May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/7] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 09 May '18

09 May '18

Hi, omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 or OMAP2 for various reasons, but mainly because neither Nokia 770 or Nokia n810 audio probes with linux-next. It does not matter if I have omap-pcm or sdma-pcm... I would be grateful if someone can give tested-by or pointers on how to get 770/n810 audio working. Regards, Peter --- Peter Ujfalusi (7): ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-hdmi-audio: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-dmic: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcpdm: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Mark the omap-pcm as deprecated, but keep it in the tree sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 ++-- sound/soc/omap/Kconfig | 33 ++++++++------- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 5 ++- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/sdma-pcm.c | 68 +++++++++++++++++++++++++++++++ sound/soc/omap/sdma-pcm.h | 21 ++++++++++ 10 files changed, 124 insertions(+), 29 deletions(-) create mode 100644 sound/soc/omap/sdma-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.h -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

4 17

[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

2 104

[alsa-devel] [PATCH 0/5][RFC] ASoC: remove Codec/Platform code
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark Now, all Codec/Platform are replaced to Component. Codec/Platform related code are no longer needed, these patches remove these. But I added [RFC] on these patches because, 1) people want to check/test 2) Media related added new driver is using Codec/Platform style I'm posting replace patch for 2) to Media / ASoC ML, but not yet accepted. (= I think it needs Mark's Acked-by ? I don't know) I think this Codec/Platform remove patch should based on it, otherwise, compile error happen. I'm happy to re-post this remove patch if Media side replace was accepted. Subject: [alsa-devel] [PATCH v3] media: i2c: tda1997: replace codec to component Date: Fri, 6 Apr 2018 05:44:25 +0000 Best regards --- Kuninori Morimoto

3 8

[alsa-devel] Applied "ASoC: remove Codec related code" to the asoc tree
by Mark Brown 09 May '18

