From faff4bb067d15a3bc0dde8c50cbc1a7075e314de Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:11 -0700 Subject: ASoC: Export debugfs root dentry A couple Tegra ASoC drivers will create debugfs entries. Mark requested these by under debugfs/asoc/ not just debugfs/. To enable this, export the dentry representing debugfs/asoc/. Also, rename debugfs_root -> asoc_debugfs_root now it's exported to prevent potential symbol name clashes. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 74921f20a1d8..96aadbba85b2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -756,4 +756,8 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) #include +#ifdef CONFIG_DEBUG_FS +extern struct dentry *asoc_debugfs_root; +#endif + #endif -- cgit v1.2.3 From 8a9dab1a555e3f2088c68cae792dfd7e854e65e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Jan 2011 22:25:21 +0000 Subject: ASoC: Update name of debugfs root symbol to snd_soc_ Everything else is using snd_soc_ so we should use it here too. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 20 ++++++++++---------- sound/soc/tegra/tegra_das.c | 5 +++-- sound/soc/tegra/tegra_i2s.c | 2 +- 4 files changed, 15 insertions(+), 14 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 96aadbba85b2..c477058ff98a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -757,7 +757,7 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) #include #ifdef CONFIG_DEBUG_FS -extern struct dentry *asoc_debugfs_root; +extern struct dentry *snd_soc_debugfs_root; #endif #endif diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0484e504b589..2ff708a41119 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -48,8 +48,8 @@ static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS -struct dentry *asoc_debugfs_root; -EXPORT_SYMBOL_GPL(asoc_debugfs_root); +struct dentry *snd_soc_debugfs_root; +EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); @@ -361,7 +361,7 @@ static const struct file_operations platform_list_fops = { static void soc_init_card_debugfs(struct snd_soc_card *card) { card->debugfs_card_root = debugfs_create_dir(card->name, - asoc_debugfs_root); + snd_soc_debugfs_root); if (!card->debugfs_card_root) { dev_warn(card->dev, "ASoC: Failed to create codec debugfs directory\n"); @@ -3584,22 +3584,22 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS - asoc_debugfs_root = debugfs_create_dir("asoc", NULL); - if (IS_ERR(asoc_debugfs_root) || !asoc_debugfs_root) { + snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); + if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { printk(KERN_WARNING "ASoC: Failed to create debugfs directory\n"); - asoc_debugfs_root = NULL; + snd_soc_debugfs_root = NULL; } - if (!debugfs_create_file("codecs", 0444, asoc_debugfs_root, NULL, + if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL, &codec_list_fops)) pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); - if (!debugfs_create_file("dais", 0444, asoc_debugfs_root, NULL, + if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL, &dai_list_fops)) pr_warn("ASoC: Failed to create DAI list debugfs file\n"); - if (!debugfs_create_file("platforms", 0444, asoc_debugfs_root, NULL, + if (!debugfs_create_file("platforms", 0444, snd_soc_debugfs_root, NULL, &platform_list_fops)) pr_warn("ASoC: Failed to create platform list debugfs file\n"); #endif @@ -3611,7 +3611,7 @@ module_init(snd_soc_init); static void __exit snd_soc_exit(void) { #ifdef CONFIG_DEBUG_FS - debugfs_remove_recursive(asoc_debugfs_root); + debugfs_remove_recursive(snd_soc_debugfs_root); #endif platform_driver_unregister(&soc_driver); } diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 796d36d5188a..01eb9c9301de 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -144,8 +144,9 @@ static const struct file_operations tegra_das_debug_fops = { static void tegra_das_debug_add(struct tegra_das *das) { - das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, asoc_debugfs_root, - das, &tegra_das_debug_fops); + das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, das, + &tegra_das_debug_fops); } static void tegra_das_debug_remove(struct tegra_das *das) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 9b7a22af52a6..1730509c8ac2 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -104,7 +104,7 @@ static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) char name[] = DRV_NAME ".0"; snprintf(name, sizeof(name), DRV_NAME".%1d", id); - i2s->debug = debugfs_create_file(name, S_IRUGO, asoc_debugfs_root, + i2s->debug = debugfs_create_file(name, S_IRUGO, snd_soc_debugfs_root, i2s, &tegra_i2s_debug_fops); } -- cgit v1.2.3 From aea170a099793abcd0e6de46b947458073204241 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 12 Jan 2011 10:38:58 +0000 Subject: ASoC: soc-cache: Add reg_size as a member to snd_soc_codec Simplify the use of reg_size, by calculating it once and storing it in the codec structure for later reference. The value of reg_size is reg_cache_size * reg_word_size. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 26 ++++++++------------------ sound/soc/soc-core.c | 1 + 3 files changed, 10 insertions(+), 18 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index c477058ff98a..d609232da82a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -459,6 +459,7 @@ struct snd_soc_codec { struct list_head card_list; int num_dai; enum snd_soc_compress_type compress_type; + size_t reg_size; /* reg_cache_size * reg_word_size */ /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 19b29fb3ca4b..b2e333f5a388 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1100,34 +1100,28 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, unsigned int reg) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; return (reg * codec_drv->reg_word_size) / - DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); + DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, unsigned int reg) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - return reg % (DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()) / + return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) / codec_drv->reg_word_size); } static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - return DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); + return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) @@ -1287,7 +1281,7 @@ static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; - size_t reg_size, bmp_size; + size_t bmp_size; const struct snd_soc_codec_driver *codec_drv; int ret, tofree, i, blksize, blkcount; const char *p, *end; @@ -1295,7 +1289,6 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) ret = 0; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; /* * If we have not been given a default register cache @@ -1307,8 +1300,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) tofree = 1; if (!codec->reg_def_copy) { - codec->reg_def_copy = kzalloc(reg_size, - GFP_KERNEL); + codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL); if (!codec->reg_def_copy) return -ENOMEM; } @@ -1356,7 +1348,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) blksize = snd_soc_lzo_get_blksize(codec); p = codec->reg_def_copy; - end = codec->reg_def_copy + reg_size; + end = codec->reg_def_copy + codec->reg_size; /* compress the register map and fill the lzo blocks */ for (i = 0; i < blkcount; ++i, p += blksize) { lzo_blocks[i]->src = p; @@ -1441,16 +1433,14 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (codec->reg_def_copy) codec->reg_cache = kmemdup(codec->reg_def_copy, - reg_size, GFP_KERNEL); + codec->reg_size, GFP_KERNEL); else - codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); + codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL); if (!codec->reg_cache) return -ENOMEM; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 04475c18003c..cbac50b69c39 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3500,6 +3500,7 @@ int snd_soc_register_codec(struct device *dev, /* allocate CODEC register cache */ if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + codec->reg_size = reg_size; /* it is necessary to make a copy of the default register cache * because in the case of using a compression type that requires * the default register cache to be marked as __devinitconst the -- cgit v1.2.3 From 066d16c3e8194677a9aaeb06a45e4014387d16f1 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:36 +0000 Subject: ASoC: soc-cache: Add support for default readable()/volatile() functions For common scenarios, device drivers can provide a table of all the registers that are at least either readable/writable/volatile. The idea is that if a register lookup fails, all of its read/write/vol members will be zero and will be treated as default. This also reduces the size of the register access array. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 22 ++++++++++++++++++++++ sound/soc/soc-cache.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index d609232da82a..b8acf99ac89d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,6 +276,10 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -366,6 +370,22 @@ int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +/** + * struct snd_soc_reg_access - Describes whether a given register is + * readable, writable or volatile. + * + * @reg: the register number + * @read: whether this register is readable + * @write: whether this register is writable + * @vol: whether this register is volatile + */ +struct snd_soc_reg_access { + u16 reg; + u16 read; + u16 write; + u16 vol; +}; + /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * @@ -515,6 +535,8 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; + short reg_access_size; + const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1a36b36c5baa..d97a59f6a249 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1603,3 +1603,52 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); + +static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int min, max, index; + + codec_drv = codec->driver; + min = 0; + max = codec_drv->reg_access_size - 1; + do { + index = (min + max) / 2; + if (codec_drv->reg_access_default[index].reg == reg) + return index; + if (codec_drv->reg_access_default[index].reg < reg) + min = index + 1; + else + max = index; + } while (min <= max); + return -1; +} + +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].vol; +} +EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); + +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].read; +} +EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); -- cgit v1.2.3 From d4754ec91c7b094298f0b2ba02327e6887671edc Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:37 +0000 Subject: ASoC: Update users of readable_register()/volatile_register() Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- sound/soc/codecs/cs4270.c | 4 ++-- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 4 ++-- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 10 +++++----- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- 16 files changed, 25 insertions(+), 25 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b8acf99ac89d..97d1832bb9df 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -529,8 +529,8 @@ struct snd_soc_codec_driver { int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); int (*display_register)(struct snd_soc_codec *, char *, size_t, unsigned int); - int (*volatile_register)(unsigned int); - int (*readable_register)(unsigned int); + int (*volatile_register)(struct snd_soc_codec *, unsigned int); + int (*readable_register)(struct snd_soc_codec *, unsigned int); short reg_cache_size; short reg_cache_step; short reg_word_size; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b51245f2318..c0fccadaea9a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -193,12 +193,12 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -static int cs4270_reg_is_readable(unsigned int reg) +static int cs4270_reg_is_readable(struct snd_soc_codec *codec, unsigned int reg) { return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG); } -static int cs4270_reg_is_volatile(unsigned int reg) +static int cs4270_reg_is_volatile(struct snd_soc_codec *codec, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 37133c40e762..b6ecc7e89673 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -608,7 +608,7 @@ static struct { { 0xFF, 0x00, 1 }, /* FF */ }; -static int max98088_volatile_register(unsigned int reg) +static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { return max98088_access[reg].vol; } diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5eb2f501ce32..83e86f077ee1 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -58,7 +58,7 @@ static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = { 0x0000, /* R8 - ZERO_DETECT */ }; -static int wm8523_volatile_register(unsigned int reg) +static int wm8523_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8523_DEVICE_ID: diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6dae1b40c9f7..6785688f8806 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -175,7 +175,7 @@ static int txsrc_put(struct snd_kcontrol *kcontrol, return 0; } -static int wm8804_volatile(unsigned int reg) +static int wm8804_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8804_RST_DEVID1: diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index cd0959926d12..449ea09a193d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -180,7 +180,7 @@ static const u16 wm8900_reg_defaults[WM8900_MAXREG] = { /* Remaining registers all zero */ }; -static int wm8900_volatile_register(unsigned int reg) +static int wm8900_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8900_REG_ID: diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 987476a5895f..a2a446cb1807 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -232,7 +232,7 @@ struct wm8903_priv { int mic_delay; }; -static int wm8903_volatile_register(unsigned int reg) +static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8903_SW_RESET_AND_ID: diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9de44a4c05c0..17a8fe9b39b9 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -596,7 +596,7 @@ static struct { { 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */ }; -static int wm8904_volatile_register(unsigned int reg) +static int wm8904_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { return wm8904_access[reg].vol; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 55252e7d02c9..cdee8103d09b 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -291,7 +291,7 @@ struct wm8961_priv { int sysclk; }; -static int wm8961_volatile_register(unsigned int reg) +static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b9cb1fcf8c92..7c02924beddf 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1938,7 +1938,7 @@ static const struct wm8962_reg_access { [21139] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21139 - VSS_XTS32_0 */ }; -static int wm8962_volatile_register(unsigned int reg) +static int wm8962_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { if (wm8962_reg_access[reg].vol) return 1; @@ -1946,7 +1946,7 @@ static int wm8962_volatile_register(unsigned int reg) return 0; } -static int wm8962_readable_register(unsigned int reg) +static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int reg) { if (wm8962_reg_access[reg].read) return 1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 18c0d9ce7c32..379fa22c5b6c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -242,7 +242,7 @@ struct wm8993_priv { int fll_src; }; -static int wm8993_volatile(unsigned int reg) +static int wm8993_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8993_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb8..0bb0bb40b842 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -109,7 +109,7 @@ struct wm8994_priv { struct wm8994_pdata *pdata; }; -static int wm8994_readable(unsigned int reg) +static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8994_GPIO_1: @@ -136,7 +136,7 @@ static int wm8994_readable(unsigned int reg) return wm8994_access_masks[reg].readable != 0; } -static int wm8994_volatile(unsigned int reg) +static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) { if (reg >= WM8994_CACHE_SIZE) return 1; @@ -164,7 +164,7 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, BUG_ON(reg > WM8994_MAX_REGISTER); - if (!wm8994_volatile(reg)) { + if (!wm8994_volatile(codec, reg)) { ret = snd_soc_cache_write(codec, reg, value); if (ret != 0) dev_err(codec->dev, "Cache write to %x failed: %d\n", @@ -182,7 +182,7 @@ static unsigned int wm8994_read(struct snd_soc_codec *codec, BUG_ON(reg > WM8994_MAX_REGISTER); - if (!wm8994_volatile(reg) && wm8994_readable(reg) && + if (!wm8994_volatile(codec, reg) && wm8994_readable(codec, reg) && reg < codec->driver->reg_cache_size) { ret = snd_soc_cache_read(codec, reg, &val); if (ret >= 0) @@ -2943,7 +2943,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) /* Read our current status back from the chip - we don't want to * reset as this may interfere with the GPIO or LDO operation. */ for (i = 0; i < WM8994_CACHE_SIZE; i++) { - if (!wm8994_readable(i) || wm8994_volatile(i)) + if (!wm8994_readable(codec, i) || wm8994_volatile(codec, i)) continue; ret = wm8994_reg_read(codec->control_data, i); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index ac210ccebd4b..f0f678de489f 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -909,7 +909,7 @@ static const struct snd_soc_dapm_route wm8995_intercon[] = { { "SPK2R", NULL, "SPK2R Driver" } }; -static int wm8995_volatile(unsigned int reg) +static int wm8995_volatile(struct snd_soc_codec *codec, unsigned int reg) { /* out of bounds registers are generally considered * volatile to support register banks that are partially diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 43825b2102a5..5c224dd917d7 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -169,7 +169,7 @@ struct wm9081_priv { struct wm9081_retune_mobile_config *retune; }; -static int wm9081_volatile_register(unsigned int reg) +static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM9081_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a788c4297046..d40bfc9f8805 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -144,7 +144,7 @@ struct wm9090_priv { void *control_data; }; -static int wm9090_volatile(unsigned int reg) +static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM9090_SOFTWARE_RESET: @@ -518,7 +518,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, for (i = 1; i < codec->driver->reg_cache_size; i++) { if (reg_cache[i] == wm9090_reg_defaults[i]) continue; - if (wm9090_volatile(i)) + if (wm9090_volatile(codec, i)) continue; ret = snd_soc_write(codec, i, reg_cache[i]); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cbac50b69c39..b5e5758456bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->driver->readable_register && !codec->driver->readable_register(i)) + if (codec->driver->readable_register && !codec->driver->readable_register(codec, i)) continue; count += sprintf(buf + count, "%2x: ", i); @@ -2030,7 +2030,7 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { if (codec->driver->volatile_register) - return codec->driver->volatile_register(reg); + return codec->driver->volatile_register(codec, reg); else return 0; } -- cgit v1.2.3 From 1500b7b5ffaacb8199e0a61162f5d349fb19acbe Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:38 +0000 Subject: ASoC: Automatically assign the default readable()/volatile() functions Ensure that all calls to readable_register()/volatile_register() go via the snd_soc_codec function pointers. If the default register access table has been given but no functions for handling readable()/volatile() registers, use the default ones provided by soc-cache. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 15 ++++++++++++--- 2 files changed, 14 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 97d1832bb9df..accb8a16c165 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -480,6 +480,8 @@ struct snd_soc_codec { int num_dai; enum snd_soc_compress_type compress_type; size_t reg_size; /* reg_cache_size * reg_word_size */ + int (*volatile_register)(struct snd_soc_codec *, unsigned int); + int (*readable_register)(struct snd_soc_codec *, unsigned int); /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5e5758456bd..30d76e8bc9df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->driver->readable_register && !codec->driver->readable_register(codec, i)) + if (codec->readable_register && !codec->readable_register(codec, i)) continue; count += sprintf(buf + count, "%2x: ", i); @@ -2029,8 +2029,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) */ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { - if (codec->driver->volatile_register) - return codec->driver->volatile_register(codec, reg); + if (codec->volatile_register) + return codec->volatile_register(codec, reg); else return 0; } @@ -3489,6 +3489,8 @@ int snd_soc_register_codec(struct device *dev, codec->write = codec_drv->write; codec->read = codec_drv->read; + codec->volatile_register = codec_drv->volatile_register; + codec->readable_register = codec_drv->readable_register; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; @@ -3517,6 +3519,13 @@ int snd_soc_register_codec(struct device *dev, } } + if (codec_drv->reg_access_size && codec_drv->reg_access_default) { + if (!codec->volatile_register) + codec->volatile_register = snd_soc_default_volatile_register; + if (!codec->readable_register) + codec->readable_register = snd_soc_default_readable_register; + } + for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- cgit v1.2.3 From 4e10bda05d6c7d4aba509bbbab5ba748d54c702f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 13 Jan 2011 22:48:52 +0530 Subject: ASoC: soc core add inline to handle card list initialzation Currently the soc_probe initializes the card hence it does the card list initialzation. But if machines directly register the card they would need to do these steps, so putting them as inline would save lot of code This patch adds an inline to do list initialzation Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Signed-off-by: Mark Brown --- include/sound/soc.h | 10 ++++++++++ sound/soc/soc-core.c | 7 +------ 2 files changed, 11 insertions(+), 6 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index accb8a16c165..541ddfaa1243 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -779,6 +779,16 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) return dev_get_drvdata(&rtd->dev); } +static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) +{ + INIT_LIST_HEAD(&card->dai_dev_list); + INIT_LIST_HEAD(&card->codec_dev_list); + INIT_LIST_HEAD(&card->platform_dev_list); + INIT_LIST_HEAD(&card->widgets); + INIT_LIST_HEAD(&card->paths); + INIT_LIST_HEAD(&card->dapm_list); +} + #include #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9c5e7cff3f01..83057127b2fa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1872,12 +1872,7 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - INIT_LIST_HEAD(&card->dai_dev_list); - INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); - INIT_LIST_HEAD(&card->widgets); - INIT_LIST_HEAD(&card->paths); - INIT_LIST_HEAD(&card->dapm_list); + snd_soc_initialize_card_lists(card); ret = snd_soc_register_card(card); if (ret != 0) { -- cgit v1.2.3 From 70a7ca34dbdcc6f0ed332baf2b308bab2871424a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 14 Jan 2011 19:22:48 +0530 Subject: ASoC: soc core allow machine driver to register the card The machine driver can't register the card directly and need to do this thru soc-audio device creation This patch allows the register and unregister card to be directly called by machine drivers Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 10 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 541ddfaa1243..9952254974b3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -258,6 +258,8 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_register_card(struct snd_soc_card *card); +int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83057127b2fa..69117b686fdc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -58,8 +58,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int snd_soc_register_card(struct snd_soc_card *card); -static int snd_soc_unregister_card(struct snd_soc_card *card); static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* @@ -1870,6 +1868,13 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; + /* + * no card, so machine driver should be registering card + * we should not be here in that case so ret error + */ + if (!card) + return -EINVAL; + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; snd_soc_initialize_card_lists(card); @@ -3105,11 +3110,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); * * @card: Card to register * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_register_card(struct snd_soc_card *card) +int snd_soc_register_card(struct snd_soc_card *card) { int i; @@ -3141,17 +3143,15 @@ static int snd_soc_register_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core * * @card: Card to unregister * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_unregister_card(struct snd_soc_card *card) +int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); @@ -3162,6 +3162,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_unregister_card); /* * Simplify DAI link configuration by removing ".-1" from device names -- cgit v1.2.3 From 20e4859dedfc7e7b620d1756b29f8483c5be5fcc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 15 Jan 2011 13:40:50 +0000 Subject: ASoC: Add support for sequencing within With larger devices there may be many widgets of the same type in series in an audio path. Allow drivers to specify an additional level of ordering within each widget type by adding a subsequence number to widgets and then splitting operations on widgets so that widgets of the same type but different sequence numbers are processed separately. A typical example would be a supply widget which requires that another widget be enabled to provide power or clocking. SND_SOC_DAPM_PGA_S() and SND_SOC_DAPM_SUPPLY_S() macros are provided allowing this to be used with PGAs and supplies as these are the most commonly affected widgets. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 13 +++++++++++++ sound/soc/soc-dapm.c | 11 ++++++++++- 2 files changed, 23 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8031769ac485..a3760c93a8a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -157,6 +157,18 @@ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} +/* additional sequencing control within an event type */ +#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, wcontrols, \ + wncontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ + .event = wevent, .event_flags = wflags, .subseq = wsubseq} +#define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \ + wflags) \ +{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .event = wevent, \ + .event_flags = wflags, .subseq = wsubseq} + /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ #define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ @@ -450,6 +462,7 @@ struct snd_soc_dapm_widget { unsigned char ext:1; /* has external widgets */ unsigned char force:1; /* force state */ unsigned char ignore_suspend:1; /* kept enabled over suspend */ + int subseq; /* sort within widget type */ int (*power_check)(struct snd_soc_dapm_widget *w); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 57e1c9f71149..eb7436c7acad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -737,6 +737,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; + if (a->subseq != b->subseq) { + if (power_up) + return a->subseq - b->subseq; + else + return b->subseq - a->subseq; + } if (a->reg != b->reg) return a->reg - b->reg; if (a->dapm != b->dapm) @@ -869,6 +875,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); int cur_sort = -1; + int cur_subseq = -1; int cur_reg = SND_SOC_NOPM; struct snd_soc_dapm_context *cur_dapm = NULL; int ret; @@ -884,12 +891,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, /* Do we need to apply any queued changes? */ if (sort[w->id] != cur_sort || w->reg != cur_reg || - w->dapm != cur_dapm) { + w->dapm != cur_dapm || w->subseq != cur_subseq) { if (!list_empty(&pending)) dapm_seq_run_coalesced(cur_dapm, &pending); INIT_LIST_HEAD(&pending); cur_sort = -1; + cur_subseq = -1; cur_reg = SND_SOC_NOPM; cur_dapm = NULL; } @@ -934,6 +942,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, default: /* Queue it up for application */ cur_sort = sort[w->id]; + cur_subseq = w->subseq; cur_reg = w->reg; cur_dapm = w->dapm; list_move(&w->power_list, &pending); -- cgit v1.2.3 From 474b62d6eee733abdcd36f8e3e5ce504fbb9110b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Jan 2011 16:14:44 +0000 Subject: ASoC: Provide per widget type callback when executing DAPM sequences Many modern devices have features such as DC servos which take time to start. Currently these are handled by per-widget events but this makes it difficult to paralleise operations on multiple widgets, meaning delays can end up being needlessly serialised. By providing a callback to drivers when all widgets of a given type have been handled during a DAPM sequence the core allows drivers to start operations separately and wait for them to complete much more simply. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 3 +++ include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 16 +++++++++++++++- 4 files changed, 22 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a3760c93a8a3..6c9ae237814b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -500,6 +500,9 @@ struct snd_soc_dapm_context { struct snd_soc_dapm_update *update; + void (*seq_notifier)(struct snd_soc_dapm_context *, + enum snd_soc_dapm_type); + struct device *dev; /* from parent - for debug */ struct snd_soc_codec *codec; /* parent codec */ struct snd_soc_card *card; /* parent card */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 9952254974b3..d244f9013767 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -546,6 +546,9 @@ struct snd_soc_codec_driver { /* codec bias level */ int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); + + void (*seq_notifier)(struct snd_soc_dapm_context *, + enum snd_soc_dapm_type); }; /* SoC platform interface */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9e68984423b2..b0e7689159c1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3496,6 +3496,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; + codec->dapm.seq_notifier = codec_drv->seq_notifier; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index eb7436c7acad..37b376f4c75d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -878,7 +878,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, int cur_subseq = -1; int cur_reg = SND_SOC_NOPM; struct snd_soc_dapm_context *cur_dapm = NULL; - int ret; + int ret, i; int *sort; if (power_up) @@ -895,6 +895,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, if (!list_empty(&pending)) dapm_seq_run_coalesced(cur_dapm, &pending); + if (cur_dapm && cur_dapm->seq_notifier) { + for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) + if (sort[i] == cur_sort) + cur_dapm->seq_notifier(cur_dapm, + i); + } + INIT_LIST_HEAD(&pending); cur_sort = -1; cur_subseq = -1; @@ -956,6 +963,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, if (!list_empty(&pending)) dapm_seq_run_coalesced(dapm, &pending); + + if (cur_dapm && cur_dapm->seq_notifier) { + for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) + if (sort[i] == cur_sort) + cur_dapm->seq_notifier(cur_dapm, + i); + } } static void dapm_widget_update(struct snd_soc_dapm_context *dapm) -- cgit v1.2.3 From dad8e7aeeb83a26d267e757e4c1cf69591850477 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 19 Jan 2011 14:53:36 +0000 Subject: ASoC: soc-cache: Introduce the cache_bypass option This is primarily needed to avoid writing back to the cache whenever we are syncing the cache with the hardware. This gives a performance benefit especially for large register maps. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 36 ++++++++++++++++++++++++------------ 2 files changed, 25 insertions(+), 12 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index d244f9013767..c184f84a354c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -488,6 +488,7 @@ struct snd_soc_codec { /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; + unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int cache_only:1; /* Suppress writes to hardware */ unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ unsigned int suspended:1; /* Codec is in suspend PM state */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d97a59f6a249..1ebff9f12b4e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -25,7 +25,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -49,7 +50,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -106,7 +108,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -130,7 +133,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -191,7 +195,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -216,7 +221,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -271,7 +277,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -295,7 +302,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -450,7 +458,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -476,7 +485,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -568,7 +578,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -595,7 +606,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[3] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; -- cgit v1.2.3 From 7cfe56172ac14d2031f1896ecb629033f71caafa Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 20 Jan 2011 13:52:08 -0700 Subject: ASoC: wm8903: Expose GPIOs through gpiolib Also, update platform_data GPIO handling to have an explicit "don't touch this pin" option. Add #defines for the GPIO pin functions. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/wm8903.h | 20 +++++++- sound/soc/codecs/wm8903.c | 126 +++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 144 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h index b4a0db2307ef..86172cf4339f 100644 --- a/include/sound/wm8903.h +++ b/include/sound/wm8903.h @@ -36,6 +36,21 @@ #define WM8903_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ #define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ +/* + * WM8903_GPn_FN values + * + * See datasheets for list of valid values per pin + */ +#define WM8903_GPn_FN_GPIO_OUTPUT 0 +#define WM8903_GPn_FN_BCLK 1 +#define WM8903_GPn_FN_IRQ_OUTPT 2 +#define WM8903_GPn_FN_GPIO_INPUT 3 +#define WM8903_GPn_FN_MICBIAS_CURRENT_DETECT 4 +#define WM8903_GPn_FN_MICBIAS_SHORT_DETECT 5 +#define WM8903_GPn_FN_DMIC_LR_CLK_OUTPUT 6 +#define WM8903_GPn_FN_FLL_LOCK_OUTPUT 8 +#define WM8903_GPn_FN_FLL_CLOCK_OUTPUT 9 + /* * R116 (0x74) - GPIO Control 1 */ @@ -231,6 +246,8 @@ #define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ #define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ +#define WM8903_NUM_GPIO 5 + struct wm8903_platform_data { bool irq_active_low; /* Set if IRQ active low, default high */ @@ -243,7 +260,8 @@ struct wm8903_platform_data { int micdet_delay; /* Delay after microphone detection (ms) */ - u32 gpio_cfg[5]; /* Default register values for GPIO pin mux */ + int gpio_base; + u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ }; #endif diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a2a446cb1807..9c4f2c4febc2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2,6 +2,7 @@ * wm8903.c -- WM8903 ALSA SoC Audio driver * * Copyright 2008 Wolfson Microelectronics + * Copyright 2011 NVIDIA, Inc. * * Author: Mark Brown * @@ -19,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -213,6 +215,7 @@ static u16 wm8903_reg_defaults[] = { }; struct wm8903_priv { + struct snd_soc_codec *codec; int sysclk; int irq; @@ -230,6 +233,10 @@ struct wm8903_priv { int mic_short; int mic_last_report; int mic_delay; + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1635,6 +1642,119 @@ static int wm8903_resume(struct snd_soc_codec *codec) return 0; } +#ifdef CONFIG_GPIOLIB +static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip) +{ + return container_of(chip, struct wm8903_priv, gpio_chip); +} + +static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) +{ + if (offset >= WM8903_NUM_GPIO) + return -EINVAL; + + return 0; +} + +static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + unsigned int mask, val; + + mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; + val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | + WM8903_GP1_DIR; + + return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); +} + +static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + int reg; + + reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset); + + return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; +} + +static int wm8903_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + unsigned int mask, val; + + mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; + val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | + (value << WM8903_GP2_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); +} + +static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + + snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); +} + +static struct gpio_chip wm8903_template_chip = { + .label = "wm8903", + .owner = THIS_MODULE, + .request = wm8903_gpio_request, + .direction_input = wm8903_gpio_direction_in, + .get = wm8903_gpio_get, + .direction_output = wm8903_gpio_direction_out, + .set = wm8903_gpio_set, + .can_sleep = 1, +}; + +static void wm8903_init_gpio(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); + int ret; + + wm8903->gpio_chip = wm8903_template_chip; + wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; + wm8903->gpio_chip.dev = codec->dev; + + if (pdata && pdata->gpio_base) + wm8903->gpio_chip.base = pdata->gpio_base; + else + wm8903->gpio_chip.base = -1; + + ret = gpiochip_add(&wm8903->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm8903_free_gpio(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm8903->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm8903_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm8903_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); @@ -1643,6 +1763,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) int trigger, irq_pol; u16 val; + wm8903->codec = codec; init_completion(&wm8903->wseq); ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); @@ -1667,7 +1788,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* Set up GPIOs and microphone detection */ if (pdata) { for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (!pdata->gpio_cfg[i]) + if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, @@ -1749,12 +1870,15 @@ static int wm8903_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(codec); + wm8903_init_gpio(codec); + return ret; } /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { + wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- cgit v1.2.3 From 67b22517d8e48a97e1d2ab10d095c538bbb2374c Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Wed, 19 Jan 2011 21:22:06 +0300 Subject: ASoC: CS4271 codec support Added support for CS4271 codec to ASoC. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/cs4271.h | 25 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4271.c | 630 ++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 661 insertions(+) create mode 100644 include/sound/cs4271.h create mode 100644 sound/soc/codecs/cs4271.c (limited to 'include') diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h new file mode 100644 index 000000000000..16f8d325d3dc --- /dev/null +++ b/include/sound/cs4271.h @@ -0,0 +1,25 @@ +/* + * Definitions for CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __CS4271_H +#define __CS4271_H + +struct cs4271_platform_data { + int gpio_nreset; /* GPIO driving Reset pin, if any */ + int gpio_disable; /* GPIO that disable serial bus, if any */ +}; + +#endif /* __CS4271_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a9cb2a04ad56..e239345a4d5d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 @@ -157,6 +158,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CS4271 + tristate + config SND_SOC_CX20442 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 68e76af894b9..83b7accd7037 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o @@ -93,6 +94,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c new file mode 100644 index 000000000000..237ece3f1046 --- /dev/null +++ b/sound/soc/codecs/cs4271.c @@ -0,0 +1,630 @@ +/* + * CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * CS4271 registers + * High byte represents SPI chip address (0x10) + write command (0) + * Low byte - codec register address + */ +#define CS4271_MODE1 0x2001 /* Mode Control 1 */ +#define CS4271_DACCTL 0x2002 /* DAC Control */ +#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x2006 /* ADC Control */ +#define CS4271_MODE2 0x2007 /* Mode Control 2 */ +#define CS4271_CHIPID 0x2008 /* Chip ID */ + +#define CS4271_FIRSTREG CS4271_MODE1 +#define CS4271_LASTREG CS4271_MODE2 +#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1) + +/* Bit masks for the CS4271 registers */ +#define CS4271_MODE1_MODE_MASK 0xC0 +#define CS4271_MODE1_MODE_1X 0x00 +#define CS4271_MODE1_MODE_2X 0x80 +#define CS4271_MODE1_MODE_4X 0xC0 + +#define CS4271_MODE1_DIV_MASK 0x30 +#define CS4271_MODE1_DIV_1 0x00 +#define CS4271_MODE1_DIV_15 0x10 +#define CS4271_MODE1_DIV_2 0x20 +#define CS4271_MODE1_DIV_3 0x30 + +#define CS4271_MODE1_MASTER 0x08 + +#define CS4271_MODE1_DAC_DIF_MASK 0x07 +#define CS4271_MODE1_DAC_DIF_LJ 0x00 +#define CS4271_MODE1_DAC_DIF_I2S 0x01 +#define CS4271_MODE1_DAC_DIF_RJ16 0x02 +#define CS4271_MODE1_DAC_DIF_RJ24 0x03 +#define CS4271_MODE1_DAC_DIF_RJ20 0x04 +#define CS4271_MODE1_DAC_DIF_RJ18 0x05 + +#define CS4271_DACCTL_AMUTE 0x80 +#define CS4271_DACCTL_IF_SLOW 0x40 + +#define CS4271_DACCTL_DEM_MASK 0x30 +#define CS4271_DACCTL_DEM_DIS 0x00 +#define CS4271_DACCTL_DEM_441 0x10 +#define CS4271_DACCTL_DEM_48 0x20 +#define CS4271_DACCTL_DEM_32 0x30 + +#define CS4271_DACCTL_SVRU 0x08 +#define CS4271_DACCTL_SRD 0x04 +#define CS4271_DACCTL_INVA 0x02 +#define CS4271_DACCTL_INVB 0x01 + +#define CS4271_DACVOL_BEQUA 0x40 +#define CS4271_DACVOL_SOFT 0x20 +#define CS4271_DACVOL_ZEROC 0x10 + +#define CS4271_DACVOL_ATAPI_MASK 0x0F +#define CS4271_DACVOL_ATAPI_M_M 0x00 +#define CS4271_DACVOL_ATAPI_M_BR 0x01 +#define CS4271_DACVOL_ATAPI_M_BL 0x02 +#define CS4271_DACVOL_ATAPI_M_BLR2 0x03 +#define CS4271_DACVOL_ATAPI_AR_M 0x04 +#define CS4271_DACVOL_ATAPI_AR_BR 0x05 +#define CS4271_DACVOL_ATAPI_AR_BL 0x06 +#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07 +#define CS4271_DACVOL_ATAPI_AL_M 0x08 +#define CS4271_DACVOL_ATAPI_AL_BR 0x09 +#define CS4271_DACVOL_ATAPI_AL_BL 0x0A +#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B +#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C +#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D +#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E +#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F + +#define CS4271_VOLA_MUTE 0x80 +#define CS4271_VOLA_VOL_MASK 0x7F +#define CS4271_VOLB_MUTE 0x80 +#define CS4271_VOLB_VOL_MASK 0x7F + +#define CS4271_ADCCTL_DITHER16 0x20 + +#define CS4271_ADCCTL_ADC_DIF_MASK 0x10 +#define CS4271_ADCCTL_ADC_DIF_LJ 0x00 +#define CS4271_ADCCTL_ADC_DIF_I2S 0x10 + +#define CS4271_ADCCTL_MUTEA 0x08 +#define CS4271_ADCCTL_MUTEB 0x04 +#define CS4271_ADCCTL_HPFDA 0x02 +#define CS4271_ADCCTL_HPFDB 0x01 + +#define CS4271_MODE2_LOOP 0x10 +#define CS4271_MODE2_MUTECAEQUB 0x08 +#define CS4271_MODE2_FREEZE 0x04 +#define CS4271_MODE2_CPEN 0x02 +#define CS4271_MODE2_PDN 0x01 + +#define CS4271_CHIPID_PART_MASK 0xF0 +#define CS4271_CHIPID_REV_MASK 0x0F + +/* + * Default CS4271 power-up configuration + * Array contains non-existing in hw register at address 0 + * Array do not include Chip ID, as codec driver does not use + * registers read operations at all + */ +static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { + 0, + 0, + CS4271_DACCTL_AMUTE, + CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, + 0, + 0, + 0, + 0, +}; + +struct cs4271_private { + /* SND_SOC_I2C or SND_SOC_SPI */ + enum snd_soc_control_type bus_type; + void *control_data; + unsigned int mclk; + bool master; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; + /* GPIO that disable serial bus, if any */ + int gpio_disable; +}; + +struct cs4271_clk_cfg { + unsigned int ratio; /* MCLK / sample rate */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + u8 mclk_master; /* ratio bit mask for Master mode */ + u8 mclk_slave; /* ratio bit mask for Slave mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, + {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, + {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + +/* + * @freq is the desired MCLK rate + * MCLK rate should (c) be the sample rate, multiplied by one of the + * ratios listed in cs4271_mclk_fs_ratios table + */ +static int cs4271_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mclk = freq; + return 0; +} + +static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4271->master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4271->master = 1; + val |= CS4271_MODE1_MASTER; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val |= CS4271_MODE1_DAC_DIF_LJ; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + break; + case SND_SOC_DAIFMT_I2S: + val |= CS4271_MODE1_DAC_DIF_I2S; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); + + return 0; +} + +static int cs4271_deemph[] = {0, 44100, 48000, 32000}; + +static int cs4271_set_deemph(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int i; + int val = CS4271_DACCTL_DEM_DIS; + + if (cs4271->deemph) { + /* Find closest de-emphasis freq */ + val = 1; + for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++) + if (abs(cs4271_deemph[i] - cs4271->rate) < + abs(cs4271_deemph[val] - cs4271->rate)) + val = i; + val <<= 4; + } + + return snd_soc_update_bits(codec, CS4271_DACCTL, + CS4271_DACCTL_DEM_MASK, val); +} + +static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = cs4271->deemph; + return 0; +} + +static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->deemph = ucontrol->value.enumerated.item[0]; + return cs4271_set_deemph(codec); +} + +static int cs4271_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int i, ratio, val; + + cs4271->rate = params_rate(params); + ratio = cs4271->mclk / cs4271->rate; + for (i = 0; i < CS4171_NR_RATIOS; i++) + if (cs4271_clk_tab[i].ratio == ratio) + break; + + if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + /* Configure DAC */ + val = cs4271_clk_tab[i].speed_mode; + + if (cs4271->master) + val |= cs4271_clk_tab[i].mclk_master; + else + val |= cs4271_clk_tab[i].mclk_slave; + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + + return cs4271_set_deemph(codec); +} + +static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int val_a = 0; + int val_b = 0; + + if (mute) { + val_a = CS4271_VOLA_MUTE; + val_b = CS4271_VOLB_MUTE; + } + + snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + return 0; +} + +/* CS4271 controls */ +static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0); + +static const struct snd_kcontrol_new cs4271_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB, + 0, 0x7F, 1, cs4271_dac_tlv), + SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + cs4271_get_deemph, cs4271_put_deemph), + SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0), + SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0), + SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0), + SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0), + SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1), + SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0), + SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1), + SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB, + 7, 1, 1), +}; + +static struct snd_soc_dai_ops cs4271_dai_ops = { + .hw_params = cs4271_hw_params, + .set_sysclk = cs4271_set_dai_sysclk, + .set_fmt = cs4271_set_dai_fmt, + .digital_mute = cs4271_digital_mute, +}; + +struct snd_soc_dai_driver cs4271_dai = { + .name = "cs4271-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .ops = &cs4271_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + /* Set power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + return 0; +} + +static int cs4271_soc_resume(struct snd_soc_codec *codec) +{ + /* Restore codec state */ + snd_soc_cache_sync(codec); + /* then disable the power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + return 0; +} +#else +#define cs4271_soc_suspend NULL +#define cs4271_soc_resume NULL +#endif /* CONFIG_PM */ + +static int cs4271_probe(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; + int ret; + int gpio_nreset = -EINVAL; + int gpio_disable = -EINVAL; + + codec->control_data = cs4271->control_data; + + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + if (gpio_is_valid(cs4271plat->gpio_disable)) + gpio_disable = cs4271plat->gpio_disable; + } + + if (gpio_disable >= 0) + if (gpio_request(gpio_disable, "CS4271 Disable")) + gpio_disable = -EINVAL; + if (gpio_disable >= 0) + gpio_direction_output(gpio_disable, 0); + + if (gpio_nreset >= 0) + if (gpio_request(gpio_nreset, "CS4271 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + } + + cs4271->gpio_nreset = gpio_nreset; + cs4271->gpio_disable = gpio_disable; + + /* + * In case of I2C, chip address specified in board data. + * So cache IO operations use 8 bit codec register address. + * In case of SPI, chip address and register address + * passed together as 16 bit value. + * Anyway, register address is masked with 0xFF inside + * soc-cache code. + */ + if (cs4271->bus_type == SND_SOC_SPI) + ret = snd_soc_codec_set_cache_io(codec, 16, 8, + cs4271->bus_type); + else + ret = snd_soc_codec_set_cache_io(codec, 8, 8, + cs4271->bus_type); + if (ret) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + snd_soc_update_bits(codec, CS4271_MODE2, 0, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + /* Power-up sequence requires 85 uS */ + udelay(85); + + return snd_soc_add_controls(codec, cs4271_snd_controls, + ARRAY_SIZE(cs4271_snd_controls)); +} + +static int cs4271_remove(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int gpio_nreset, gpio_disable; + + gpio_nreset = cs4271->gpio_nreset; + gpio_disable = cs4271->gpio_disable; + + if (gpio_is_valid(gpio_nreset)) { + /* Set codec to the reset state */ + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + } + + if (gpio_is_valid(gpio_disable)) + gpio_free(gpio_disable); + + return 0; +}; + +struct snd_soc_codec_driver soc_codec_dev_cs4271 = { + .probe = cs4271_probe, + .remove = cs4271_remove, + .suspend = cs4271_soc_suspend, + .resume = cs4271_soc_resume, + .reg_cache_default = cs4271_dflt_reg, + .