParameters() #19 0x55c2511bd455 audio::mojom::SystemInfoStubDispatch::AcceptWithResponder() #20 0x55c253d0bad3 mojo::InterfaceEndpointClient::HandleValidatedMessage() #21 0x55c253d0e8bd mojo::MessageDispatcher::Accept() #22 0x55c253d12380 mojo::internal::MultiplexRouter::ProcessIncomingMessage() #23 0x55c253d11b09 mojo::internal::MultiplexRouter::Accept() #24 0x55c253d0e8f9 mojo::MessageDispatcher::Accept() #25 0x55c253d0a184 mojo::Connector::DispatchMessage() #26 0x55c253d0a8b7 mojo::Connector::ReadAllAvailableMessages() #27 0x55c25056d034 base::internal::Invoker<>::RunOnce() #28 0x55c2535b52b2 base::TaskAnnotator::RunTask() #29 0x55c2535c6c11 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() #30 0x55c2535c68ea base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() #31 0x55c25357669a base::MessagePumpDefault::Run() #32 0x55c2535c725b base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() #33 0x55c25359820b base::RunLoop::Run() #34 0x55c2534d01da content::UtilityMain() #35 0x55c2534fba9c content::RunZygote() #36 0x55c2534fcd6b content::ContentMainRunnerImpl::Run() #37 0x55c2534fa35d content::RunContentProcess() #38 0x55c2534facfd content::ContentMain() #39 0x55c25056a068 ChromeMain #40 0x7f35ac565e6b __libc_start_main #41 0x55c250569eaa _start r8: 0000000000000000 r9: 00007ffd94a02da0 r10: 0000000000000008 r11: 0000000000000246 r12: 00007f35ace23e21 r13: 000000000000089f r14: 00007f35ace2af41 r15: 0000000000000004 di: 0000000000000002 si: 00007ffd94a02da0 bp: 00007f35ac6ebf38 bx: 00007f35a5d83f00 dx: 0000000000000000 ax: 0000000000000000 cx: 00007f35ac583c7b sp: 00007ffd94a02e18 ip: 00007f35ac583c7b efl: 0000000000000246 cgf: 002b000000000033 erf: 0000000000000000 trp: 0000000000000000 msk: 0000000000000000 cr2: 0000000000000000 [end of stack trace] Calling _exit(1). Core file will not be generated.

2 2

Re: [PATCH] ASoC: rt715: add main capture switch and main capture volume
by Mark Brown 18 Mar '21

18 Mar '21

On Thu, Mar 18, 2021 at 03:35:31AM +0000, Jack Yu wrote: > > > - *l_val = (val_h << 8); > > > + *l_val = val_h << 8; > > > ret = regmap_read(rt715->regmap, addr_h, l_val); > > > if (ret < 0) > > > pr_err("Failed to get L channel gain.\n"); > > This looks like an unrelated formatting change. > Because regmap has its own callback for read/write, so the formatting change is required for callback. This change is literally just removing some brackets that don't do anything, the regmap API has no impact here. > > > if (mc->invert) { > > > /* for mute status */ > > > - read_ll = !((read_ll & 0x80) >> RT715_MUTE_SFT); > > > - read_rl = !((read_rl & 0x80) >> RT715_MUTE_SFT); > > > + read_ll = !(read_ll & 0x80); > > > + read_rl = !(read_rl & 0x80); > > > } else { > > > /* for gain */ > > > read_ll = read_ll & 0x7f; > > This too. > This change is just to simplify the code. The result will still be the same. Sure, but that should be a separate patch with a separate changelog which explicitly says that this is being done.