09 May '18

The patch ASoC: remove Codec related code has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 999f7f5af8eb7766f68d925a22bf296011abc84c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 8 May 2018 03:20:24 +0000 Subject: [PATCH] ASoC: remove Codec related code Now no one is using Codec related code. Let's remove all Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 265 -------------------------------- sound/soc/soc-core.c | 331 ++-------------------------------------- sound/soc/soc-io.c | 62 -------- sound/soc/soc-jack.c | 22 --- sound/soc/soc-pcm.c | 5 - 6 files changed, 11 insertions(+), 675 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 35ebb0be5114..568f6a72c974 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -313,7 +313,6 @@ struct snd_soc_dai { unsigned int sample_bits; /* parent platform/codec */ - struct snd_soc_codec *codec; struct snd_soc_component *component; /* CODEC TDM slot masks and params (for fixup) */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 131185563532..21861f366dcb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -402,8 +402,6 @@ struct snd_soc_pcm_runtime; struct snd_soc_dai; struct snd_soc_dai_driver; struct snd_soc_dai_link; -struct snd_soc_codec; -struct snd_soc_codec_driver; struct snd_soc_component; struct snd_soc_component_driver; struct soc_enum; @@ -428,13 +426,6 @@ enum snd_soc_card_subclass { SND_SOC_CARD_CLASS_RUNTIME = 1, }; -int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - int source, unsigned int freq, int dir); -int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out); -int snd_soc_codec_set_jack(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, void *data); - int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); @@ -453,10 +444,6 @@ static inline int snd_soc_resume(struct device *dev) } #endif int snd_soc_poweroff(struct device *dev); -int snd_soc_register_codec(struct device *dev, - const struct snd_soc_codec_driver *codec_drv, - struct snd_soc_dai_driver *dai_drv, int num_dai); -void snd_soc_unregister_codec(struct device *dev); int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -559,23 +546,7 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -/* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value); -int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned int reg, unsigned int mask, - unsigned int value); -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value); - #ifdef CONFIG_SND_SOC_AC97_BUS -#define snd_soc_alloc_ac97_codec(codec) \ - snd_soc_alloc_ac97_component(&codec->component) -#define snd_soc_new_ac97_codec(codec, id, id_mask) \ - snd_soc_new_ac97_component(&codec->component, id, id_mask) -#define snd_soc_free_ac97_codec(ac97) \ - snd_soc_free_ac97_component(ac97) - struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); @@ -609,8 +580,6 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); int snd_soc_add_component_controls(struct snd_soc_component *component, const struct snd_kcontrol_new *controls, unsigned int num_controls); -int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_card_controls(struct snd_soc_card *soc_card, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, @@ -843,8 +812,6 @@ struct snd_soc_component { unsigned int active; - unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ - unsigned int registered_as_component:1; unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; @@ -856,9 +823,6 @@ struct snd_soc_component { struct list_head dai_list; int num_dai; - int (*read)(struct snd_soc_component *, unsigned int, unsigned int *); - int (*write)(struct snd_soc_component *, unsigned int, unsigned int); - struct regmap *regmap; int val_bytes; @@ -880,22 +844,6 @@ struct snd_soc_component { /* Don't use these, use snd_soc_component_get_dapm() */ struct snd_soc_dapm_context dapm; - struct snd_soc_codec *codec; - - int (*probe)(struct snd_soc_component *); - void (*remove)(struct snd_soc_component *); - int (*suspend)(struct snd_soc_component *); - int (*resume)(struct snd_soc_component *); - - int (*set_sysclk)(struct snd_soc_component *component, - int clk_id, int source, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_component *component, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out); - int (*set_jack)(struct snd_soc_component *component, - struct snd_soc_jack *jack, void *data); - int (*set_bias_level)(struct snd_soc_component *component, - enum snd_soc_bias_level level); - /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -916,50 +864,6 @@ snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, #define for_each_rtdcom_safe(rtd, rtdcom1, rtdcom2) \ list_for_each_entry_safe(rtdcom1, rtdcom2, &(rtd)->component_list, list) -/* SoC Audio Codec device */ -struct snd_soc_codec { - struct device *dev; - const struct snd_soc_codec_driver *driver; - - struct list_head list; - - /* component */ - struct snd_soc_component component; -}; - -/* codec driver */ -struct snd_soc_codec_driver { - - /* driver ops */ - int (*probe)(struct snd_soc_codec *); - int (*remove)(struct snd_soc_codec *); - int (*suspend)(struct snd_soc_codec *); - int (*resume)(struct snd_soc_codec *); - struct snd_soc_component_driver component_driver; - - /* codec wide operations */ - int (*set_sysclk)(struct snd_soc_codec *codec, - int clk_id, int source, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out); - int (*set_jack)(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, void *data); - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - - /* codec bias level */ - int (*set_bias_level)(struct snd_soc_codec *, - enum snd_soc_bias_level level); - bool idle_bias_off; - bool suspend_bias_off; - - void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); - - bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ -}; - struct snd_soc_dai_link_component { const char *name; struct device_node *of_node; @@ -1213,7 +1117,6 @@ struct snd_soc_pcm_runtime { /* runtime devices */ struct snd_pcm *pcm; struct snd_compr *compr; - struct snd_soc_codec *codec; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; @@ -1281,19 +1184,6 @@ struct soc_enum { struct snd_soc_dobj dobj; }; -/** - * snd_soc_component_to_codec() - Casts a component to the CODEC it is embedded in - * @component: The component to cast to a CODEC - * - * This function must only be used on components that are known to be CODECs. - * Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_component_to_codec( - struct snd_soc_component *component) -{ - return container_of(component, struct snd_soc_codec, component); -} - /** * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is * embedded in @@ -1309,19 +1199,6 @@ static inline struct snd_soc_component *snd_soc_dapm_to_component( return container_of(dapm, struct snd_soc_component, dapm); } -/** - * snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in - * @dapm: The DAPM context to cast to the CODEC - * - * This function must only be used on DAPM contexts that are known to be part of - * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_dapm_to_codec( - struct snd_soc_dapm_context *dapm) -{ - return snd_soc_component_to_codec(snd_soc_dapm_to_component(dapm)); -} - /** * snd_soc_component_get_dapm() - Returns the DAPM context associated with a * component @@ -1333,31 +1210,6 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( return &component->dapm; } -/** - * snd_soc_codec_get_dapm() - Returns the DAPM context for the CODEC - * @codec: The CODEC for which to get the DAPM context - * - * Note: Use this function instead of directly accessing the CODEC's dapm field - */ -static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( - struct snd_soc_codec *codec) -{ - return snd_soc_component_get_dapm(&codec->component); -} - -/** - * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level - * @codec: The CODEC for which to initialize the DAPM bias level - * @level: The DAPM level to initialize to - * - * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level(). - */ -static inline void snd_soc_codec_init_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - snd_soc_dapm_init_bias_level(snd_soc_codec_get_dapm(codec), level); -} - /** * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level * @component: The COMPONENT for which to initialize the DAPM bias level @@ -1373,18 +1225,6 @@ snd_soc_component_init_bias_level(struct snd_soc_component *component, snd_soc_component_get_dapm(component), level); } -/** - * snd_soc_dapm_get_bias_level() - Get current CODEC DAPM bias level - * @codec: The CODEC for which to get the DAPM bias level - * - * Returns: The current DAPM bias level of the CODEC. - */ -static inline enum snd_soc_bias_level snd_soc_codec_get_bias_level( - struct snd_soc_codec *codec) -{ - return snd_soc_dapm_get_bias_level(snd_soc_codec_get_dapm(codec)); -} - /** * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level * @component: The COMPONENT for which to get the DAPM bias level @@ -1398,21 +1238,6 @@ snd_soc_component_get_bias_level(struct snd_soc_component *component) snd_soc_component_get_dapm(component)); } -/** - * snd_soc_codec_force_bias_level() - Set the CODEC DAPM bias level - * @codec: The CODEC for which to set the level - * @level: The level to set to - * - * Forces the CODEC bias level to a specific state. See - * snd_soc_dapm_force_bias_level(). - */ -static inline int snd_soc_codec_force_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - return snd_soc_dapm_force_bias_level(snd_soc_codec_get_dapm(codec), - level); -} - /** * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level * @component: The COMPONENT for which to set the level @@ -1430,19 +1255,6 @@ snd_soc_component_force_bias_level(struct snd_soc_component *component, level); } -/** - * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol - * @kcontrol: The kcontrol - * - * This function must only be used on DAPM contexts that are known to be part of - * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec( - struct snd_kcontrol *kcontrol) -{ - return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); -} - /** * snd_soc_dapm_kcontrol_component() - Returns the component associated to a kcontrol * @kcontrol: The kcontrol @@ -1456,22 +1268,6 @@ static inline struct snd_soc_component *snd_soc_dapm_kcontrol_component( return snd_soc_dapm_to_component(snd_soc_dapm_kcontrol_dapm(kcontrol)); } -/* codec IO */ -unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); -int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val); - -/** - * snd_soc_cache_sync() - Sync the register cache with the hardware - * @codec: CODEC to sync - * - * Note: This function will call regcache_sync() - */ -static inline int snd_soc_cache_sync(struct snd_soc_codec *codec) -{ - return regcache_sync(codec->component.regmap); -} - /** * snd_soc_component_cache_sync() - Sync the register cache with the hardware * @component: COMPONENT to sync @@ -1514,37 +1310,6 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component, struct regmap *regmap); void snd_soc_component_exit_regmap(struct snd_soc_component *component); -/** - * snd_soc_codec_init_regmap() - Initialize regmap instance for the CODEC - * @codec: The CODEC for which to initialize the regmap instance - * @regmap: The regmap instance that should be used by the CODEC - * - * This function allows deferred assignment of the regmap instance that is - * associated with the CODEC. Only use this if the regmap instance is not yet - * ready when the CODEC is registered. The function must also be called before - * the first IO attempt of the CODEC. - */ -static inline void snd_soc_codec_init_regmap(struct snd_soc_codec *codec, - struct regmap *regmap) -{ - snd_soc_component_init_regmap(&codec->component, regmap); -} - -/** - * snd_soc_codec_exit_regmap() - De-initialize regmap instance for the CODEC - * @codec: The CODEC for which to de-initialize the regmap instance - * - * Calls regmap_exit() on the regmap instance associated to the CODEC and - * removes the regmap instance from the CODEC. - * - * This function should only be used if snd_soc_codec_init_regmap() was used to - * initialize the regmap instance. - */ -static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec) -{ - snd_soc_component_exit_regmap(&codec->component); -} - #endif /* device driver data */ @@ -1571,17 +1336,6 @@ static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c) return dev_get_drvdata(c->dev); } -static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, - void *data) -{ - snd_soc_component_set_drvdata(&codec->component, data); -} - -static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec) -{ - return snd_soc_component_get_drvdata(&codec->component); -} - static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) { INIT_LIST_HEAD(&card->widgets); @@ -1633,11 +1387,6 @@ static inline bool snd_soc_component_is_active( return component->active != 0; } -static inline bool snd_soc_codec_is_active(struct snd_soc_codec *codec) -{ - return snd_soc_component_is_active(&codec->component); -} - /** * snd_soc_kcontrol_component() - Returns the component that registered the * control @@ -1654,20 +1403,6 @@ static inline struct snd_soc_component *snd_soc_kcontrol_component( return snd_kcontrol_chip(kcontrol); } -/** - * snd_soc_kcontrol_codec() - Returns the CODEC that registered the control - * @kcontrol: The control for which to get the CODEC - * - * Note: This function will only work correctly if the control has been - * registered with snd_soc_add_codec_controls() or via table based setup of - * snd_soc_codec_driver. Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_kcontrol_codec( - struct snd_kcontrol *kcontrol) -{ - return snd_soc_component_to_codec(snd_soc_kcontrol_component(kcontrol)); -} - int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3b78868969d5..a4cb141b8f38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -56,7 +56,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(codec_list); static LIST_HEAD(component_list); /* @@ -174,21 +173,6 @@ static void soc_cleanup_component_debugfs(struct snd_soc_component *component) debugfs_remove_recursive(component->debugfs_root); } -static int codec_list_show(struct seq_file *m, void *v) -{ - struct snd_soc_codec *codec; - - mutex_lock(&client_mutex); - - list_for_each_entry(codec, &codec_list, list) - seq_printf(m, "%s\n", codec->component.name); - - mutex_unlock(&client_mutex); - - return 0; -} -DEFINE_SHOW_ATTRIBUTE(codec_list); - static int dai_list_show(struct seq_file *m, void *v) { struct snd_soc_component *component; @@ -241,10 +225,6 @@ static void snd_soc_debugfs_init(void) return; } - if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL, - &codec_list_fops)) - pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); - if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL, &dai_list_fops)) pr_warn("ASoC: Failed to create DAI list debugfs file\n"); @@ -536,8 +516,8 @@ int snd_soc_suspend(struct device *dev) } case SND_SOC_BIAS_OFF: - if (component->suspend) - component->suspend(component); + if (component->driver->suspend) + component->driver->suspend(component); component->suspended = 1; if (component->regmap) regcache_mark_dirty(component->regmap); @@ -608,8 +588,8 @@ static void soc_resume_deferred(struct work_struct *work) list_for_each_entry(component, &card->component_dev_list, card_list) { if (component->suspended) { - if (component->resume) - component->resume(component); + if (component->driver->resume) + component->driver->resume(component); component->suspended = 0; } } @@ -892,7 +872,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; - rtd->codec = rtd->codec_dai->codec; /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; @@ -931,8 +910,8 @@ static void soc_remove_component(struct snd_soc_component *component) list_del(&component->card_list); - if (component->remove) - component->remove(component); + if (component->driver->remove) + component->driver->remove(component); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); @@ -1271,8 +1250,8 @@ static int soc_probe_component(struct snd_soc_card *card, } } - if (component->probe) { - ret = component->probe(component); + if (component->driver->probe) { + ret = component->driver->probe(component); if (ret < 0) { dev_err(component->dev, "ASoC: failed to probe component %d\n", ret); @@ -1663,8 +1642,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ /* the component which has non_legacy_dai_naming is Codec */ - if (cpu_dai->codec || - cpu_dai->component->driver->non_legacy_dai_naming) { + if (cpu_dai->component->driver->non_legacy_dai_naming) { unsigned int inv_dai_fmt; inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; @@ -2256,25 +2234,6 @@ int snd_soc_add_component_controls(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_add_component_controls); -/** - * snd_soc_add_codec_controls - add an array of controls to a codec. - * Convenience function to add a list of controls. Many codecs were - * duplicating this code. - * - * @codec: codec to add controls to - * @controls: array of controls to add - * @num_controls: number of elements in the array - * - * Return 0 for success, else error. - */ -int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, unsigned int num_controls) -{ - return snd_soc_add_component_controls(&codec->component, controls, - num_controls); -} -EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); - /** * snd_soc_add_card_controls - add an array of controls to a SoC card. * Convenience function to add a list of controls. @@ -2335,27 +2294,6 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); -/** - * snd_soc_codec_set_sysclk - configure CODEC system or master clock. - * @codec: CODEC - * @clk_id: DAI specific clock ID - * @source: Source for the clock - * @freq: new clock frequency in Hz - * @dir: new clock direction - input/output. - * - * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. - */ -int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - int source, unsigned int freq, int dir) -{ - if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, source, - freq, dir); - else - return -ENOTSUPP; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); - /** * snd_soc_component_set_sysclk - configure COMPONENT system or master clock. * @component: COMPONENT @@ -2369,11 +2307,6 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir) { - /* will be removed */ - if (component->set_sysclk) - return component->set_sysclk(component, clk_id, source, - freq, dir); - if (component->driver->set_sysclk) return component->driver->set_sysclk(component, clk_id, source, freq, dir); @@ -2424,27 +2357,6 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); -/* - * snd_soc_codec_set_pll - configure codec PLL. - * @codec: CODEC - * @pll_id: DAI specific PLL ID - * @source: DAI specific source for the PLL - * @freq_in: PLL input clock frequency in Hz - * @freq_out: requested PLL output clock frequency in Hz - * - * Configures and enables PLL to generate output clock based on input clock. - */ -int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out) -{ - if (codec->driver->set_pll) - return codec->driver->set_pll(codec, pll_id, source, - freq_in, freq_out); - else - return -EINVAL; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); - /* * snd_soc_component_set_pll - configure component PLL. * @component: COMPONENT @@ -2459,11 +2371,6 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - /* will be removed */ - if (component->set_pll) - return component->set_pll(component, pll_id, source, - freq_in, freq_out); - if (component->driver->set_pll) return component->driver->set_pll(component, pll_id, source, freq_in, freq_out); @@ -2964,13 +2871,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; - component->probe = component->driver->probe; - component->remove = component->driver->remove; - component->suspend = component->driver->suspend; - component->resume = component->driver->resume; - component->set_sysclk = component->driver->set_sysclk; - component->set_pll = component->driver->set_pll; - component->set_jack = component->driver->set_jack; dapm = snd_soc_component_get_dapm(component); dapm->dev = dev; @@ -3041,7 +2941,7 @@ EXPORT_SYMBOL_GPL(snd_soc_component_exit_regmap); static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { - if (!component->write && !component->read) { + if (!component->driver->write && !component->driver->read) { if (!component->regmap) component->regmap = dev_get_regmap(component->dev, NULL); if (component->regmap) @@ -3123,9 +3023,6 @@ int snd_soc_add_component(struct device *dev, if (ret) goto err_free; - component->ignore_pmdown_time = true; - component->registered_as_component = true; - if (component_driver->endianness) { for (i = 0; i < num_dai; i++) { convert_endianness_formats(&dai_drv[i].playback); @@ -3180,8 +3077,7 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_lock(&client_mutex); list_for_each_entry(component, &component_list, list) { - if (dev != component->dev || - !component->registered_as_component) + if (dev != component->dev) continue; snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL); @@ -3230,211 +3126,6 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_lookup_component); -static int snd_soc_codec_drv_probe(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->probe(codec); -} - -static void snd_soc_codec_drv_remove(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - codec->driver->remove(codec); -} - -static int snd_soc_codec_drv_suspend(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->suspend(codec); -} - -static int snd_soc_codec_drv_resume(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->resume(codec); -} - -static int snd_soc_codec_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->write(codec, reg, val); -} - -static int snd_soc_codec_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - *val = codec->driver->read(codec, reg); - - return 0; -} - -static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component, - int clk_id, int source, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir); -} - -static int snd_soc_codec_set_pll_(struct snd_soc_component *component, - int pll_id, int source, unsigned int freq_in, - unsigned int freq_out) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out); -} - -static int snd_soc_codec_set_jack_(struct snd_soc_component *component, - struct snd_soc_jack *jack, void *data) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return snd_soc_codec_set_jack(codec, jack, data); -} - -static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); - - return codec->driver->set_bias_level(codec, level); -} - -/** - * snd_soc_register_codec - Register a codec with the ASoC core - * - * @dev: The parent device for this codec - * @codec_drv: Codec driver - * @dai_drv: The associated DAI driver - * @num_dai: Number of DAIs - */ -int snd_soc_register_codec(struct device *dev, - const struct snd_soc_codec_driver *codec_drv, - struct snd_soc_dai_driver *dai_drv, - int num_dai) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - int ret, i; - - dev_dbg(dev, "codec register %s\n", dev_name(dev)); - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - codec->component.codec = codec; - - ret = snd_soc_component_initialize(&codec->component, - &codec_drv->component_driver, dev); - if (ret) - goto err_free; - - if (codec_drv->probe) - codec->component.probe = snd_soc_codec_drv_probe; - if (codec_drv->remove) - codec->component.remove = snd_soc_codec_drv_remove; - if (codec_drv->suspend) - codec->component.suspend = snd_soc_codec_drv_suspend; - if (codec_drv->resume) - codec->component.resume = snd_soc_codec_drv_resume; - if (codec_drv->write) - codec->component.write = snd_soc_codec_drv_write; - if (codec_drv->read) - codec->component.read = snd_soc_codec_drv_read; - if (codec_drv->set_sysclk) - codec->component.set_sysclk = snd_soc_codec_set_sysclk_; - if (codec_drv->set_pll) - codec->component.set_pll = snd_soc_codec_set_pll_; - if (codec_drv->set_jack) - codec->component.set_jack = snd_soc_codec_set_jack_; - codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - - dapm = snd_soc_codec_get_dapm(codec); - dapm->idle_bias_off = codec_drv->idle_bias_off; - dapm->suspend_bias_off = codec_drv->suspend_bias_off; - if (codec_drv->seq_notifier) - dapm->seq_notifier = codec_drv->seq_notifier; - if (codec_drv->set_bias_level) - dapm->set_bias_level = snd_soc_codec_set_bias_level; - codec->dev = dev; - codec->driver = codec_drv; - -#ifdef CONFIG_DEBUG_FS - codec->component.debugfs_prefix = "codec"; -#endif - - for (i = 0; i < num_dai; i++) { - convert_endianness_formats(&dai_drv[i].playback); - convert_endianness_formats(&dai_drv[i].capture); - } - - ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret); - goto err_cleanup; - } - - list_for_each_entry(dai, &codec->component.dai_list, list) - dai->codec = codec; - - mutex_lock(&client_mutex); - snd_soc_component_add_unlocked(&codec->component); - list_add(&codec->list, &codec_list); - mutex_unlock(&client_mutex); - - dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", - codec->component.name); - return 0; - -err_cleanup: - snd_soc_component_cleanup(&codec->component); -err_free: - kfree(codec); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_register_codec); - -/** - * snd_soc_unregister_codec - Unregister a codec from the ASoC core - * - * @dev: codec to unregister - */ -void snd_soc_unregister_codec(struct device *dev) -{ - struct snd_soc_codec *codec; - - mutex_lock(&client_mutex); - list_for_each_entry(codec, &codec_list, list) { - if (dev == codec->dev) - goto found; - } - mutex_unlock(&client_mutex); - return; - -found: - list_del(&codec->list); - snd_soc_component_del_unlocked(&codec->component); - mutex_unlock(&client_mutex); - - dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", - codec->component.name); - - snd_soc_component_cleanup(&codec->component); - kfree(codec); -} -EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); - /* Retrieve a card's name from device tree */ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c92a04bac3c5..026cd5347e53 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -32,8 +32,6 @@ int snd_soc_component_read(struct snd_soc_component *component, if (component->regmap) ret = regmap_read(component->regmap, reg, val); - else if (component->read) - ret = component->read(component, reg, val); else if (component->driver->read) { *val = component->driver->read(component, reg); ret = 0; @@ -72,8 +70,6 @@ int snd_soc_component_write(struct snd_soc_component *component, { if (component->regmap) return regmap_write(component->regmap, reg, val); - else if (component->write) - return component->write(component, reg, val); else if (component->driver->write) return component->driver->write(component, reg, val); else @@ -209,61 +205,3 @@ int snd_soc_component_test_bits(struct snd_soc_component *component, return old != new; } EXPORT_SYMBOL_GPL(snd_soc_component_test_bits); - -unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg) -{ - unsigned int val; - int ret; - - ret = snd_soc_component_read(&codec->component, reg, &val); - if (ret < 0) - return -1; - - return val; -} -EXPORT_SYMBOL_GPL(snd_soc_read); - -int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - return snd_soc_component_write(&codec->component, reg, val); -} -EXPORT_SYMBOL_GPL(snd_soc_write); - -/** - * snd_soc_update_bits - update codec register bits - * @codec: audio codec - * @reg: codec register - * @mask: register mask - * @value: new value - * - * Writes new register value. - * - * Returns 1 for change, 0 for no change, or negative error code. - */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value) -{ - return snd_soc_component_update_bits(&codec->component, reg, mask, - value); -} -EXPORT_SYMBOL_GPL(snd_soc_update_bits); - -/** - * snd_soc_test_bits - test register for change - * @codec: audio codec - * @reg: codec register - * @mask: register mask - * @value: new value - * - * Tests a register with a new value and checks if the new value is - * different from the old value. - * - * Returns 1 for change else 0. - */ -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value) -{ - return snd_soc_component_test_bits(&codec->component, reg, mask, value); -} -EXPORT_SYMBOL_GPL(snd_soc_test_bits); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99902ae1a2d9..b2b16044ae80 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -28,24 +28,6 @@ struct jack_gpio_tbl { struct snd_soc_jack_gpio *gpios; }; -/** - * snd_soc_codec_set_jack - configure codec jack. - * @codec: CODEC - * @jack: structure to use for the jack - * @data: can be used if codec driver need extra data for configuring jack - * - * Configures and enables jack detection function. - */ -int snd_soc_codec_set_jack(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, void *data) -{ - if (codec->driver->set_jack) - return codec->driver->set_jack(codec, jack, data); - else - return -ENOTSUPP; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); - /** * snd_soc_component_set_jack - configure component jack. * @component: COMPONENTs @@ -57,10 +39,6 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); int snd_soc_component_set_jack(struct snd_soc_component *component, struct snd_soc_jack *jack, void *data) { - /* will be removed */ - if (component->set_jack) - return component->set_jack(component, jack, data); - if (component->driver->set_jack) return component->driver->set_jack(component, jack, data); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3f6375499102..87c9af2158d0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -135,7 +135,6 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; - int i; bool ignore = true; if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) @@ -147,10 +146,6 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) ignore &= !component->driver->use_pmdown_time; } - /* this will be removed */ - for (i = 0; i < rtd->num_codecs; i++) - ignore &= rtd->codec_dais[i]->component->ignore_pmdown_time; - return ignore; } -- 2.17.0

