From 81c7cfd1b22a0ee5e40efef72ec2cd17dbf12e6d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:18 +0200 Subject: ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 20 ++++++--- sound/soc/soc-core.c | 122 ++++++++++++++++++++++----------------------------- 2 files changed, 67 insertions(+), 75 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index be6ecae247b0..0ab8b1e4a5d2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -728,9 +728,24 @@ struct snd_soc_component { struct mutex io_mutex; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_root; +#endif + + /* + * DO NOT use any of the fields below in drivers, they are temporary and + * are going to be removed again soon. If you use them in driver code the + * driver will be marked as BROKEN when these fields are removed. + */ + /* Don't use these, use snd_soc_component_get_dapm() */ struct snd_soc_dapm_context dapm; struct snd_soc_dapm_context *dapm_ptr; + +#ifdef CONFIG_DEBUG_FS + void (*init_debugfs)(struct snd_soc_component *component); + const char *debugfs_prefix; +#endif }; /* SoC Audio Codec device */ @@ -766,7 +781,6 @@ struct snd_soc_codec { struct snd_soc_dapm_context dapm; #ifdef CONFIG_DEBUG_FS - struct dentry *debugfs_codec_root; struct dentry *debugfs_reg; #endif }; @@ -879,10 +893,6 @@ struct snd_soc_platform { struct list_head list; struct snd_soc_component component; - -#ifdef CONFIG_DEBUG_FS - struct dentry *debugfs_platform_root; -#endif }; struct snd_soc_dai_link { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a9076f..79371a77f324 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,79 +270,56 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; -static struct dentry *soc_debugfs_create_dir(struct dentry *parent, - const char *fmt, ...) +static void soc_init_component_debugfs(struct snd_soc_component *component) { - struct dentry *de; - va_list ap; - char *s; + if (component->debugfs_prefix) { + char *name; - va_start(ap, fmt); - s = kvasprintf(GFP_KERNEL, fmt, ap); - va_end(ap); + name = kasprintf(GFP_KERNEL, "%s:%s", + component->debugfs_prefix, component->name); + if (name) { + component->debugfs_root = debugfs_create_dir(name, + component->card->debugfs_card_root); + kfree(name); + } + } else { + component->debugfs_root = debugfs_create_dir(component->name, + component->card->debugfs_card_root); + } - if (!s) - return NULL; + if (!component->debugfs_root) { + dev_warn(component->dev, + "ASoC: Failed to create component debugfs directory\n"); + return; + } - de = debugfs_create_dir(s, parent); - kfree(s); + snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component), + component->debugfs_root); - return de; + if (component->init_debugfs) + component->init_debugfs(component); } -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +static void soc_cleanup_component_debugfs(struct snd_soc_component *component) { - struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; + debugfs_remove_recursive(component->debugfs_root); +} - codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, - "codec:%s", - codec->component.name); - if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, - "ASoC: Failed to create codec debugfs directory\n"); - return; - } +static void soc_init_codec_debugfs(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->debugfs_codec_root, + codec->component.debugfs_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); - - snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_codec_root); -} - -static void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ - struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; - - platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, - "platform:%s", - platform->component.name); - if (!platform->debugfs_platform_root) { - dev_warn(platform->dev, - "ASoC: Failed to create platform debugfs directory\n"); - return; - } - - snd_soc_dapm_debugfs_init(&platform->component.dapm, - platform->debugfs_platform_root); -} - -static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) -{ - debugfs_remove_recursive(platform->debugfs_platform_root); } static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, @@ -474,19 +451,15 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) #else -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} +#define soc_init_codec_debugfs NULL -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +static inline void soc_init_component_debugfs( + struct snd_soc_component *component) { } -static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ -} - -static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_cleanup_component_debugfs( + struct snd_soc_component *component) { } @@ -1026,7 +999,7 @@ static int soc_remove_platform(struct snd_soc_platform *platform) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&platform->component.dapm); - soc_cleanup_platform_debugfs(platform); + soc_cleanup_component_debugfs(&platform->component); platform->probed = 0; module_put(platform->dev->driver->owner); @@ -1046,7 +1019,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&codec->dapm); - soc_cleanup_codec_debugfs(codec); + soc_cleanup_component_debugfs(&codec->component); codec->probed = 0; list_del(&codec->card_list); module_put(codec->dev->driver->owner); @@ -1187,7 +1160,7 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - soc_init_codec_debugfs(codec); + soc_init_component_debugfs(&codec->component); if (driver->dapm_widgets) { ret = snd_soc_dapm_new_controls(&codec->dapm, @@ -1242,7 +1215,7 @@ static int soc_probe_codec(struct snd_soc_card *card, return 0; err_probe: - soc_cleanup_codec_debugfs(codec); + soc_cleanup_component_debugfs(&codec->component); module_put(codec->dev->driver->owner); return ret; @@ -1262,7 +1235,7 @@ static int soc_probe_platform(struct snd_soc_card *card, if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; - soc_init_platform_debugfs(platform); + soc_init_component_debugfs(&platform->component); if (driver->dapm_widgets) snd_soc_dapm_new_controls(&platform->component.dapm, @@ -1302,7 +1275,7 @@ static int soc_probe_platform(struct snd_soc_card *card, return 0; err_probe: - soc_cleanup_platform_debugfs(platform); + soc_cleanup_component_debugfs(&platform->component); module_put(platform->dev->driver->owner); return ret; @@ -4266,6 +4239,10 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, if (platform_drv->read) platform->component.read = snd_soc_platform_drv_read; +#ifdef CONFIG_DEBUG_FS + platform->component.debugfs_prefix = "platform"; +#endif + mutex_lock(&client_mutex); snd_soc_component_add_unlocked(&platform->component); list_add(&platform->list, &platform_list); @@ -4455,6 +4432,11 @@ int snd_soc_register_codec(struct device *dev, codec->component.val_bytes = codec_drv->reg_word_size; mutex_init(&codec->mutex); +#ifdef CONFIG_DEBUG_FS + codec->component.init_debugfs = soc_init_codec_debugfs; + codec->component.debugfs_prefix = "codec"; +#endif + if (!codec->component.write) { if (codec_drv->get_regmap) regmap = codec_drv->get_regmap(dev); -- cgit v1.2.3 From f1d45cc3ae96a6173129b2c164c216272faa5fc0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:19 +0200 Subject: ASoC: Consolidate platform and CODEC probe/remove The platform and CODEC probe and remove code is now largely identical. This patch consolidates it at the component level. The resulting code is slightly larger due to all the boiler plate code setting up the indirection for the table based control and DAPM registration. Once all drivers have been update to no longer use the snd_soc_codec_driver and snd_soc_platform_driver specific fields for this the indirection can be removed again. This patch contains two noteworthy hacks that are only meant to be temporary to be able to update drivers and the core in separate incremental patches. The first hack is related to that some DPCM platforms expect that the DAPM widgets for the DAIs of a snd_soc_component are created in the DAPM context of the snd_soc_platform that has the same parent device. For handling this the steal_sibling_dai_widgets attribute is introduced. It gets set for snd_soc_platforms that register DAPM elements. When creating the DAI widgets for a component this flag is checked and if it is found on one of the siblings the component will not create any DAI widgets in its own DAPM context. If the attribute is set on a platform it will look for siblings components and create DAI widgets for them in its own context. The fix for this will be to update the offending drivers to only register a single component rather than two. The second hack deals with the fact that the ASoC card suspend and resume code still needs a list of CODECs that have been registered for the card. To handle this the generic probe and remove path have a check to see if the component is CODEC and if yes add/remove it to the card's CODEC list. While it is possible to clean up the suspend/resume code to not need the CODEC list anymore this is a bit of a chicken and egg problem since it will become easier to clean up the suspend/resume code once there is a unified component layer. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 27 ++- sound/soc/soc-core.c | 335 ++++++++++++++++++---------------- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- 3 files changed, 194 insertions(+), 172 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0ab8b1e4a5d2..22543acfae4b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -697,6 +697,10 @@ struct snd_soc_component_driver { void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, int subseq); int (*stream_event)(struct snd_soc_component *, int event); + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; struct snd_soc_component { @@ -710,6 +714,7 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; + unsigned int probed:1; struct list_head list; @@ -742,6 +747,18 @@ struct snd_soc_component { struct snd_soc_dapm_context dapm; struct snd_soc_dapm_context *dapm_ptr; + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + unsigned int num_dapm_routes; + bool steal_sibling_dai_widgets; + struct snd_soc_codec *codec; + + int (*probe)(struct snd_soc_component *); + void (*remove)(struct snd_soc_component *); + #ifdef CONFIG_DEBUG_FS void (*init_debugfs)(struct snd_soc_component *component); const char *debugfs_prefix; @@ -761,7 +778,6 @@ struct snd_soc_codec { struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int suspended:1; /* Codec is in suspend PM state */ - unsigned int probed:1; /* Codec has been probed */ unsigned int ac97_registered:1; /* Codec has been AC97 registered */ unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int cache_init:1; /* codec cache has been initialized */ @@ -827,10 +843,6 @@ struct snd_soc_codec_driver { enum snd_soc_dapm_type, int); bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ - - /* probe ordering - for components with runtime dependencies */ - int probe_order; - int remove_order; }; /* SoC platform interface */ @@ -867,10 +879,6 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; - /* probe ordering - for components with runtime dependencies */ - int probe_order; - int remove_order; - /* platform IO - used for platform DAPM */ unsigned int (*read)(struct snd_soc_platform *, unsigned int); int (*write)(struct snd_soc_platform *, unsigned int, unsigned int); @@ -888,7 +896,6 @@ struct snd_soc_platform { const struct snd_soc_platform_driver *driver; unsigned int suspended:1; /* platform is suspended */ - unsigned int probed:1; struct list_head list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 79371a77f324..b833cc6fd86d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -985,44 +985,20 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } -static int soc_remove_platform(struct snd_soc_platform *platform) +static void soc_remove_component(struct snd_soc_component *component) { - int ret; - - if (platform->driver->remove) { - ret = platform->driver->remove(platform); - if (ret < 0) - dev_err(platform->dev, "ASoC: failed to remove %d\n", - ret); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->component.dapm); - - soc_cleanup_component_debugfs(&platform->component); - platform->probed = 0; - module_put(platform->dev->driver->owner); - - return 0; -} - -static void soc_remove_codec(struct snd_soc_codec *codec) -{ - int err; + /* This is a HACK and will be removed soon */ + if (component->codec) + list_del(&component->codec->card_list); - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - dev_err(codec->dev, "ASoC: failed to remove %d\n", err); - } + if (component->remove) + component->remove(component); - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); + snd_soc_dapm_free(snd_soc_component_get_dapm(component)); - soc_cleanup_component_debugfs(&codec->component); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); + soc_cleanup_component_debugfs(component); + component->probed = 0; + module_put(component->dev->driver->owner); } static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) @@ -1086,25 +1062,24 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, int i; /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - soc_remove_platform(platform); - } + if (platform && platform->component.probed && + platform->component.driver->remove_order == order) + soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { codec = rtd->codec_dais[i]->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (codec && codec->component.probed && + codec->component.driver->remove_order == order) + soc_remove_component(&codec->component); } /* remove any CPU-side CODEC */ if (cpu_dai) { codec = cpu_dai->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (codec && codec->component.probed && + codec->component.driver->remove_order == order) + soc_remove_component(&codec->component); } } @@ -1146,137 +1121,108 @@ static void soc_set_name_prefix(struct snd_soc_card *card, } } -static int soc_probe_codec(struct snd_soc_card *card, - struct snd_soc_codec *codec) +static int soc_probe_component(struct snd_soc_card *card, + struct snd_soc_component *component) { - int ret = 0; - const struct snd_soc_codec_driver *driver = codec->driver; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct snd_soc_component *dai_component, *component2; struct snd_soc_dai *dai; + int ret; - codec->component.card = card; - codec->dapm.card = card; - soc_set_name_prefix(card, &codec->component); + component->card = card; + dapm->card = card; + soc_set_name_prefix(card, component); - if (!try_module_get(codec->dev->driver->owner)) + if (!try_module_get(component->dev->driver->owner)) return -ENODEV; - soc_init_component_debugfs(&codec->component); + soc_init_component_debugfs(component); - if (driver->dapm_widgets) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - driver->dapm_widgets, - driver->num_dapm_widgets); + if (component->dapm_widgets) { + ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets, + component->num_dapm_widgets); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create new controls %d\n", ret); goto err_probe; } } - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) { - ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + /* + * This is rather ugly, but certain platforms expect that the DAPM + * widgets for the DAIs for components with the same parent device are + * created in the platforms DAPM context. Until that is fixed we need to + * keep this. + */ + if (component->steal_sibling_dai_widgets) { + dai_component = NULL; + list_for_each_entry(component2, &component_list, list) { + if (component == component2) + continue; - if (ret != 0) { - dev_err(codec->dev, - "Failed to create DAI widgets %d\n", ret); - goto err_probe; + if (component2->dev == component->dev && + !list_empty(&component2->dai_list)) { + dai_component = component2; + break; + } } - } - - codec->dapm.idle_bias_off = driver->idle_bias_off; - - if (driver->probe) { - ret = driver->probe(codec); - if (ret < 0) { - dev_err(codec->dev, - "ASoC: failed to probe CODEC %d\n", ret); - goto err_probe; + } else { + dai_component = component; + list_for_each_entry(component2, &component_list, list) { + if (component2->dev == component->dev && + component2->steal_sibling_dai_widgets) { + dai_component = NULL; + break; + } } - WARN(codec->dapm.idle_bias_off && - codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->component.name); } - if (driver->controls) - snd_soc_add_codec_controls(codec, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, - driver->num_dapm_routes); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); - list_add(&codec->dapm.list, &card->dapm_list); - - return 0; - -err_probe: - soc_cleanup_component_debugfs(&codec->component); - module_put(codec->dev->driver->owner); - - return ret; -} - -static int soc_probe_platform(struct snd_soc_card *card, - struct snd_soc_platform *platform) -{ - int ret = 0; - const struct snd_soc_platform_driver *driver = platform->driver; - struct snd_soc_component *component; - struct snd_soc_dai *dai; - - platform->component.card = card; - platform->component.dapm.card = card; - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - soc_init_component_debugfs(&platform->component); - - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->component.dapm, - driver->dapm_widgets, driver->num_dapm_widgets); - - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(component, &component_list, list) { - if (component->dev != platform->dev) - continue; - list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->component.dapm, - dai); + if (dai_component) { + list_for_each_entry(dai, &dai_component->dai_list, list) { + snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", + ret); + goto err_probe; + } + } } - platform->component.dapm.idle_bias_off = 1; - - if (driver->probe) { - ret = driver->probe(platform); + if (component->probe) { + ret = component->probe(component); if (ret < 0) { - dev_err(platform->dev, - "ASoC: failed to probe platform %d\n", ret); + dev_err(component->dev, + "ASoC: failed to probe component %d\n", ret); goto err_probe; } + + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); } - if (driver->controls) - snd_soc_add_platform_controls(platform, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->component.dapm, - driver->dapm_routes, driver->num_dapm_routes); + if (component->controls) + snd_soc_add_component_controls(component, component->controls, + component->num_controls); + if (component->dapm_routes) + snd_soc_dapm_add_routes(dapm, component->dapm_routes, + component->num_dapm_routes); - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->component.dapm.list, &card->dapm_list); + component->probed = 1; + list_add(&dapm->list, &card->dapm_list); + + /* This is a HACK and will be removed soon */ + if (component->codec) + list_add(&component->codec->card_list, &card->codec_dev_list); return 0; err_probe: - soc_cleanup_component_debugfs(&platform->component); - module_put(platform->dev->driver->owner); + soc_cleanup_component_debugfs(component); + module_put(component->dev->driver->owner); return ret; } @@ -1334,33 +1280,36 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_component *component; int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (cpu_dai->codec && - !cpu_dai->codec->probed && - cpu_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, cpu_dai->codec); - if (ret < 0) - return ret; + if (rtd->cpu_dai->codec) { + component = &rtd->cpu_dai->codec->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; + } } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->codec->probed && - rtd->codec_dais[i]->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + component = &rtd->codec_dais[i]->codec->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } } /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); + if (!platform->component.probed && + platform->component.driver->probe_order == order) { + ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; } @@ -1647,12 +1596,12 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->probed) { + if (rtd->codec->component.probed) { dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_codec(card, rtd->codec); + ret = soc_probe_component(card, &rtd->codec->component); if (ret < 0) return ret; @@ -1681,8 +1630,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->probed) - soc_remove_codec(codec); + if (codec && codec->component.probed) + soc_remove_component(&codec->component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -4198,6 +4147,20 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); +static int snd_soc_platform_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_platform *platform = snd_soc_component_to_platform(component); + + return platform->driver->probe(platform); +} + +static void snd_soc_platform_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_platform *platform = snd_soc_component_to_platform(component); + + platform->driver->remove(platform); +} + static int snd_soc_platform_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4234,6 +4197,24 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; + if (platform_drv->controls) { + platform->component.controls = platform_drv->controls; + platform->component.num_controls = platform_drv->num_controls; + } + if (platform_drv->dapm_widgets) { + platform->component.dapm_widgets = platform_drv->dapm_widgets; + platform->component.num_dapm_widgets = platform_drv->num_dapm_widgets; + platform->component.steal_sibling_dai_widgets = true; + } + if (platform_drv->dapm_routes) { + platform->component.dapm_routes = platform_drv->dapm_routes; + platform->component.num_dapm_routes = platform_drv->num_dapm_routes; + } + + if (platform_drv->probe) + platform->component.probe = snd_soc_platform_drv_probe; + if (platform_drv->remove) + platform->component.remove = snd_soc_platform_drv_remove; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) @@ -4363,6 +4344,20 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) stream->formats |= codec_format_map[i]; } +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_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4411,12 +4406,30 @@ int snd_soc_register_codec(struct device *dev, return -ENOMEM; codec->component.dapm_ptr = &codec->dapm; + codec->component.codec = codec; ret = snd_soc_component_initialize(&codec->component, &codec_drv->component_driver, dev); if (ret) goto err_free; + if (codec_drv->controls) { + codec->component.controls = codec_drv->controls; + codec->component.num_controls = codec_drv->num_controls; + } + if (codec_drv->dapm_widgets) { + codec->component.dapm_widgets = codec_drv->dapm_widgets; + codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; + } + if (codec_drv->dapm_routes) { + codec->component.dapm_routes = codec_drv->dapm_routes; + codec->component.num_dapm_routes = codec_drv->num_dapm_routes; + } + + 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->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6307f85e871b..b329b84bc5af 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -336,10 +336,12 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { }; static const struct snd_soc_platform_driver dmaengine_pcm_platform = { + .component_driver = { + .probe_order = SND_SOC_COMP_ORDER_LATE, + }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, .pcm_free = dmaengine_pcm_free, - .probe_order = SND_SOC_COMP_ORDER_LATE, }; static const char * const dmaengine_pcm_dma_channel_names[] = { -- cgit v1.2.3 From 93c3ce76ccced3a8718149e8734ccaa931e9a1f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:20 +0200 Subject: ASoC: Make rtd->codec optional There are some place in the ASoC core that expect rtd->codec to be non NULL (mainly CODEC specific sysfs files). With componentization going forward rtd->codec might be NULL in some cases. This patch prepares the core for this by not registering CODEC specific sysfs files if rtd->codec is NULL. sysfs file removal does not need to be conditionalized as it handles the removal of non-existing files just fine. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b833cc6fd86d..1c705c28389c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1261,17 +1261,21 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } rtd->dev_registered = 1; - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); + if (rtd->codec) { + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(rtd->dev); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec dapm sysfs entries: %d\n", + ret); - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", ret); + /* add codec sysfs entries */ + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec sysfs files: %d\n", + ret); + } return 0; } -- cgit v1.2.3 From 61aca5646b736a794d40de29a197144db3f0c5ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:21 +0200 Subject: ASoC: Add component level probe/remove support Now that we have a unified probe and remove path make sure to call them for all components. soc_{probe,remove}_component are responsible for setting up the DAPM context for the component, initialize the component prefix, manage the debugfs entries as well as do the registration of table based controls and DAPM elements. They also call the component drivers probe and remove callbacks. This patch makes these things available for generic snd_soc_component drivers rather than only having them for snd_soc_codec and snd_soc_platform drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 11 +++++++++++ sound/soc/soc-core.c | 42 ++++++++++++++++++++++++------------------ 2 files changed, 35 insertions(+), 18 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 22543acfae4b..4a223a895f00 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -690,6 +690,17 @@ struct snd_soc_compr_ops { struct snd_soc_component_driver { const char *name; + /* Default control and setup, added after probe() is run */ + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + unsigned int num_dapm_routes; + + int (*probe)(struct snd_soc_component *); + void (*remove)(struct