1 0

[PATCH] ALSA: hda: Fix spelling mistakes
by hjh 18 Mar '21

18 Mar '21

Signed-off-by: hjh <huangjianghui(a)uniontech.com> --- sound/hda/hdac_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index a6ed3dc35f7e..1eb8563db2df 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -618,7 +618,7 @@ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger); /** - * snd_hdac_stream_sync - sync with start/strop trigger operation + * snd_hdac_stream_sync - sync with start/stop trigger operation * @azx_dev: HD-audio core stream (master stream) * @start: true = start, false = stop * @streams: bit flags of streams to sync -- 2.20.1

2 1

[PATCH] dt-bindings: Drop type references on common properties
by Rob Herring 18 Mar '21

18 Mar '21

Users of common properties shouldn't have a type definition as the common schemas already have one. Drop all the unnecessary type references in the tree. A meta-schema update to catch these is pending. Cc: Nicolas Saenz Julienne <nsaenzjulienne(a)suse.de> Cc: Maxime Ripard <mripard(a)kernel.org> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Marc Kleine-Budde <mkl(a)pengutronix.de> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Ohad Ben-Cohen <ohad(a)wizery.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Benson Leung <bleung(a)chromium.org> Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: Daniel Lezcano <daniel.lezcano(a)linaro.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Stefan Wahren <wahrenst(a)gmx.net> Cc: Masahiro Yamada <yamada.masahiro(a)socionext.com> Cc: Odelu Kukatla <okukatla(a)codeaurora.org> Cc: Alex Elder <elder(a)kernel.org> Cc: Suman Anna <s-anna(a)ti.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> Cc: linux-gpio(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-can(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-remoteproc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-usb(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/arm/bcm/raspberrypi,bcm2835-firmware.yaml | 5 +---- Documentation/devicetree/bindings/arm/cpus.yaml | 2 -- .../bindings/display/allwinner,sun4i-a10-tcon.yaml | 1 - .../devicetree/bindings/gpio/socionext,uniphier-gpio.yaml | 3 +-- .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 1 - .../devicetree/bindings/interconnect/qcom,rpmh.yaml | 1 - .../bindings/memory-controllers/nvidia,tegra210-emc.yaml | 2 +- Documentation/devicetree/bindings/net/can/fsl,flexcan.yaml | 1 - Documentation/devicetree/bindings/net/qcom,ipa.yaml | 1 - Documentation/devicetree/bindings/nvmem/nvmem-consumer.yaml | 2 -- .../devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml | 2 +- Documentation/devicetree/bindings/sound/ak4642.yaml | 2 -- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 - .../devicetree/bindings/thermal/qcom-spmi-adc-tm5.yaml | 1 - Documentation/devicetree/bindings/usb/usb.yaml | 1 - 16 files changed, 5 insertions(+), 23 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/bcm/raspberrypi,bcm2835-firmware.yaml b/Documentation/devicetree/bindings/arm/bcm/raspberrypi,bcm2835-firmware.yaml index a2c63c8b1d10..c6144c8421fa 100644 --- a/Documentation/devicetree/bindings/arm/bcm/raspberrypi,bcm2835-firmware.yaml +++ b/Documentation/devicetree/bindings/arm/bcm/raspberrypi,bcm2835-firmware.yaml @@ -26,10 +26,7 @@ properties: - const: simple-mfd mboxes: - $ref: '/schemas/types.yaml#/definitions/phandle' - description: | - Phandle to the firmware device's Mailbox. - (See: ../mailbox/mailbox.txt for more information) + maxItems: 1 clocks: type: object diff --git a/Documentation/devicetree/bindings/arm/cpus.yaml b/Documentation/devicetree/bindings/arm/cpus.yaml