June 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v4] soundwire: intel: move to auxiliary bus
by Bard Liao 14 Jun '21

14 Jun '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. v3: - Modify the function description of sdw_intel_probe() which was missing in previous version. v4: - Move intel_link_dev_unregister(ldev) before pm_runtime_put_noidle( ldev->link_res.dev) to fix use-after-free reported by KASAN. --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 56 ++++--- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 232 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 202 insertions(+), 107 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 30ce95ec2d70..9e283bef53d2 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); + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); + + intel_link_dev_unregister(ldev); } 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); if (!link->cdns) { dev_err(&adev->dev, "failed to get link->cdns\n"); @@ -194,8 +266,7 @@ 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; @@ -213,8 +284,14 @@ 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; } @@ -222,7 +299,7 @@ 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; @@ -241,27 +318,28 @@ 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); } } @@ -272,8 +350,8 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) * sdw_intel_probe() - SoundWire Intel probe routine * @res: resource data * - * This registers a platform device for each Master handled by the controller, - * and SoundWire Master and Slave devices will be created by the platform + * This registers an auxiliary device for each Master handled by the controller, + * and SoundWire Master and Slave devices will be created by the auxiliary * device probe. All the information necessary is stored in the context, and * the res argument pointer can be freed after this step. * This function will be called after sdw_intel_acpi_scan() by SOF probe. @@ -306,27 +384,31 @@ 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

5 14

[PATCH] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Kuninori Morimoto 14 Jun '21

14 Jun '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..abe9d539709b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -393,7 +393,7 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, name, parent, 0, 0); if (IS_ERR(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; -- 2.25.1

2 2

Some minor fixes for Topology2.0
by GitHub pull_request - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-utils pull request #98 was opened from ranj063: Request URL : https://github.com/alsa-project/alsa-utils/pull/98 Patch URL : https://github.com/alsa-project/alsa-utils/pull/98.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH -next 0/9] ASoC: fsl: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 13 Jun '21

13 Jun '21

patch #1 ~ #8: Use devm_platform_get_and_ioremap_resource() patch #9 check return value of platform_get_resource_byname() Yang Yingliang (9): ASoC: fsl_asrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_aud2htx: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_easrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_esai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_micfil: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_sai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_spdif: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_ssi: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_xcvr: check return value after calling platform_get_resource_byname() sound/soc/fsl/fsl_asrc.c | 3 +-- sound/soc/fsl/fsl_aud2htx.c | 3 +-- sound/soc/fsl/fsl_easrc.c | 3 +-- sound/soc/fsl/fsl_esai.c | 3 +-- sound/soc/fsl/fsl_micfil.c | 3 +-- sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/fsl/fsl_spdif.c | 3 +-- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/fsl_xcvr.c | 4 ++++ 9 files changed, 12 insertions(+), 16 deletions(-) -- 2.25.1

2 10

alsa-lib 1.2.5 breaks espeak
by GitHub issues - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-lib issue #155 was opened from MrBitBucket: In archlinux amd64 upgrading alsa-lib from 1.2.4 - 1.2.5 breaks the espeak application https://sourceforge.net/projects/espeak/. I see extra errors in espeak output eg ```Expression 'framesAvail' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3629 Expression 'PaAlsaStreamComponent_GetAvailableFrames( self, &framesAvail, xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3989 Expression 'PaAlsaStreamComponent_RegisterChannels( &self->playback, &self->bufferProcessor, &playbackFrames, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4118 Expression 'PaAlsaStream_SetUpBuffers( stream, &framesGot, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4359``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/155 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ALSA: bebob/fireworks: device entry corrections
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Hi, After working for hardware database of systemd[1], I realize that current device entries of ALSA bebob/fireworks drivers includes some issues. This patchset corrects them. [1] https://github.com/systemd/systemd/pull/19124 Takashi Sakamoto (6): ALSA: bebob: fulfil device entries ALSA: fireworks: fulfil device entries ALSA: bebob: correct device entry for Mackie D.2 FireWire option card ALSA: bebob: correct device entry for Acoustic Reality eAR Master One, Eroica, Figaro, and Ciaccona ALSA: bebob: code refactoring for M-Audio models ALSA: bebob: correct device entries for Phonic Helix Board and FireFly series sound/firewire/bebob/bebob.c | 92 ++++++++++++++-------------- sound/firewire/bebob/bebob.h | 9 --- sound/firewire/fireworks/fireworks.c | 15 +++++ sound/firewire/fireworks/fireworks.h | 8 --- 4 files changed, 61 insertions(+), 63 deletions(-) -- 2.30.2

2 7

[PATCH][next] ALSA: i2c: tea6330t: Remove redundant initialization of variable err
by Colin King 12 Jun '21