1 0

[alsa-devel] Applied "ASoC: convert platform explanation to component" to the asoc tree
by Mark Brown 09 May '18

09 May '18

The patch ASoC: convert platform explanation to component has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d9f29e9ebe6ec7193c160ac3d38d5d1d6b55adda Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 8 May 2018 03:21:24 +0000 Subject: [PATCH] ASoC: convert platform explanation to component commit ef050bece1b55 ("ASoC: Remove platform code now everything is componentised") removed platform code, but it didn't care about platform documentation. This patch convert platform explanation to component Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/soc/platform.rst | 30 ++++++++++++---------------- 1 file changed, 13 insertions(+), 17 deletions(-) diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst index d5574904d981..cc817078daa6 100644 --- a/Documentation/sound/soc/platform.rst +++ b/Documentation/sound/soc/platform.rst @@ -23,30 +23,26 @@ The platform DMA driver optionally supports the following ALSA operations:- }; The platform driver exports its DMA functionality via struct -snd_soc_platform_driver:- +snd_soc_component_driver:- :: - struct snd_soc_platform_driver { - char *name; + struct snd_soc_component_driver { + const char *name; - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); - int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + ... + int (*probe)(struct snd_soc_component *); + void (*remove)(struct snd_soc_component *); + int (*suspend)(struct snd_soc_component *); + int (*resume)(struct snd_soc_component *); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); + int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); - /* - * For platform caused delay reporting. - * Optional. - */ - snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, - struct snd_soc_dai *); - - /* platform stream ops */ - struct snd_pcm_ops *pcm_ops; + ... + const struct snd_pcm_ops *ops; + const struct snd_compr_ops *compr_ops; + ... }; Please refer to the ALSA driver documentation for details of audio DMA. -- 2.17.0