index 26b886b20b27..6be4a8852ee5 100644 --- a/Documentation/devicetree/bindings/arm/cpus.yaml +++ b/Documentation/devicetree/bindings/arm/cpus.yaml @@ -256,13 +256,11 @@ properties: where voltage is in V, frequency is in MHz. power-domains: - $ref: '/schemas/types.yaml#/definitions/phandle-array' description: List of phandles and PM domain specifiers, as defined by bindings of the PM domain provider (see also ../power_domain.txt). power-domain-names: - $ref: '/schemas/types.yaml#/definitions/string-array' description: A list of power domain name strings sorted in the same order as the power-domains property. diff --git a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml index c13faf3e6581..3a7d5d731712 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml @@ -73,7 +73,6 @@ properties: clock-output-names: description: Name of the LCD pixel clock created. - $ref: /schemas/types.yaml#/definitions/string-array maxItems: 1 dmas: diff --git a/Documentation/devicetree/bindings/gpio/socionext,uniphier-gpio.yaml b/Documentation/devicetree/bindings/gpio/socionext,uniphier-gpio.yaml index 1a54db04f29d..bcafa494ed7a 100644 --- a/Documentation/devicetree/bindings/gpio/socionext,uniphier-gpio.yaml +++ b/Documentation/devicetree/bindings/gpio/socionext,uniphier-gpio.yaml @@ -43,8 +43,7 @@ properties: gpio-ranges: true - gpio-ranges-group-names: - $ref: /schemas/types.yaml#/definitions/string-array + gpio-ranges-group-names: true socionext,interrupt-ranges: description: | diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 6f2398cdc82d..1e7894e524f9 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -102,7 +102,6 @@ patternProperties: st,adc-channel-names: description: List of single-ended channel names. - $ref: /schemas/types.yaml#/definitions/string-array st,filter-order: description: | diff --git a/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml b/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml index 799e73cdb90b..13da7b29c707 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml @@ -82,7 +82,6 @@ properties: this interconnect to send RPMh commands. qcom,bcm-voter-names: - $ref: /schemas/types.yaml#/definitions/string-array description: | Names for each of the qcom,bcm-voters specified. diff --git a/Documentation/devicetree/bindings/memory-controllers/nvidia,tegra210-emc.yaml b/Documentation/devicetree/bindings/memory-controllers/nvidia,tegra210-emc.yaml index 49ab09252e52..bc8477e7ab19 100644 --- a/Documentation/devicetree/bindings/memory-controllers/nvidia,tegra210-emc.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/nvidia,tegra210-emc.yaml @@ -34,7 +34,7 @@ properties: - description: EMC general interrupt memory-region: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle to a reserved memory region describing the table of EMC frequencies trained by the firmware diff --git a/Documentation/devicetree/bindings/net/can/fsl,flexcan.yaml b/Documentation/devicetree/bindings/net/can/fsl,flexcan.yaml index fe6a949a2eab..55bff1586b6f 100644 --- a/Documentation/devicetree/bindings/net/can/fsl,flexcan.yaml +++ b/Documentation/devicetree/bindings/net/can/fsl,flexcan.yaml @@ -57,7 +57,6 @@ properties: - const: per clock-frequency: - $ref: /schemas/types.yaml#/definitions/uint32 description: | The oscillator frequency driving the flexcan device, filled in by the boot loader. This property should only be used the used operating system diff --git a/Documentation/devicetree/bindings/net/qcom,ipa.yaml b/Documentation/devicetree/bindings/net/qcom,ipa.yaml index 8f86084bf12e..4e8dee4aa90d 100644 --- a/Documentation/devicetree/bindings/net/qcom,ipa.yaml +++ b/Documentation/devicetree/bindings/net/qcom,ipa.yaml @@ -100,7 +100,6 @@ properties: - description: Whether the IPA clock is enabled (if valid) qcom,smem-state-names: - $ref: /schemas/types.yaml#/definitions/string-array description: The names of the state bits used for SMP2P output items: - const: ipa-clock-enabled-valid diff --git a/Documentation/devicetree/bindings/nvmem/nvmem-consumer.yaml b/Documentation/devicetree/bindings/nvmem/nvmem-consumer.yaml index d5d7f113bade..828e4a1ece41 100644 --- a/Documentation/devicetree/bindings/nvmem/nvmem-consumer.yaml +++ b/Documentation/devicetree/bindings/nvmem/nvmem-consumer.yaml @@ -23,12 +23,10 @@ properties: List of phandle to the nvmem data cells. nvmem-names: - $ref: /schemas/types.yaml#/definitions/string-array description: Names for the each nvmem provider. nvmem-cell-names: - $ref: /schemas/types.yaml#/definitions/string-array description: Names for each nvmem-cells specified. diff --git a/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml b/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml index 1a1159097a2a..73400bc6e91d 100644 --- a/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml @@ -93,7 +93,7 @@ properties: # The following are the optional properties: memory-region: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: | phandle to the reserved memory node to be associated with the remoteproc device. The reserved memory node diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml index 6cd213be2266..1e2caa29790e 100644 --- a/Documentation/devicetree/bindings/sound/ak4642.yaml +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -29,11 +29,9 @@ properties: clock-frequency: description: common clock binding; frequency of MCKO - $ref: /schemas/types.yaml#/definitions/uint32 clock-output-names: description: common clock name - $ref: /schemas/types.yaml#/definitions/string required: - compatible diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index acfb9db021dc..77adbebed824 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -32,7 +32,7 @@ properties: The last one integer is the length of the shared memory. memory-region: - $ref: '/schemas/types.yaml#/definitions/phandle' + maxItems: 1 description: | Shared memory region to EC. A "shared-dma-pool". See ../reserved-memory/reserved-memory.txt for details. diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 2e1046513603..e494a0416748 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -78,7 +78,6 @@ properties: clock-frequency: description: for audio_clkout0/1/2/3 - $ref: /schemas/types.yaml#/definitions/uint32-array clkout-lr-asynchronous: description: audio_clkoutn is asynchronizes with lr-clock. diff --git a/Documentation/devicetree/bindings/thermal/qcom-spmi-adc-tm5.yaml b/Documentation/devicetree/bindings/thermal/qcom-spmi-adc-tm5.yaml index 95a728f4d333..3ea8c0c1f45f 100644 --- a/Documentation/devicetree/bindings/thermal/qcom-spmi-adc-tm5.yaml +++ b/Documentation/devicetree/bindings/thermal/qcom-spmi-adc-tm5.yaml @@ -59,7 +59,6 @@ patternProperties: properties: reg: - $ref: /schemas/types.yaml#/definitions/uint32 description: Specify the sensor channel. There are 8 channels in PMIC5's ADC TM minimum: 0 maximum: 7 diff --git a/Documentation/devicetree/bindings/usb/usb.yaml b/Documentation/devicetree/bindings/usb/usb.yaml index 78491e66ed24..939f217b8c7b 100644 --- a/Documentation/devicetree/bindings/usb/usb.yaml +++ b/Documentation/devicetree/bindings/usb/usb.yaml @@ -16,7 +16,6 @@ properties: pattern: "^usb(@.*)?" phys: - $ref: /schemas/types.yaml#/definitions/phandle-array description: List of all the USB PHYs on this HCD -- 2.27.0