12 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The variable err is being initialized with a value that is never read, it is being updated later on. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/tea6330t.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 742d0f724375..037d6293f728 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -284,7 +284,7 @@ int snd_tea6330t_update_mixer(struct snd_card *card, struct tea6330t *tea; const struct snd_kcontrol_new *knew; unsigned int idx; - int err = -ENOMEM; + int err; u8 default_treble, default_bass; unsigned char bytes[7]; -- 2.31.1

2 1

[PATCH] ALSA: firewire-lib: replace in_interrupt() with in_softirq()
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Tasklet is used to handle isochronous context of 1394 OHCI. The explicit usage of in_softirq() may be preferable than in_interrupt(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 2 +- sound/firewire/amdtp-stream.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index aa53c13b89d3..5fd2aeccdfc2 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -49,7 +49,7 @@ TRACE_EVENT(amdtp_packet, __entry->data_blocks = data_blocks; __entry->data_block_counter = data_block_counter, __entry->packet_index = packet_index; - __entry->irq = !!in_interrupt(); + __entry->irq = !!in_softirq(); __entry->index = index; ), TP_printk( diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1d9bc7b07df1..aad9778d1c4d 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -610,7 +610,7 @@ static void update_pcm_pointers(struct amdtp_stream *s, // buffer associated to PCM substream to process PCM frames in the buffer, instead // of receiving notification of period elapsed by poll wait. if (!pcm->runtime->no_period_wakeup) { - if (in_interrupt()) { + if (in_softirq()) { // In software IRQ context for 1394 OHCI. snd_pcm_period_elapsed(pcm); } else { @@ -1026,7 +1026,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, const __be32 *ctx_header, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (in_softirq()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1738,7 +1738,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. - if (!in_interrupt()) + if (!in_softirq()) fw_iso_context_flush_completions(irq_target->context); } -- 2.30.2

2 1

[PATCH -next] ALSA: n64: check return value after calling platform_get_resource()
by Yang Yingliang 12 Jun '21

12 Jun '21

It will cause null-ptr-deref if platform_get_resource() returns NULL, we need check the return value. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/mips/snd-n64.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index e35e93157755..463a6fe589eb 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -338,6 +338,10 @@ static int __init n64audio_probe(struct platform_device *pdev) strcpy(card->longname, "N64 Audio"); res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + err = -EINVAL; + goto fail_dma_alloc; + } if (devm_request_irq(&pdev->dev, res->start, n64audio_isr, IRQF_SHARED, "N64 Audio", priv)) { err = -EBUSY; -- 2.25.1

2 1

[PATCH -next] ALSA: hda/tegra: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 12 Jun '21

12 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/pci/hda/hda_tegra.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 6f2b743b9d75..5aeef6123781 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -235,11 +235,9 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) { struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); struct hdac_bus *bus = azx_bus(chip); - struct device *dev = hda->dev; struct resource *res; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - hda->regs = devm_ioremap_resource(dev, res); + hda->regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(hda->regs)) return PTR_ERR(hda->regs); -- 2.25.1

2 1

[PATCH -next] ALSA: trident: Fix build error
by YueHaibing 12 Jun '21

12 Jun '21

sound/pci/trident/trident_memory.c: In function ‘set_tlb_bus’: sound/pci/trident/trident_memory.c:85:35: error: ‘pagetr’ undeclared (first use in this function); did you mean ‘page’? for (i = 0; i < UNIT_PAGES; i++, pagetr++) { ^~~~~~ page Fixes: 74fb98311c4e ("ALSA: trident: Drop shadow TLB pointer table") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/trident/trident_memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index f831ec02702f..4ad3855101c9 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -82,7 +82,7 @@ static inline void set_tlb_bus(struct snd_trident *trident, int page, { int i; page *= UNIT_PAGES; - for (i = 0; i < UNIT_PAGES; i++, pagetr++) { + for (i = 0; i < UNIT_PAGES; i++, page++) { __set_tlb_bus(trident, page, addr); addr += SNDRV_TRIDENT_PAGE_SIZE; } -- 2.17.1

2 1

Question about upmix plugin
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-plugins issue #25 was opened from krumpfwylg: Hi, I've noticed the [latest commits](https://github.com/alsa-project/alsa-plugins/pull/2) done to the upmix plugin, but there's something I don't understand. Line 24 of pcm_upmix.c is _#define UPMIX_PCM_FORMAT SND_PCM_FORMAT_S16_ Does that mean the upmix plugin is still only capable of using S16_LE format ? Or does the UPMIX_PCM_FORMAT value can get changed to S32_LE by tweaking it in asoundrc file ? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/25 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