1 0

[alsa-devel] [PATCH 00/10] ASoC: remove old platform/codec code
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark We are now using component base ASoC. Thus, old platform/codec code are no longer needed. These removes it. Then, I very surprised that platform code was already removed :) But, it seems this patch didn't care about documentation and platform specific callback (= .pcm_new/free). 4), 7) handle it. One concern is that it completely removed "platform_list_show" for debugfs. But, I think we want to keep it as "component_list_show" (My original patch did it). 6) reborned it. 8) - 10) are cleanup which can be done *after* removing platform/codec. Kuninori Morimoto (10): 1) ASoC: remove codec hw_write/control_data 2) ASoC: remove codec reg_cache 3) ASoC: remove .get_regmap 4) ASoC: remove unneeded .pcm_new/free 5) ASoC: remove Codec related code 6) ASoC: add component_list_show() 7) ASoC: convert platform explanation to component 8) ASoC: soc.h: merge CONFIG_DEBUG_FS 9) ASoC: soc-core: remove snd_soc_component_add_unlocked() 10) ASoC: soc-core: remove legacy_dai_naming from snd_soc_register_dais() Documentation/sound/soc/platform.rst | 30 +- include/sound/soc-dai.h | 1 - include/sound/soc.h | 289 +---------------- sound/soc/Makefile | 2 +- sound/soc/soc-cache.c | 53 ---- sound/soc/soc-core.c | 585 ++--------------------------------- sound/soc/soc-io.c | 62 ---- sound/soc/soc-jack.c | 22 -- sound/soc/soc-pcm.c | 13 +- 9 files changed, 50 insertions(+), 1007 deletions(-) delete mode 100644 sound/soc/soc-cache.c Best regards --- Kuninori Morimoto

3 25

[alsa-devel] [PATCH] ASoC: soc-core: remove legacy_dai_naming from snd_soc_register_dais()
by Kuninori Morimoto 09 May '18

09 May '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We can get legacy dai name flag from component driver. Thus, there is no need to have its parameter on snd_soc_register_dais(). Let's remove unneeded parameter Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1faebc04..9c65477 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3146,8 +3146,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count, - bool legacy_dai_naming) + struct snd_soc_dai_driver *dai_drv, size_t count) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -3159,7 +3158,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, for (i = 0; i < count; i++) { dai = soc_add_dai(component, dai_drv + i, - count == 1 && legacy_dai_naming); + count == 1 && !component->driver->non_legacy_dai_naming); if (dai == NULL) { ret = -ENOMEM; goto err; @@ -3440,8 +3439,7 @@ int snd_soc_add_component(struct device *dev, } } - ret = snd_soc_register_dais(component, dai_drv, num_dai, - !component_driver->non_legacy_dai_naming); + ret = snd_soc_register_dais(component, dai_drv, num_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret); goto err_cleanup; -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: uniphier: add digital output volume for UniPhier sound system
by Katsuhiro Suzuki 09 May '18