11 10

[PATCH] ALSA: ctxfi: fix comment syntax in file headers
by Aditya Srivastava 18 Mar '21

18 Mar '21

The opening comment mark '/**' is used for highlighting the beginning of kernel-doc comments. There are files in sound/pci/ctxfi which follow this syntax in their file headers, i.e. start with '/**' like comments, which causes unexpected warnings from kernel-doc. E.g., running scripts/kernel-doc -none on sound/pci/ctxfi/ctresource.c causes this warning: "warning: wrong kernel-doc identifier on line: * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved." Similarly for other files too. Provide a simple fix by replacing the kernel-doc like comment syntax with general format, i.e. "/*", to prevent kernel-doc from parsing it. Signed-off-by: Aditya Srivastava <yashsri421(a)gmail.com> --- sound/pci/ctxfi/ct20k1reg.h | 2 +- sound/pci/ctxfi/ct20k2reg.h | 2 +- sound/pci/ctxfi/ctamixer.c | 2 +- sound/pci/ctxfi/ctamixer.h | 2 +- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/ctxfi/ctatc.h | 2 +- sound/pci/ctxfi/ctdaio.c | 2 +- sound/pci/ctxfi/ctdaio.h | 2 +- sound/pci/ctxfi/cthardware.h | 2 +- sound/pci/ctxfi/cthw20k1.h | 2 +- sound/pci/ctxfi/cthw20k2.h | 2 +- sound/pci/ctxfi/ctimap.h | 2 +- sound/pci/ctxfi/ctmixer.h | 2 +- sound/pci/ctxfi/ctpcm.h | 2 +- sound/pci/ctxfi/ctresource.c | 2 +- sound/pci/ctxfi/ctresource.h | 2 +- sound/pci/ctxfi/ctsrc.c | 2 +- sound/pci/ctxfi/ctsrc.h | 2 +- sound/pci/ctxfi/ctvmem.c | 2 +- sound/pci/ctxfi/ctvmem.h | 2 +- 20 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h index d4bfee499fb1..05bb006c0f4c 100644 --- a/sound/pci/ctxfi/ct20k1reg.h +++ b/sound/pci/ctxfi/ct20k1reg.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. */ diff --git a/sound/pci/ctxfi/ct20k2reg.h b/sound/pci/ctxfi/ct20k2reg.h index af94ea66fdda..02f67828eabe 100644 --- a/sound/pci/ctxfi/ct20k2reg.h +++ b/sound/pci/ctxfi/ct20k2reg.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. */ diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index d4ff377eb3a3..da6e6350ceaf 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0-only -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctamixer.c diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h index 4fafb397abed..4498e6139d0e 100644 --- a/sound/pci/ctxfi/ctamixer.h +++ b/sound/pci/ctxfi/ctamixer.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctamixer.h diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index f8ac96cf38a4..78f35e88aed6 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0-only -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctatc.c diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index ac31b32b277b..0bc7b71d910b 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctatc.h diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 4cb47b5a792c..f589da045342 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0-only -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctdaio.c diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h index 431583bb0a3e..bd6310f48013 100644 --- a/sound/pci/ctxfi/ctdaio.h +++ b/sound/pci/ctxfi/ctdaio.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctdaio.h diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 9e6b83bd432d..f406b626a28c 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File cthardware.h diff --git a/sound/pci/ctxfi/cthw20k1.h b/sound/pci/ctxfi/cthw20k1.h index b7cbe82d71bd..ffb019abf651 100644 --- a/sound/pci/ctxfi/cthw20k1.h +++ b/sound/pci/ctxfi/cthw20k1.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File cthw20k1.h diff --git a/sound/pci/ctxfi/cthw20k2.h b/sound/pci/ctxfi/cthw20k2.h index 797b13dcd84c..6993a3d5277a 100644 --- a/sound/pci/ctxfi/cthw20k2.h +++ b/sound/pci/ctxfi/cthw20k2.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File cthw20k2.h diff --git a/sound/pci/ctxfi/ctimap.h b/sound/pci/ctxfi/ctimap.h index 79bc94bce4d5..49b1bb831410 100644 --- a/sound/pci/ctxfi/ctimap.h +++ b/sound/pci/ctxfi/ctimap.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctimap.h diff --git a/sound/pci/ctxfi/ctmixer.h b/sound/pci/ctxfi/ctmixer.h index 770dc18a85e8..e812f6c93b41 100644 --- a/sound/pci/ctxfi/ctmixer.h +++ b/sound/pci/ctxfi/ctmixer.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctmixer.h diff --git a/sound/pci/ctxfi/ctpcm.h b/sound/pci/ctxfi/ctpcm.h index dfa1c62f7d1e..8b39bdd262b4 100644 --- a/sound/pci/ctxfi/ctpcm.h +++ b/sound/pci/ctxfi/ctpcm.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctpcm.h diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 6d0a01b189e1..81ad26934518 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0-only -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctresource.c diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h index 93e47488a1c1..fdbfd808816d 100644 --- a/sound/pci/ctxfi/ctresource.h +++ b/sound/pci/ctxfi/ctresource.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctresource.h diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index 37c18ce84974..bd4697b44233 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0-only -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctsrc.c diff --git a/sound/pci/ctxfi/ctsrc.h b/sound/pci/ctxfi/ctsrc.h index 1204962280c8..1124daf50c9b 100644 --- a/sound/pci/ctxfi/ctsrc.h +++ b/sound/pci/ctxfi/ctsrc.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctsrc.h diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index bde28aa9e139..7a805c4a58e1 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0-only -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctvmem.c diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index 54818a3c245d..da54cbcdb0be 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. * * @File ctvmem.h -- 2.17.1

3 2

[PATCH] ALSA: asihpi: fix comment syntax in file headers
by Aditya Srivastava 18 Mar '21