Config parsing of pcm_rate broken since 1.1.7 (at least for a52 pcm plugin)
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #154 was opened from quequotion: As noted [in the mailing list](https://mailman.alsa-project.org/pipermail/alsa-devel/2019-December/1… back in 2019, config parsing (for alsa-plugins a52 pcm at least) has been broken since alsa-lib 1.1.7. As far as I know this was never bisected to a single commit in either alsa-lib or alsa-plugins. Looking at the mailing list discussion, it occurs to me that it may have given the impression that the config parser was broken when using [a patch that ports alsa-plugins to libswresample](https://git.harting.dev/anonfunc/alsa-plugins/commit/9cdbbb9874757b6f8fda7fb4ac2e3fc59da65946). Config parsing is broken *regardless* of that patch. @IdleGandalf worked around this by applying a patch to alsa-lib, in [src/pcm/pcm_rate.c](https://git.harting.dev/anonfunc/alsa-lib/commit/646bc80a63b0220300678a38e0c26a9a13af6fc0), but never got a response if this was an appropriate method to resolve the issue. Should this be fixed in alsa-lib or in alsa-plugins? Note: above links are to mirrors of the original patches, also in a repository owned by @IdleGandalf as far as I can tell. The links in the mailing list discussion are now 404. Issue URL : https://github.com/alsa-project/alsa-lib/issues/154 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

SNDRV_PCM_IOCTL_READI_FRAMES returns EIO
by Muni Sekhar 11 Jun '21

11 Jun '21

Hi all, I am using a USB sound card(snd-usb-audio), and it fails while doing the audio recording "arecord: pcm_read:2032: read error: Input/output error". I noticed recording(arecord) audio data on a USB sound card fails at SNDRV_PCM_IOCTL_READI_FRAMES is giving EIO. Can someone please clarify when the kernel returns EIO for Ioctl SNDRV_PCM_IOCTL_READI_FRAMES? and is there a workaround for this? Output of the alsamixer is attached. -- Thanks, Sekhar

2 5

[PATCH 0/3] ALSA: bebob: remove workarounds for suppressed quirks
by Takashi Sakamoto 11 Jun '21

11 Jun '21

Hi, As a result to apply the commit 1bd1b3be8655 ("ALSA: bebob: perform sequence replay for media clock recovery"), some quirks seem to be suppressed. This patchset removes workarounds for the suppressed quirks and includes code cleanup. Takashi Sakamoto (3): ALSA: bebob: dismiss sleep after breaking connections ALSA: bebob: delete workaround for protocol version 3 ALSA: bebob: code refactoring for model-dependent quirks sound/firewire/bebob/bebob.c | 41 ++++++++++++++++++++--------- sound/firewire/bebob/bebob.h | 10 ++++--- sound/firewire/bebob/bebob_stream.c | 32 ++++++---------------- 3 files changed, 42 insertions(+), 41 deletions(-) -- 2.27.0

2 4

Add support for automatic attributes in Topology2.0
by GitHub pull_request - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-utils pull request #97 was opened from ranj063: Add support for automatic attributes in Topology2.0 and also add support for processing objects that are simply used to update the parent object's private data. Request URL : https://github.com/alsa-project/alsa-utils/pull/97 Patch URL : https://github.com/alsa-project/alsa-utils/pull/97.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - edited 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was edited from Cean1024: alsa info: alsa-config-1.0 alsa-lib-1.1.5 alsa-plugins-1.1.5 alsa-utils-1.1.5 asound.conf: ``` pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 format "S32_LE" channels 2 } bindings { 0 0 1 1 } } ``` test cmd: ``` arecord -f dat | aplay -f dat -D plug:dmixer ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - edited 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was edited from Cean1024: alsa info: alsa-config-1.0 alsa-lib-1.1.5 alsa-plugins-1.1.5 alsa-utils-1.1.5 asound.conf: pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 format "S32_LE" channels 2 } bindings { 0 0 1 1 } } test cmd: arecord -f dat | aplay -f dat -D plug:dmixer Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 11 Jun '21

11 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609012244.24296-1-o-takashi@sakamoc… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v1: * fix context section of kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 71 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 +++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 71 insertions(+), 48 deletions(-) -- 2.27.0

2 16

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was opened from Cean1024: Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq()
by Shengjiu Wang 11 Jun '21

11 Jun '21

On Thu, Jun 10, 2021 at 12:02 PM Zhongjun Tan <hbut_tan(a)163.com> wrote: > > From: Tan Zhongjun <tanzhongjun(a)yulong.com> > > The platform_get_irq() prints error message telling that interrupt is > missing, hence there is no need to duplicated that message. > > Signed-off-by: Tan Zhongjun <tanzhongjun(a)yulong.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> > --- > sound/soc/fsl/fsl_spdif.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c > index 2a76714eb8e6..29cefd459241 100644 > --- a/sound/soc/fsl/fsl_spdif.c > +++ b/sound/soc/fsl/fsl_spdif.c > @@ -1368,10 +1368,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) > > for (i = 0; i < spdif_priv->soc->interrupts; i++) { > irq = platform_get_irq(pdev, i); > - if (irq < 0) { > - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); > + if (irq < 0) > return irq; > - } > > ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, > dev_name(&pdev->dev), spdif_priv); > -- > 2.17.1 >