09 May '18

This patch adds controllers for digital volume of PCM output. Volume effects simply linear, not dB scale as follows: Gained PCM = Original * 0x4000 / Volume The value range of volume is from 0x0001 to 0xffff. 0x0000 works as mute. Initial value is 0x4000 (+0dB). Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 58 ++++++++++++++ sound/soc/uniphier/aio-cpu.c | 140 ++++++++++++++++++++++++++++++++++ sound/soc/uniphier/aio-reg.h | 33 ++++++-- sound/soc/uniphier/aio.h | 7 ++ 4 files changed, 233 insertions(+), 5 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index e37b80921abb..638cb3fc5f7b 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -659,6 +659,64 @@ void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable) } } +/** + * aio_port_get_volume - get volume of AIO port block + * @sub: the AIO substream pointer + * + * Return: current volume, range is 0x0000 - 0xffff + */ +int aio_port_get_volume(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 v; + + regmap_read(r, OPORTMXTYVOLGAINSTATUS(sub->swm->oport.map, 0), &v); + + return FIELD_GET(OPORTMXTYVOLGAINSTATUS_CUR_MASK, v); +} + +/** + * aio_port_set_volume - set volume of AIO port block + * @sub: the AIO substream pointer + * @vol: target volume, range is 0x0000 - 0xffff. + * + * Change digital volume and perfome fade-out/fade-in effect for specified + * output slot of port. Gained PCM value can calculate as the following: + * Gained = Original * vol / 0x4000 + */ +void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) +{ + struct regmap *r = sub->aio->chip->regmap; + int oport_map = sub->swm->oport.map; + int cur, diff, slope = 0, fs; + + if (sub->swm->dir == PORT_DIR_INPUT) + return; + + cur = aio_port_get_volume(sub); + diff = abs(vol - cur); + fs = params_rate(&sub->params); + if (fs) + slope = diff / AUD_VOL_FADE_TIME * 1000 / fs; + slope = max(1, slope); + + regmap_update_bits(r, OPORTMXTYVOLPARA1(oport_map, 0), + OPORTMXTYVOLPARA1_SLOPEU_MASK, slope << 16); + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_TARGET_MASK, vol); + + if (cur < vol) + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_FADE_MASK, + OPORTMXTYVOLPARA2_FADE_FADEIN); + else + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_FADE_MASK, + OPORTMXTYVOLPARA2_FADE_FADEOUT); + + regmap_write(r, AOUTFADECTR0, BIT(oport_map)); +} + /** * aio_if_set_param - set parameters of AIO DMA I/F block * @sub: the AIO substream pointer diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 00b6441bf195..80daec17be25 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -32,6 +32,35 @@ static bool is_valid_pll(struct uniphier_aio_chip *chip, int pll_id) return chip->plls[pll_id].enable; } +/** + * find_volume - find volume supported HW port by HW port number + * @chip: the AIO chip pointer + * @oport_hw: HW port number, one of AUD_HW_XXXX + * + * Find AIO device from device list by HW port number. Volume feature is + * available only in Output and PCM ports, this limitation comes from HW + * specifications. + * + * Return: The pointer of AIO substream if successful, otherwise NULL on error. + */ +static struct uniphier_aio_sub *find_volume(struct uniphier_aio_chip *chip, + int oport_hw) +{ + int i; + + for (i = 0; i < chip->num_aios; i++) { + struct uniphier_aio_sub *sub = &chip->aios[i].sub[0]; + + if (!sub->swm) + continue; + + if (sub->swm->oport.hw == oport_hw) + return sub; + } + + return NULL; +} + static bool match_spec(const struct uniphier_aio_spec *spec, const char *name, int dir) { @@ -287,6 +316,7 @@ static int uniphier_aio_hw_params(struct snd_pcm_substream *substream, sub->setting = 1; aio_port_reset(sub); + aio_port_set_volume(sub, sub->vol); aio_src_reset(sub); return 0; @@ -373,6 +403,8 @@ int uniphier_aio_dai_probe(struct snd_soc_dai *dai) sub->swm = &spec->swm; sub->spec = spec; + + sub->vol = AUD_VOL_INIT; } aio_iecout_set_enable(aio->chip, true); @@ -449,8 +481,116 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) } EXPORT_SYMBOL_GPL(uniphier_aio_dai_resume); +static int uniphier_aio_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = AUD_VOL_MAX; + + return 0; +} + +static int uniphier_aio_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct uniphier_aio_chip *chip = snd_soc_component_get_drvdata(comp); + struct uniphier_aio_sub *sub; + int oport_hw = kcontrol->private_value; + + sub = find_volume(chip, oport_hw); + if (!sub) + return 0; + + ucontrol->value.integer.value[0] = sub->vol; + + return 0; +} + +static int uniphier_aio_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct uniphier_aio_chip *chip = snd_soc_component_get_drvdata(comp); + struct uniphier_aio_sub *sub; + int oport_hw = kcontrol->private_value; + + sub = find_volume(chip, oport_hw); + if (!sub) + return 0; + + if (sub->vol == ucontrol->value.integer.value[0]) + return 0; + sub->vol = ucontrol->value.integer.value[0]; + + aio_port_set_volume(sub, sub->vol); + + return 0; +} + +static const struct snd_kcontrol_new uniphier_aio_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "HPCMOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_HPCMOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT2 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT2, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT3 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT3, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "HIECOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_HIECOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "IECOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_IECOUT1, + }, +}; + static const struct snd_soc_component_driver uniphier_aio_component = { .name = "uniphier-aio", + .controls = uniphier_aio_controls, + .num_controls = ARRAY_SIZE(uniphier_aio_controls), }; int uniphier_aio_probe(struct platform_device *pdev) diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 511ea3c01847..45fdc6ae358a 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -169,6 +169,7 @@ #define PBINMXPAUSECTR1(n) (0x20208 + 0x40 * (n)) /* AOUT */ +#define AOUTFADECTR0 0x40020 #define AOUTENCTR0 0x40040 #define AOUTENCTR1 0x40044 #define AOUTENCTR2 0x40048 @@ -179,6 +180,9 @@ #define AOUTSRCRSTCTR1 0x400c4 #define AOUTSRCRSTCTR2 0x400c8 +/* AOUT PCMOUT has 5 slots, slot0-3: D0-3, slot4: DMIX */ +#define OPORT_SLOT_MAX 5 + /* AOUT(PCMOUTN) */ #define OPORTMXCTR1(n) (0x42000 + 0x400 * (n)) #define OPORTMXCTR1_I2SLRSEL_MASK (0x11 << 10) @@ -359,11 +363,30 @@ #define OPORTMXMASK_XCKMSK_ON (0x0 << 0) #define OPORTMXMASK_XCKMSK_OFF (0x7 << 0) #define OPORTMXDEBUG(n) (0x420fc + 0x400 * (n)) -#define OPORTMXT0RSTCTR(n) (0x4211c + 0x400 * (n)) -#define OPORTMXT1RSTCTR(n) (0x4213c + 0x400 * (n)) -#define OPORTMXT2RSTCTR(n) (0x4215c + 0x400 * (n)) -#define OPORTMXT3RSTCTR(n) (0x4217c + 0x400 * (n)) -#define OPORTMXT4RSTCTR(n) (0x4219c + 0x400 * (n)) +#define OPORTMXTYVOLPARA1(n, m) (0x42100 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLPARA1_SLOPEU_MASK GENMASK(31, 16) +#define OPORTMXTYVOLPARA2(n, m) (0x42104 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLPARA2_FADE_MASK GENMASK(17, 16) +#define OPORTMXTYVOLPARA2_FADE_NOOP (0x0 << 16) +#define OPORTMXTYVOLPARA2_FADE_FADEOUT (0x1 << 16) +#define OPORTMXTYVOLPARA2_FADE_FADEIN (0x2 << 16) +#define OPORTMXTYVOLPARA2_TARGET_MASK GENMASK(15, 0) +#define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) +#define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT2 (0xa << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT3 (0xb << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT4 (0xc << 8) +#define OPORTMXT0SLOTCTR_MUTEOFF_MASK BIT(1) +#define OPORTMXT0SLOTCTR_MUTEOFF_MUTE (0x0 << 1) +#define OPORTMXT0SLOTCTR_MUTEOFF_UNMUTE (0x1 << 1) +#define OPORTMXTYRSTCTR(n, m) (0x4211c + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXT0RSTCTR_RST_MASK BIT(1) +#define OPORTMXT0RSTCTR_RST_OFF (0x0 << 1) +#define OPORTMXT0RSTCTR_RST_ON (0x1 << 1) #define SBF_(frame, shift) (((frame) * 2 - 1) << shift) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 52670126084f..aa89c2f6fa24 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -130,6 +130,10 @@ enum IEC61937_PC { #define AUD_PLLDIV_1_1 2 #define AUD_PLLDIV_2_3 3 +#define AUD_VOL_INIT 0x4000 /* +0dB */ +#define AUD_VOL_MAX 0xffff /* +6dB */ +#define AUD_VOL_FADE_TIME 20 /* 20ms */ + #define AUD_RING_SIZE (128 * 1024) #define AUD_MIN_FRAGMENT 4 @@ -231,6 +235,7 @@ struct uniphier_aio_sub { /* For PCM audio */ struct snd_pcm_substream *substream; struct snd_pcm_hw_params params; + int vol; /* For compress audio */ struct snd_compr_stream *cstream; @@ -323,6 +328,8 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); +int aio_port_get_volume(struct uniphier_aio_sub *sub); +void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol); int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through); int aio_oport_set_stream_type(struct uniphier_aio_sub *sub, enum IEC61937_PC pc); -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: nau8824: fix spelling mistake: "semaphone" -> "semaphore"
by Colin King 09 May '18