18 Mar '21

The opening comment mark '/**' is used for highlighting the beginning of kernel-doc comments. There are files in sound/pci/asihpi which follow this syntax in their file headers, i.e. start with '/**' like comments, which causes unexpected warnings from kernel-doc. E.g., running scripts/kernel-doc -none on sound/pci/asihpi/hpidspcd.h causes this warning: "warning: Cannot understand on line 4 - I thought it was a doc line" Provide a simple fix by replacing the kernel-doc like comment syntax with general format, i.e. "/*", to prevent kernel-doc from parsing it. Signed-off-by: Aditya Srivastava <yashsri421(a)gmail.com> --- * Applies perfectly on next-20210312 sound/pci/asihpi/hpicmn.h | 2 +- sound/pci/asihpi/hpidspcd.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h index de3bedd29d94..8ec656cf8848 100644 --- a/sound/pci/asihpi/hpicmn.h +++ b/sound/pci/asihpi/hpicmn.h @@ -1,5 +1,5 @@ /* SPDX-License-Identifier: GPL-2.0-only */ -/** +/* AudioScience HPI driver Copyright (C) 1997-2014 AudioScience Inc. <support(a)audioscience.com> diff --git a/sound/pci/asihpi/hpidspcd.h b/sound/pci/asihpi/hpidspcd.h index a01e8c6092bd..9f1468ed7096 100644 --- a/sound/pci/asihpi/hpidspcd.h +++ b/sound/pci/asihpi/hpidspcd.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /***********************************************************************/ -/** +/* AudioScience HPI driver Copyright (C) 1997-2011 AudioScience Inc. <support(a)audioscience.com> -- 2.17.1

3 2

[PATCH v4 3/3] mfd: Add Rockchip rk817 audio CODEC support
by Chris Morgan 17 Mar '21

17 Mar '21

Add rk817 codec support cell to rk808 mfd driver. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- drivers/mfd/rk808.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..adb8a7da29db 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -163,6 +163,12 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + .of_compatible = "rockchip,rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { -- 2.25.1

2 1

[PATCH] sound: soc: ti: Fix a typo in the file ams-delta.c
by Bhaskar Chowdhury 17 Mar '21

17 Mar '21

s/functonality/functionality/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/soc/ti/ams-delta.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 57feb473a579..31462587f922 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -408,7 +408,7 @@ static struct tty_ldisc_ops cx81801_ops = { /* * Even if not very useful, the sound card can still work without any of the - * above functonality activated. You can still control its audio input/output + * above functionality activated. You can still control its audio input/output * constellation and speakerphone gain from userspace by issuing AT commands * over the modem port. */ -- 2.30.2

2 3

[PATCH] ASoC: rt715: add main capture switch and main capture volume
by Jack Yu 17 Mar '21