1 0

Re: [PATCH] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq()
by Shengjiu Wang 11 Jun '21

11 Jun '21

On Thu, Jun 10, 2021 at 8:52 PM Zhongjun Tan <hbut_tan(a)163.com> wrote: > > From: Tan Zhongjun <tanzhongjun(a)yulong.com> > > Clean up the check for irq.dev_err is superfluous as platform_get_irq() > already prints an error.Remove curly braces to confirm to styling > requirements. > > Signed-off-by: Tan Zhongjun <tanzhongjun(a)yulong.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> > --- > sound/soc/fsl/fsl_easrc.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c > index b1765c7d3bcd..25747433916e 100644 > --- a/sound/soc/fsl/fsl_easrc.c > +++ b/sound/soc/fsl/fsl_easrc.c > @@ -1901,10 +1901,8 @@ static int fsl_easrc_probe(struct platform_device *pdev) > } > > irq = platform_get_irq(pdev, 0); > - if (irq < 0) { > - dev_err(dev, "no irq for node %pOF\n", np); > + if (irq < 0) > return irq; > - } > > ret = devm_request_irq(&pdev->dev, irq, fsl_easrc_isr, 0, > dev_name(dev), easrc); > -- > 2.17.1 > >

1 0

[PATCH] ALSA: seq: Fix race of snd_seq_timer_open()
by Takashi Iwai 10 Jun '21

10 Jun '21

The timer instance per queue is exclusive, and snd_seq_timer_open() should have managed the concurrent accesses. It looks as if it's checking the already existing timer instance at the beginning, but it's not right, because there is no protection, hence any later concurrent call of snd_seq_timer_open() may override the timer instance easily. This may result in UAF, as the leftover timer instance can keep running while the queue itself gets closed, as spotted by syzkaller recently. For avoiding the race, add a proper check at the assignment of tmr->timeri again, and return -EBUSY if it's been already registered. Reported-by: syzbot+ddc1260a83ed1cbf6fb5(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000dce34f05c42f110c@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_timer.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 1645e4142e30..9863be6fd43e 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -297,8 +297,16 @@ int snd_seq_timer_open(struct snd_seq_queue *q) return err; } spin_lock_irq(&tmr->lock); - tmr->timeri = t; + if (tmr->timeri) + err = -EBUSY; + else + tmr->timeri = t; spin_unlock_irq(&tmr->lock); + if (err < 0) { + snd_timer_close(t); + snd_timer_instance_free(t); + return err; + } return 0; } -- 2.26.2

1 0

pulseaudio can't load configuration for snd_hda_intel
by GitHub issues - edited 10 Jun '21

10 Jun '21

alsa-project/alsa-ucm-conf issue #99 was edited from hzhuang1: In v1.2.2, the content of HDA-Intel.conf is in below. v1.2.2 is widely used, such as in ubuntu 20.04. ``` Syntax 3 Define.Use "" # a non-empty string to use UCM configuration for HDA devices Define.AcpCardId "$${CardIdByName:acp}" ``` With this content, pulseaudio can't load configuration from UCM. Then I tried to edit the content. Change it to ``` Define.Use "2" ``` or ``` Define.Use "3" ``` Pulseaudio worked with "options snd-hda-intel probe_mask=1" on DELL laptop. My question is what's the real meaning of **Define.Use** at here. Why the default value of **Define.Use** is empty string? Even in latest code, the **Define.Use** is also an empty string. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/99 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

pulseaudio can't load configuration for snd_hda_intel
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-ucm-conf issue #99 was opened from hzhuang1: In v1.2.2, the content of HDA-Intel.conf is in below. v1.2.2 is widely used, such as in ubuntu 20.04. ``` Syntax 3 Define.Use "" # a non-empty string to use UCM configuration for HDA devices Define.AcpCardId "$${CardIdByName:acp}" ``` With this content, pulseaudio can't load configuration from UCM. Then I tried to edit the content. Change it to ``` Define.Use "2" ``` or ``` Define.Use "3" ``` Pulseaudio worked with "options snd-hda-intel probe_mask=1" on DELL laptop. My question is what's the real meaning of **Define.Use** at here. Why the default value of **Define.Use** is empty string? Even in v1.2.4, the **Define.Use** is also an empty string. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/99 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0