snd_soc_component *); + /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c705c28389c..08fd85e8c751 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1058,7 +1058,7 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_codec *codec; + struct snd_soc_component *component; int i; /* remove the platform */ @@ -1068,18 +1068,17 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { - codec = rtd->codec_dais[i]->codec; - if (codec && codec->component.probed && - codec->component.driver->remove_order == order) - soc_remove_component(&codec->component); + component = rtd->codec_dais[i]->component; + if (component->probed && + component->driver->remove_order == order) + soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - codec = cpu_dai->codec; - if (codec && codec->component.probed && - codec->component.driver->remove_order == order) - soc_remove_component(&codec->component); + if (cpu_dai->component->probed && + cpu_dai->component->driver->remove_order == order) + soc_remove_component(cpu_dai->component); } } @@ -1289,19 +1288,17 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (rtd->cpu_dai->codec) { - component = &rtd->cpu_dai->codec->component; - if (!component->probed && - component->driver->probe_order == order) { - ret = soc_probe_component(card, component); - if (ret < 0) - return ret; - } + component = rtd->cpu_dai->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - component = &rtd->codec_dais[i]->codec->component; + component = rtd->codec_dais[i]->component; if (!component->probed && component->driver->probe_order == order) { ret = soc_probe_component(card, component); @@ -4042,6 +4039,8 @@ 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; if (!component->dapm_ptr) component->dapm_ptr = &component->dapm; @@ -4055,6 +4054,13 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, if (driver->stream_event) dapm->stream_event = snd_soc_component_stream_event; + component->controls = driver->controls; + component->num_controls = driver->num_controls; + component->dapm_widgets = driver->dapm_widgets; + component->num_dapm_widgets = driver->num_dapm_widgets; + component->dapm_routes = driver->dapm_routes; + component->num_dapm_routes = driver->num_dapm_routes; + INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); -- cgit v1.2.3 From 65d9361f0cb50a20641802ee3075145d72e4409c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:22 +0200 Subject: ASoC: Move AUX dev support to the component level This patch makes it possible to register arbitrary components as a AUX dev for a card. This was previously only possible for CODEC components. With componentization having made it possible for components to have DAPM contexts and controls there is no reason why AUX devs should be artificially limited to snd_soc_codec devices. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 48 ++++++++++++++++++++++++++++++++++++------------ 2 files changed, 37 insertions(+), 12 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4a223a895f00..fbc2ad840244 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1140,6 +1140,7 @@ struct snd_soc_pcm_runtime { struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_component *component; /* Only valid for AUX dev rtds */ struct snd_soc_dai **codec_dais; unsigned int num_codecs; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08fd85e8c751..08c04f4c7e62 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -860,6 +860,23 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; +static struct snd_soc_component *soc_find_component( + const struct device_node *of_node, const char *name) +{ + struct snd_soc_component *component; + + list_for_each_entry(component, &component_list, list) { + if (of_node) { + if (component->dev->of_node == of_node) + return component; + } else if (strcmp(component->name, name) == 0) { + return component; + } + } + + return NULL; +} + static struct snd_soc_codec *soc_find_codec( const struct device_node *codec_of_node, const char *codec_name) @@ -1577,17 +1594,24 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; + const char *name = aux_dev->codec_name; - rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); - if (!rtd->codec) { + rtd->component = soc_find_component(aux_dev->codec_of_node, name); + if (!rtd->component) { if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); + name = of_node_full_name(aux_dev->codec_of_node); - dev_err(card->dev, "ASoC: %s not registered\n", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } + /* + * Some places still reference rtd->codec, so we have to keep that + * initialized if the component is a CODEC. Once all those references + * have been removed, this code can be removed as well. + */ + rtd->codec = rtd->component->codec; + return 0; } @@ -1597,18 +1621,18 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->component.probed) { - dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); + if (rtd->component->probed) { + dev_err(rtd->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_component(card, &rtd->codec->component); + ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(&rtd->codec->dapm); + ret = aux_dev->init(snd_soc_component_get_dapm(rtd->component)); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); @@ -1622,7 +1646,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) static void soc_remove_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_component *component = rtd->component; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1631,8 +1655,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->component.probed) - soc_remove_component(&codec->component); + if (component && component->probed) + soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) -- cgit v1.2.3 From 57bf772687700e206c760ba2e4097f78bde97887 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:23 +0200 Subject: ASoC: Pass component instead of DAPM context to AUX dev init callback Given that the component is the containing structure it makes more sense to pass the component rather than the DAPM context to the AUX dev init callback. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/samsung/speyside.c | 6 ++++-- sound/soc/soc-core.c | 2 +- 3 files changed, 6 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index fbc2ad840244..3a0031e1f9b4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1022,7 +1022,7 @@ struct snd_soc_aux_dev { const struct device_node *codec_of_node; /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_dapm_context *dapm); + int (*init)(struct snd_soc_component *component); }; /* SoC card */ diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 9902efcb8ea1..a05482651aae 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -228,10 +228,12 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; -static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) +static int speyside_wm9081_init(struct snd_soc_component *component) { + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, MCLK_AUDIO_RATE, 0); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08c04f4c7e62..4393bc33d3af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1632,7 +1632,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(snd_soc_component_get_dapm(rtd->component)); + ret = aux_dev->init(rtd->component); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); -- cgit v1.2.3 From 70090bbb8b7d7da7a6f64969b43a61c493c560ff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:24 +0200 Subject: ASoC: Move component->probed check into soc_{remove,probe}_component() Having the check in a centralized place makes the code a bit cleaner and shorter. Note: There is a slight semantic change in this patch. soc_probe_aux_dev() will no longer return -EBUSY if the AUX dev has already been probed before. This is fine though since it will simply do nothing in that case and return success. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4393bc33d3af..2fbfbfca48dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1004,6 +1004,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) static void soc_remove_component(struct snd_soc_component *component) { + if (!component->probed) + return; + /* This is a HACK and will be removed soon */ if (component->codec) list_del(&component->codec->card_list); @@ -1079,22 +1082,19 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, int i; /* remove the platform */ - if (platform && platform->component.probed && - platform->component.driver->remove_order == order) + if (platform && platform->component.driver->remove_order == order) soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { component = rtd->codec_dais[i]->component; - if (component->probed && - component->driver->remove_order == order) + if (component->driver->remove_order == order) soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - if (cpu_dai->component->probed && - cpu_dai->component->driver->remove_order == order) + if (cpu_dai->component->driver->remove_order == order) soc_remove_component(cpu_dai->component); } } @@ -1145,6 +1145,9 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; + if (component->probed) + return 0; + component->card = card; dapm->card = card; soc_set_name_prefix(card, component); @@ -1306,8 +1309,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, /* probe the CPU-side component, if it is a CODEC */ component = rtd->cpu_dai->component; - if (!component->probed && - component->driver->probe_order == order) { + if (component->driver->probe_order == order) { ret = soc_probe_component(card, component); if (ret < 0) return ret; @@ -1316,8 +1318,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { component = rtd->codec_dais[i]->component; - if (!component->probed && - component->driver->probe_order == order) { + if (component->driver->probe_order == order) { ret = soc_probe_component(card, component); if (ret < 0) return ret; @@ -1325,8 +1326,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, } /* probe the platform */ - if (!platform->component.probed && - platform->component.driver->probe_order == order) { + if (platform->component.driver->probe_order == order) { ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; @@ -1621,11 +1621,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->component->probed) { - dev_err(rtd->dev, "ASoC: codec already probed\n"); - return -EBUSY; - } - ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; -- cgit v1.2.3 From ffbd7dd72bd3ad9bcae9190788c858e57f1e8e4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:25 +0200 Subject: ASoC: Cleanup DAI module reference counting Currently when a DAI has no CODEC associated to it the reference on the module containing the DAI driver is increased when the DAI is probed and decrease when the DAI is removed. For DAIs with CODECs the module reference count was already incremented when the CODEC is probed. Now that all components have their module reference count incremented when they are probed and all DAIs do have a component it is possible to remove the module reference counting on DAI probe and removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fbfbfca48dc..4dc2876c06de 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1067,8 +1067,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) cpu_dai->name, err); } cpu_dai->probed = 0; - if (!cpu_dai->codec) - module_put(cpu_dai->dev->driver->owner); } } @@ -1422,18 +1420,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - if (!cpu_dai->codec) { - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - } - if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: failed to probe CPU DAI %s: %d\n", cpu_dai->name, ret); - module_put(cpu_dai->dev->driver->owner); return ret; } } -- cgit v1.2.3 From e60cd14f0bf6c004cd7032a24a036ba32d56e08a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:26 +0200 Subject: ASoC: Consolidate CPU and CODEC DAI removal CPU and CODEC DAI works exactly the same way. There is already a helper function for CODEC DAI removal, use that one as well for CPU DAI removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dc2876c06de..5f6f97874ca2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1021,28 +1021,27 @@ static void soc_remove_component(struct snd_soc_component *component) module_put(component->dev->driver->owner); } -static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) +static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); + if (dai && dai->probed && + dai->driver->remove_order == order) { + if (dai->driver->remove) { + err = dai->driver->remove(dai); if (err < 0) - dev_err(codec_dai->dev, + dev_err(dai->dev, "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + dai->name, err); } - codec_dai->probed = 0; + dai->probed = 0; } } static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, err; + int i; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1054,20 +1053,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* remove the CODEC DAI */ for (i = 0; i < rtd->num_codecs; i++) - soc_remove_codec_dai(rtd->codec_dais[i], order); + soc_remove_dai(rtd->codec_dais[i], order); - /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed && - cpu_dai->driver->remove_order == order) { - if (cpu_dai->driver->remove) { - err = cpu_dai->driver->remove(cpu_dai); - if (err < 0) - dev_err(cpu_dai->dev, - "ASoC: failed to remove %s: %d\n", - cpu_dai->name, err); - } - cpu_dai->probed = 0; - } + soc_remove_dai(rtd->cpu_dai, order); } static void soc_remove_link_components(struct snd_soc_card *card, int num, -- cgit v1.2.3 From 14621c7e5e72200ec021a7580121130ce7f2ff22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:27 +0200 Subject: ASoC: Consolidate CPU and CODEC DAI lookup The lookup of CPU and CODEC DAIs is fairly similar and can easily be consolidated into a single helper function. There are two main differences in the current implementation of the CPU and CODEC DAI lookup: 1) CPU DAIs can be looked up by the DAI name alone and do not necessarily require a component name/of_node. 2) The CODEC DAI search only considers DAIs from CODEC components. For 1) the new helper function will allow to lookup DAIs without providing a component name or of_node, but since snd_soc_register_card() already rejects CODEC DAI link components without neither a of_node or a name we'll never get into the situation where we try to lookup a CODEC DAI without a name/of_node. For 2) the new helper function just always considers all components. Componentization is now at a point where it is possible to register a CODEC as a snd_soc_component rather than a snd_soc_codec, by considering DAIs from all components it is possible to use such a CODEC in a DAI link. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 72 ++++++++++++++-------------------------------------- 1 file changed, 19 insertions(+), 53 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5f6f97874ca2..140f43f91635 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -877,35 +877,23 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_codec *soc_find_codec( - const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc) { - struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct snd_soc_dai *dai; - list_for_each_entry(codec, &codec_list, list) { - if (codec_of_node) { - if (codec->dev->of_node != codec_of_node) - continue; - } else { - if (strcmp(codec->component.name, codec_name)) + /* Find CPU DAI from registered DAIs*/ + list_for_each_entry(component, &component_list, list) { + if (dlc->of_node && component->dev->of_node != dlc->of_node) + continue; + if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + continue; + list_for_each_entry(dai, &component->dai_list, list) { + if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) continue; - } - - return codec; - } - - return NULL; -} - -static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, - const char *codec_dai_name) -{ - struct snd_soc_dai *codec_dai; - list_for_each_entry(codec_dai, &codec->component.dai_list, list) { - if (!strcmp(codec_dai->name, codec_dai_name)) { - return codec_dai; + return dai; } } @@ -916,33 +904,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_component *component; struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; - struct snd_soc_dai *cpu_dai; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); - /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { - if (dai_link->cpu_of_node && - component->dev->of_node != dai_link->cpu_of_node) - continue; - if (dai_link->cpu_name && - strcmp(dev_name(component->dev), dai_link->cpu_name)) - continue; - list_for_each_entry(cpu_dai, &component->dai_list, list) { - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - - rtd->cpu_dai = cpu_dai; - } - } - + cpu_dai_component.name = dai_link->cpu_name; + cpu_dai_component.of_node = dai_link->cpu_of_node; + cpu_dai_component.dai_name = dai_link->cpu_dai_name; + rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); @@ -953,15 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CODEC from registered CODECs */ for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_codec *codec; - codec = soc_find_codec(codecs[i].of_node, codecs[i].name); - if (!codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - codecs[i].name); - return -EPROBE_DEFER; - } - - codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); -- cgit v1.2.3 From 886f5692253de1a9509f5cb708432b2157afb57c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:28 +0200 Subject: ASoC: Automatically initialize regmap for all components So far regmap is only automatically initialized for CODECs. Now that we have the infrastructure in place to let components have DAPM widgets and controls that want to use the generic regmap based IO also make sure to automatically initialize regmap for all components. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- sound/soc/soc-core.c | 35 +++++++++++++++++------------------ sound/soc/soc-io.c | 28 ---------------------------- 3 files changed, 17 insertions(+), 49 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3a0031e1f9b4..8ebee30311e3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1289,9 +1289,6 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); -int snd_soc_component_init_io(struct snd_soc_component *component, - struct regmap *regmap); - /* device driver data */ static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 140f43f91635..96f286643ca1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4032,8 +4032,23 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, return 0; } +static void snd_soc_component_init_regmap(struct snd_soc_component *component) +{ + if (!component->regmap) + component->regmap = dev_get_regmap(component->dev, NULL); + if (component->regmap) { + int val_bytes = regmap_get_val_bytes(component->regmap); + /* Errors are legitimate for non-integer byte multiples */ + if (val_bytes > 0) + component->val_bytes = val_bytes; + } +} + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { + if (!component->write && !component->read) + snd_soc_component_init_regmap(component); + list_add(&component->list, &component_list); } @@ -4371,7 +4386,6 @@ int snd_soc_register_codec(struct device *dev, { struct snd_soc_codec *codec; struct snd_soc_dai *dai; - struct regmap *regmap; int ret, i; dev_dbg(dev, "codec register %s\n", dev_name(dev)); @@ -4425,23 +4439,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.debugfs_prefix = "codec"; #endif - if (!codec->component.write) { - if (codec_drv->get_regmap) - regmap = codec_drv->get_regmap(dev); - else - regmap = dev_get_regmap(dev, NULL); - - if (regmap) { - ret = snd_soc_component_init_io(&codec->component, - regmap); - if (ret) { - dev_err(codec->dev, - "Failed to set cache I/O:%d\n", - ret); - goto err_cleanup; - } - } - } + if (codec_drv->get_regmap) + codec->component.regmap = codec_drv->get_regmap(dev); for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 7767fbd73eb7..9b3939049cef 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -271,31 +271,3 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, return snd_soc_component_write(&platform->component, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); - -/** - * snd_soc_component_init_io() - Initialize regmap IO - * - * @component: component to initialize - * @regmap: regmap instance to use for IO operations - * - * Return: 0 on success, a negative error code otherwise - */ -int snd_soc_component_init_io(struct snd_soc_component *component, - struct regmap *regmap) -{ - int ret; - - if (!regmap) - return -EINVAL; - - ret = regmap_get_val_bytes(regmap); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - component->val_bytes = ret; - - component->regmap = regmap; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_component_init_io); -- cgit v1.2.3 From 75af7c081982d76cef0daf26e96b5d1e8cb9d631 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:29 +0200 Subject: ASoC: Remove support for legacy snd_soc_platform IO There were never any actual users of this in upstream and by we have with regmap a replacement in place, which should be used by new drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- sound/soc/soc-core.c | 22 ---------------------- 2 files changed, 25 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ebee30311e3..edbb0d72ab38 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -890,9 +890,6 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; - /* platform IO - used for platform DAPM */ - unsigned int (*read)(struct snd_soc_platform *, unsigned int); - int (*write)(struct snd_soc_platform *, unsigned int, unsigned int); int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f286643ca1..2d7a9ecbb0e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4151,24 +4151,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) -{ - struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - - return platform->driver->write(platform, reg, val); -} - -static int snd_soc_platform_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) -{ - struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - - *val = platform->driver->read(platform, reg); - - return 0; -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -4205,10 +4187,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->write) - platform->component.write = snd_soc_platform_drv_write; - if (platform_drv->read) - platform->component.read = snd_soc_platform_drv_read; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; -- cgit v1.2.3 From c5599b87a8317738a541d8893cb327df5d04b007 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:30 +0200 Subject: ASoC: Replace list_empty(&card->codec_dev_list) with !card->instantiated With componentization we no longer necessarily need a snd_soc_codec struct for a card. Instead of checking if the card's CODEC list is empty just use card->instantiated to check if the card has been instantiated yet. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d7a9ecbb0e3..c36983a133fa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -552,10 +552,8 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_codec *codec; int i, j; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* Due to the resume being scheduled into a workqueue we could @@ -808,10 +806,8 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* activate pins from sleep state */ -- cgit v1.2.3 From 5819c2fa55d4a6eaf7fe025a393dce98fc4b2116 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 24 Aug 2014 15:36:55 +0200 Subject: ASoC: Restore idle_bias_off initialization This was accidentally lost in commit f1d45cc3ae96 ("ASoC: Consolidate platform and CODEC probe/remove"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c36983a133fa..419682693886 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4010,6 +4010,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; + dapm->idle_bias_off = true; if (driver->seq_notifier) dapm->seq_notifier = snd_soc_component_seq_notifier; if (driver->stream_event) @@ -4399,6 +4400,7 @@ int snd_soc_register_codec(struct device *dev, codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.codec = codec; + codec->dapm.idle_bias_off = codec_drv->idle_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) -- cgit v1.2.3 From b792346fa8660a22a06f118cebe47709f507914f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Aug 2014 14:07:11 +0300 Subject: ASoC: Remove unused cache_only from struct snd_soc_codec There are no real users for cache_only in "struct snd_soc_codec" so remove it and needless debugfs node. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 2 -- 2 files changed, 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index edbb0d72ab38..ce09302bfd6d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -792,7 +792,6 @@ struct snd_soc_codec { unsigned int ac97_registered:1; /* Codec has been AC97 registered */ unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int cache_init:1; /* codec cache has been initialized */ - u32 cache_only; /* Suppress writes to hardware */ u32 cache_sync; /* Cache needs to be synced to hardware */ /* codec IO */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 419682693886..1b422c5c36c8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -311,8 +311,6 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, - &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->component.debugfs_root, -- cgit v1.2.3 From b43cfb245f7346cbb25c1919577d9607d2adb974 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:30 +0200 Subject: ASoC: adau1373: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Also drop the regcache_cache_only() calls from the suspend and resume handlers. There shouldn't be any IO happening after suspend and before resume. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ff7d4d027e9..194756549ef4 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1454,23 +1454,10 @@ static int adau1373_remove(struct snd_soc_codec *codec) return 0; } -static int adau1373_suspend(struct snd_soc_codec *codec) -{ - struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adau1373->regmap, true); - - return ret; -} - static int adau1373_resume(struct snd_soc_codec *codec) { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adau1373->regmap, false); - adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau1373->regmap); return 0; @@ -1502,7 +1489,6 @@ static const struct regmap_config adau1373_regmap_config = { static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, - .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, -- cgit v1.2.3 From 8e6fe35eabc64f35eff5844a2e542c403a00db15 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:31 +0200 Subject: ASoC: lm49453: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 275b3f72f3f4..c1ae5764983f 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1395,18 +1395,6 @@ static struct snd_soc_dai_driver lm49453_dai[] = { }, }; -static int lm49453_suspend(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int lm49453_resume(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1416,8 +1404,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { .remove = lm49453_remove, - .suspend = lm49453_suspend, - .resume = lm49453_resume, .set_bias_level = lm49453_set_bias_level, .controls = lm49453_snd_controls, .num_controls = ARRAY_SIZE(lm49453_snd_controls), -- cgit v1.2.3 From 7d1a99da0861330f02de5c0f59df1d338477cb54 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:32 +0200 Subject: ASoC: tlv320aic3x: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64f179ee9834..f2c416d16b6c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1222,20 +1222,6 @@ static struct snd_soc_dai_driver aic3x_dai = { .symmetric_rates = 1, }; -static int aic3x_suspend(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int aic3x_resume(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void aic3x_mono_init(struct snd_soc_codec *codec) { /* DAC to Mono Line Out default volume and route to Output mixer */ @@ -1429,8 +1415,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .idle_bias_off = true, .probe = aic3x_probe, .remove = aic3x_remove, - .suspend = aic3x_suspend, - .resume = aic3x_resume, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), .dapm_widgets = aic3x_dapm_widgets, -- cgit v1.2.3 From a7edeba4cbbd0f3d22d6d54da7c507bda29b2658 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:33 +0200 Subject: ASoC: wm8804: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 0ea01dfcb6e1..3addc5fe5cb2 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -518,23 +518,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8804_suspend(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8804_resume(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8804_suspend NULL -#define wm8804_resume NULL -#endif - static int wm8804_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; @@ -671,8 +654,6 @@ static struct snd_soc_dai_driver wm8804_dai = { static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .probe = wm8804_probe, .remove = wm8804_remove, - .suspend = wm8804_suspend, - .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, -- cgit v1.2.3 From e02c716d2ec065fd58c2fc8100fd5f359ab61e7e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:34 +0200 Subject: ASoC: wm8995: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index cae4ac5a5730..1288edeb8c7d 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1998,23 +1998,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8995_suspend(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8995_resume(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8995_suspend NULL -#define wm8995_resume NULL -#endif - static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; @@ -2220,8 +2203,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .probe = wm8995_probe, .remove = wm8995_remove, - .suspend = wm8995_suspend, - .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, .idle_bias_off = true, }; -- cgit v1.2.3 From 01e0df6647e713469466c7bb6d7157c2e3046192 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:04 +0200 Subject: ASoC: Set card->instantiated to false when removing the card Set card->instantiated to false when the card is removed to make sure that operations that expect the card to be fully instantiated do not run anymore during card removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b422c5c36c8..ff9d2892f473 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3810,8 +3810,10 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) + if (card->instantiated) { + card->instantiated = false; soc_cleanup_card_resources(card); + } dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; -- cgit v1.2.3 From 1c325f771a88579f227fe017e4ee77d852cf5435 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:05 +0200 Subject: ASoC: Shutdown DAPM contexts when removing a card Currently when a ASoC sound card is unregistered we leave the individual components in their current state, just call the remove() callback and leave it to the drivers to do the proper shutdown/cleanup. This patch introduces a call to snd_soc_dapm_shutdown() when removing the card. This will make sure that all DAPM widgets are properly powered down and all DAPM contexts are put at the SND_SOC_BIAS_OFF level. This will ensure that all components are properly powered down when the card is removed. Since a lot of drivers manually go to SND_SOC_BIAS_OFF in their remove callback this will also allow us to remove a bit of duplicated code. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ff9d2892f473..068785fa1a06 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3812,6 +3812,7 @@ int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) { card->instantiated = false; + snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); } dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); -- cgit v1.2.3 From 86dbf2ac6fcb2d2932d4610f2dfe0954aa0633f7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:06 +0200 Subject: ASoC: Add support for automatically going to BIAS_OFF on suspend There is a substantial amount of drivers that in go to SND_SOC_BIAS_OFF on suspend and go back to SND_SOC_BIAS_SUSPEND on resume (Often this is even the only thing done in the suspend and resume handlers). This patch introduces a new suspend_bias_off flag, which when set by a driver will let the ASoC core automatically put the device's DAPM context at the SND_SOC_BIAS_OFF level during suspend. Once the device is resumed the DAPM context will go back to SND_SOC_BIAS_STANDBY (if the context is idle, otherwise to SND_SOC_BIAS_ON). This will allow us to remove a fair bit of duplicated code from the drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 20 ++++++++++++++++++-- 4 files changed, 22 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index aac04ff84eea..f955d65c5656 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -587,7 +587,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - + /* Go to BIAS_OFF in suspend if the DAPM context is idle */ + unsigned int suspend_bias_off:1; void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/include/sound/soc.h b/include/sound/soc.h index ce09302bfd6d..ac99fc083eec 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -848,6 +848,7 @@ struct snd_soc_codec_driver { 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); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068785fa1a06..2bdf9a4ac2b4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4402,6 +4402,7 @@ int snd_soc_register_codec(struct device *dev, codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.codec = codec; codec->dapm.idle_bias_off = codec_drv->idle_bias_off; + codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352dc2c6..a2025a6b6a29 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1683,6 +1683,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } +static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) +{ + if (dapm->idle_bias_off) + return true; + + switch (snd_power_get_state(dapm->card->snd_card)) { + case SNDRV_CTL_POWER_D3hot: + case SNDRV_CTL_POWER_D3cold: + return dapm->suspend_bias_off; + default: + break; + } + + return false; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1706,7 +1722,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->idle_bias_off) + if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1772,7 +1788,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - if (!d->idle_bias_off) + if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); -- cgit v1.2.3 From a80932979a72ef9d4e66a69520c7588cc6de5699 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:07 +0200 Subject: ASoC: Always run default suspend/resume code We do a bit more than just running the callbacks during suspend and resume these days (e.g. call regcache_mark_dirty() during suspend). But this is only when suspend and resume callbacks are specified for the driver, otherwise nothing is done. This means that drivers which don't want to do anything special during suspend and resume, but still want the standard operations to run, need to provide empty suspend and resume callback functions (rather than no callbacks). This patch updates the suspend and resume code to always run standard sequence regardless of whether suspend and resume handlers are provided. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2bdf9a4ac2b4..c612900c80ff 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -637,7 +637,7 @@ int snd_soc_suspend(struct device *dev) list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ - if (!codec->suspended && codec->driver->suspend) { + if (!codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* @@ -651,8 +651,10 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + case SND_SOC_BIAS_OFF: - codec->driver->suspend(codec); + if (codec->driver->suspend) + codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; if (codec->component.regmap) @@ -726,11 +728,12 @@ static void soc_resume_deferred(struct work_struct *work) * left with bias OFF or STANDBY and suspended so we must now * resume. Otherwise the suspend was suppressed. */ - if (codec->driver->resume && codec->suspended) { + if (codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: - codec->driver->resume(codec); + if (codec->driver->resume) + codec->driver->resume(codec); codec->suspended = 0; break; default: -- cgit v1.2.3 From d7858bd647cda68bf832997a280a2f44aec01f1b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:08 +0200 Subject: ASoC: adau1373: Cleanup manual bias level transitions The ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC, no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 194756549ef4..7c784ad3e8b2 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1448,12 +1448,6 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int adau1373_remove(struct snd_soc_codec *codec) -{ - adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int adau1373_resume(struct snd_soc_codec *codec) { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); @@ -1488,7 +1482,6 @@ static const struct regmap_config adau1373_regmap_config = { static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, - .remove = adau1373_remove, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, -- cgit v1.2.3 From 0e0f9b960a011a9e3815004f37cc475229170dfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:09 +0200 Subject: ASoC: adau17x1: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 8 -------- sound/soc/codecs/adau17x1.h | 1 - 4 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 848cab839553..5518ebd6947c 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -714,9 +714,9 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1761_codec_driver = { .probe = adau1761_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1761_set_bias_level, + .suspend_bias_off = true, .controls = adau1761_controls, .num_controls = ARRAY_SIZE(adau1761_controls), diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 045a61413840..e9fc00fb13dd 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -446,9 +446,9 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1781_codec_driver = { .probe = adau1781_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1781_set_bias_level, + .suspend_bias_off = true, .controls = adau1781_controls, .num_controls = ARRAY_SIZE(adau1781_controls), diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 0b659704e60c..3e16c1c64115 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -815,13 +815,6 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(adau17x1_add_routes); -int adau17x1_suspend(struct snd_soc_codec *codec) -{ - codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} -EXPORT_SYMBOL_GPL(adau17x1_suspend); - int adau17x1_resume(struct snd_soc_codec *codec) { struct adau *adau = snd_soc_codec_get_drvdata(codec); @@ -829,7 +822,6 @@ int adau17x1_resume(struct snd_soc_codec *codec) if (adau->switch_mode) adau->switch_mode(codec->dev); - codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau->regmap); return 0; diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 3ffabaf4c7a8..e4a557fd7155 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -52,7 +52,6 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); -int adau17x1_suspend(struct snd_soc_codec *codec); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -- cgit v1.2.3 From cd5d3a151118cd815be15970db099bcdb3f0ad12 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:10 +0200 Subject: ASoC: adav80x: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. While we are at it also remove the regcache_cache_only() calls from suspend/resume as there shouldn't be any IO between suspend and resume. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 23 ++--------------------- 1 file changed, 2 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c43b93fdf0df..ce3cdca9fc62 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -812,42 +812,23 @@ static int adav80x_probe(struct snd_soc_codec *codec) /* Disable DAC zero flag */ regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); - return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int adav80x_suspend(struct snd_soc_codec *codec) -{ - struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adav80x->regmap, true); - - return ret; + return 0; } static int adav80x_resume(struct snd_soc_codec *codec) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adav80x->regmap, false); - adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adav80x->regmap); return 0; } -static int adav80x_remove(struct snd_soc_codec *codec) -{ - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - static struct snd_soc_codec_driver adav80x_codec_driver = { .probe = adav80x_probe, - .remove = adav80x_remove, - .suspend = adav80x_suspend, .resume = adav80x_resume, .set_bias_level = adav80x_set_bias_level, + .suspend_bias_off = true, .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, -- cgit v1.2.3 From 0f0cc5a775ebe88d9be12489874bd2799b42e242 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:11 +0200 Subject: ASoC: ssm2518: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_OFF at the end of CODEC probe() can also be removed as the CODEC is already in OFF state at this point. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index e8680bea5f86..67ea55adb307 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -646,17 +646,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { .ops = &ssm2518_dai_ops, }; -static int ssm2518_probe(struct snd_soc_codec *codec) -{ - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int ssm2518_remove(struct snd_soc_codec *codec) -{ - ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -727,8 +716,6 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, } static struct snd_soc_codec_driver ssm2518_codec_driver = { - .probe = ssm2518_probe, - .remove = ssm2518_remove, .set_bias_level = ssm2518_set_bias_level, .set_sysclk = ssm2518_set_sysclk, .idle_bias_off = true, -- cgit v1.2.3 From 85362efb80070bed890602483f71cd103be303c2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:12 +0200 Subject: ASoC: ssm2602: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. While we are at it also remove the regcache_cache_only() calls from suspend/resume as there shouldn't be any IO between suspend and resume. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 24 ++---------------------- 1 file changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bbe8624..0dec13648563 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -502,18 +502,11 @@ static struct snd_soc_dai_driver ssm2602_dai = { .symmetric_samplebits = 1, }; -static int ssm2602_suspend(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2602_resume(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); regcache_sync(ssm2602->regmap); - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -586,27 +579,14 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) break; } - if (ret) - return ret; - - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* remove everything here */ -static int ssm2602_remove(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; + return ret; } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .probe = ssm260x_codec_probe, - .remove = ssm2602_remove, - .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, + .suspend_bias_off = true, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), -- cgit v1.2.3