17 Mar '21

Add main capture switch and main capture volume control. Main capture control has its own channel value respectivelly. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 400 ++++++++++++++++++++++++++++++++------- 1 file changed, 327 insertions(+), 73 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index f419d001a3fb..a9e4741269ba 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -57,14 +57,14 @@ static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, { int ret; /* R Channel */ - *r_val = (val_h << 8); + *r_val = val_h << 8; ret = regmap_read(rt715->regmap, addr_l, r_val); if (ret < 0) pr_err("Failed to get R channel gain.\n"); /* L Channel */ val_h |= 0x20; - *l_val = (val_h << 8); + *l_val = val_h << 8; ret = regmap_read(rt715->regmap, addr_h, l_val); if (ret < 0) pr_err("Failed to get L channel gain.\n"); @@ -80,85 +80,102 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; - unsigned int read_ll, read_rl; - int i; - - /* Can't use update bit function, so read the original value first */ - addr_h = mc->reg; - addr_l = mc->rreg; - if (mc->shift == RT715_DIR_OUT_SFT) /* output */ - val_h = 0x80; - else /* input */ - val_h = 0x0; - - rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); - - /* L Channel */ - if (mc->invert) { - /* for mute */ - val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; - /* keep gain */ - read_ll = read_ll & 0x7f; - val_ll |= read_ll; - } else { - /* for gain */ - val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); - if (val_ll > mc->max) - val_ll = mc->max; - /* keep mute status */ - read_ll = read_ll & 0x80; - val_ll |= read_ll; - } - - /* R Channel */ - if (mc->invert) { - regmap_write(rt715->regmap, - RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); - /* for mute */ - val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; - /* keep gain */ - read_rl = read_rl & 0x7f; - val_lr |= read_rl; - } else { - /* for gain */ - val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); - if (val_lr > mc->max) - val_lr = mc->max; - /* keep mute status */ - read_rl = read_rl & 0x80; - val_lr |= read_rl; - } + unsigned int read_ll, read_rl, i, j, loop_cnt; - for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (strstr(ucontrol->id.name, "Main Capture Switch") || + strstr(ucontrol->id.name, "Main Capture Volume")) + loop_cnt = 4; + else + loop_cnt = 1; - if (val_ll == val_lr) { - /* Set both L/R channels at the same time */ - val_h = (1 << mc->shift) | (3 << 4); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_ll)); + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + if (loop_cnt == 1) { + addr_h = mc->reg; + addr_l = mc->rreg; } else { - /* Lch*/ - val_h = (1 << mc->shift) | (1 << 5); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - /* Rch */ - val_h = (1 << mc->shift) | (1 << 4); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_lr)); + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; } - /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ val_h = 0x80; else /* input */ val_h = 0x0; - rt715_get_gain(rt715, addr_h, addr_l, val_h, - &read_rl, &read_ll); - if (read_rl == val_lr && read_ll == val_ll) - break; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + } + + /* R Channel */ + if (mc->invert) { + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } } /* D0:power on state, D3: power saving mode */ if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) @@ -188,8 +205,8 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, if (mc->invert) { /* for mute status */ - read_ll = !((read_ll & 0x80) >> RT715_MUTE_SFT); - read_rl = !((read_rl & 0x80) >> RT715_MUTE_SFT); + read_ll = !(read_ll & 0x80); + read_rl = !(read_rl & 0x80); } else { /* for gain */ read_ll = read_ll & 0x7f; @@ -201,9 +218,227 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt715_set_main_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + unsigned int k_shift = RT715_DIR_IN_SFT; + + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for mute */ + val_ll = (!ucontrol->value.integer.value[j * 2]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + + /* R Channel */ + /* for mute */ + val_lr = (!ucontrol->value.integer.value[j * 2 + 1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_main_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = !(read_ll & 0x80); + ucontrol->value.integer.value[i * 2 + 1] = !(read_rl & 0x80); + } + + return 0; +} + +static int rt715_set_main_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + unsigned int k_shift = RT715_DIR_IN_SFT, k_max = 0x3f; + + for (j = 0; j < loop_cnt; j++) { + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for gain */ + val_ll = ((ucontrol->value.integer.value[j * 2]) & 0x7f); + if (val_ll > k_max) + val_ll = k_max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + + /* R Channel */ + /* for gain */ + val_lr = ((ucontrol->value.integer.value[j * 2 + 1]) & 0x7f); + if (val_lr > k_max) + val_lr = k_max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_main_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = read_ll & 0x7f; + ucontrol->value.integer.value[i * 2 + 1] = read_rl & 0x7f; + } + + return 0; +} + static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); +static int rt715_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int rt715_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x3f; + return 0; +} + #define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -212,6 +447,21 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define RT715_MAIN_SWITCH_EXT(xname, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt715_switch_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + +#define RT715_MAIN_VOL_EXT_TLV(xname, xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt715_vol_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, @@ -226,6 +476,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, rt715_set_amp_gain_get, rt715_set_amp_gain_put), + RT715_MAIN_SWITCH_EXT("Main Capture Switch", + rt715_set_main_switch_get, rt715_set_main_switch_put), /* Volume Control */ SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, @@ -243,6 +495,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, in_vol_tlv), + RT715_MAIN_VOL_EXT_TLV("Main Capture Volume", + rt715_set_main_vol_get, rt715_set_main_vol_put, in_vol_tlv), /* MIC Boost Control */ SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, -- 2.29.0

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