09 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_warn messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 76502c090654..6bd14453f06e 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -205,11 +205,11 @@ static int nau8824_sema_acquire(struct nau8824 *nau8824, long timeout) if (timeout) { ret = down_timeout(&nau8824->jd_sem, timeout); if (ret < 0) - dev_warn(nau8824->dev, "Acquire semaphone timeout\n"); + dev_warn(nau8824->dev, "Acquire semaphore timeout\n"); } else { ret = down_interruptible(&nau8824->jd_sem); if (ret < 0) - dev_warn(nau8824->dev, "Acquire semaphone fail\n"); + dev_warn(nau8824->dev, "Acquire semaphore fail\n"); } return ret; -- 2.17.0

2 2

[alsa-devel] [PATCH] ASoC: uniphier: evea: use DAPM to change source of line-in
by Katsuhiro Suzuki 09 May '18

09 May '18

This patch replaces mixer switch to DAPM one for changing audio source of line-in. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 55 +++++++++++++++++++++++---------------- 1 file changed, 33 insertions(+), 22 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index 73fd6730095c..f9c10165fbc1 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -54,8 +54,21 @@ struct evea_priv { int switch_hp; }; +static const char * const linsw1_sel1_text[] = { + "LIN1", "LIN2", "LIN3" +}; + +static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, + ALINSW1, ALINSW1_SEL1_SHIFT, + linsw1_sel1_text); + +static const struct snd_kcontrol_new linesw1_mux[] = { + SOC_DAPM_ENUM("Line In 1 Source", linsw1_sel1_enum), +}; + static const struct snd_soc_dapm_widget evea_widgets[] = { - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("Line In 1 Mux", SND_SOC_NOPM, 0, 0, linesw1_mux), SND_SOC_DAPM_INPUT("LIN1_LP"), SND_SOC_DAPM_INPUT("LIN1_RP"), SND_SOC_DAPM_INPUT("LIN2_LP"), @@ -63,7 +76,9 @@ static const struct snd_soc_dapm_widget evea_widgets[] = { SND_SOC_DAPM_INPUT("LIN3_LP"), SND_SOC_DAPM_INPUT("LIN3_RP"), - SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC HP", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC LO1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC LO2", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUTPUT("HP1_L"), SND_SOC_DAPM_OUTPUT("HP1_R"), SND_SOC_DAPM_OUTPUT("LO2_L"), @@ -71,17 +86,22 @@ static const struct snd_soc_dapm_widget evea_widgets[] = { }; static const struct snd_soc_dapm_route evea_routes[] = { - { "ADC", NULL, "LIN1_LP" }, - { "ADC", NULL, "LIN1_RP" }, - { "ADC", NULL, "LIN2_LP" }, - { "ADC", NULL, "LIN2_RP" }, - { "ADC", NULL, "LIN3_LP" }, - { "ADC", NULL, "LIN3_RP" }, - - { "HP1_L", NULL, "DAC" }, - { "HP1_R", NULL, "DAC" }, - { "LO2_L", NULL, "DAC" }, - { "LO2_R", NULL, "DAC" }, + { "Line In 1", NULL, "ADC" }, + { "ADC", NULL, "Line In 1 Mux" }, + { "Line In 1 Mux", "LIN1", "LIN1_LP" }, + { "Line In 1 Mux", "LIN1", "LIN1_RP" }, + { "Line In 1 Mux", "LIN2", "LIN2_LP" }, + { "Line In 1 Mux", "LIN2", "LIN2_RP" }, + { "Line In 1 Mux", "LIN3", "LIN3_LP" }, + { "Line In 1 Mux", "LIN3", "LIN3_RP" }, + + { "DAC HP", NULL, "Headphone 1" }, + { "DAC LO1", NULL, "Line Out 1" }, + { "DAC LO2", NULL, "Line Out 2" }, + { "HP1_L", NULL, "DAC HP" }, + { "HP1_R", NULL, "DAC HP" }, + { "LO2_L", NULL, "DAC LO2" }, + { "LO2_R", NULL, "DAC LO2" }, }; static void evea_set_power_state_on(struct evea_priv *evea) @@ -280,16 +300,7 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } -static const char * const linsw1_sel1_text[] = { - "LIN1", "LIN2", "LIN3" -}; - -static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, - ALINSW1, ALINSW1_SEL1_SHIFT, - linsw1_sel1_text); - static const struct snd_kcontrol_new evea_controls[] = { - SOC_ENUM("Line Capture Source", linsw1_sel1_enum), SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, -- 2.17.0

2 2