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), + .reg_word_size = sizeof(cs4271_dflt_reg[0]), + .compress_type = SND_SOC_FLAT_COMPRESSION, +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + spi_set_drvdata(spi, cs4271); + cs4271->control_data = spi; + cs4271->bus_type = SND_SOC_SPI; + + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .probe = cs4271_spi_probe, + .remove = __devexit_p(cs4271_spi_remove), +}; +#endif /* defined(CONFIG_SPI_MASTER) */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_device_id cs4271_i2c_id[] = { + {"cs4271", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static int __devinit cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + i2c_set_clientdata(client, cs4271); + cs4271->control_data = client; + cs4271->bus_type = SND_SOC_I2C; + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .id_table = cs4271_i2c_id, + .probe = cs4271_i2c_probe, + .remove = __devexit_p(cs4271_i2c_remove), +}; +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +/* + * We only register our serial bus driver here without + * assignment to particular chip. So if any of the below + * fails, there is some problem with I2C or SPI subsystem. + * In most cases this module will be compiled with support + * of only one serial bus. + */ +static int __init cs4271_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&cs4271_i2c_driver); + if (ret) { + pr_err("Failed to register CS4271 I2C driver: %d\n", ret); + return ret; + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4271_spi_driver); + if (ret) { + pr_err("Failed to register CS4271 SPI driver: %d\n", ret); + return ret; + } +#endif + + return 0; +} +module_init(cs4271_modinit); + +static void __exit cs4271_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4271_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&cs4271_i2c_driver); +#endif +} +module_exit(cs4271_modexit); + +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From c358e640a669b528b32af5442c92b856de623e1c Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 21 Jan 2011 15:29:02 +0000 Subject: ASoC: soc-cache: Add trace event for snd_soc_cache_sync() This patch makes it easy to see when the syncing process begins and ends. You can also enable the snd_soc_reg_write tracepoint to see which registers are being synced. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/trace/events/asoc.h | 25 +++++++++++++++++++++++++ sound/soc/soc-cache.c | 10 ++++++++++ 2 files changed, 35 insertions(+) (limited to 'include') diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 186e84db4b54..ae973d2e27a1 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -229,6 +229,31 @@ TRACE_EVENT(snd_soc_jack_notify, TP_printk("jack=%s %x", __get_str(name), (int)__entry->val) ); +TRACE_EVENT(snd_soc_cache_sync, + + TP_PROTO(struct snd_soc_codec *codec, const char *type, + const char *status), + + TP_ARGS(codec, type, status), + + TP_STRUCT__entry( + __string( name, codec->name ) + __string( status, status ) + __string( type, type ) + __field( int, id ) + ), + + TP_fast_assign( + __assign_str(name, codec->name); + __assign_str(status, status); + __assign_str(type, type); + __entry->id = codec->id; + ), + + TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), + (int)__entry->id, __get_str(type), __get_str(status)) +); + #endif /* _TRACE_ASOC_H */ /* This part must be outside protection */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f83483963791..db66dc44add2 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include #include +#include + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -1601,18 +1603,26 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); int snd_soc_cache_sync(struct snd_soc_codec *codec) { int ret; + const char *name; if (!codec->cache_sync) { return 0; } + if (codec->cache_ops->name) + name = codec->cache_ops->name; + else + name = "unknown"; + if (codec->cache_ops && codec->cache_ops->sync) { if (codec->cache_ops->name) dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); ret = codec->cache_ops->sync(codec); if (!ret) codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); return ret; } -- cgit v1.2.3 From 4d805f7b6607f6e547dc22e5d57c201e43d21c05 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Jan 2011 11:46:02 +0900 Subject: ASoC: sh: fsi: Add snd_soc_dai_set_fmt support This patch add snd_soc_dai_ops :: set_fmt to FSI driver and select master/slave clock mode by snd_soc_dai_set_fmt on fsi-xxx.c instead of platform infomation code. This patch remove fsi_is_master function which is no longer needed. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Acked-by: Paul Mundt Signed-off-by: Mark Brown --- arch/arm/mach-shmobile/board-ag5evm.c | 4 +-- arch/arm/mach-shmobile/board-ap4evb.c | 2 -- arch/arm/mach-shmobile/board-mackerel.c | 2 -- arch/sh/boards/mach-ecovec24/setup.c | 2 -- arch/sh/boards/mach-se/7724/setup.c | 2 -- include/sound/sh_fsi.h | 8 +---- sound/soc/sh/fsi-ak4642.c | 13 +++++--- sound/soc/sh/fsi-da7210.c | 12 ++++++-- sound/soc/sh/fsi-hdmi.c | 11 +++++++ sound/soc/sh/fsi.c | 54 +++++++++++++++++++-------------- 10 files changed, 63 insertions(+), 47 deletions(-) (limited to 'include') diff --git a/arch/arm/mach-shmobile/board-ag5evm.c b/arch/arm/mach-shmobile/board-ag5evm.c index c18a740a4159..9ee55e0fbeb1 100644 --- a/arch/arm/mach-shmobile/board-ag5evm.c +++ b/arch/arm/mach-shmobile/board-ag5evm.c @@ -119,9 +119,7 @@ static struct platform_device keysc_device = { /* FSI A */ static struct sh_fsi_platform_info fsi_info = { - .porta_flags = SH_FSI_OUT_SLAVE_MODE | - SH_FSI_IN_SLAVE_MODE | - SH_FSI_OFMT(I2S) | + .porta_flags = SH_FSI_OFMT(I2S) | SH_FSI_IFMT(I2S), }; diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 3cf0951caa2d..d503a74e30e4 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -674,8 +674,6 @@ static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) static struct sh_fsi_platform_info fsi_info = { .porta_flags = SH_FSI_BRS_INV | - SH_FSI_OUT_SLAVE_MODE | - SH_FSI_IN_SLAVE_MODE | SH_FSI_OFMT(PCM) | SH_FSI_IFMT(PCM), diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 7b15d21f0f68..425962d5b29c 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -611,8 +611,6 @@ fsi_set_rate_end: static struct sh_fsi_platform_info fsi_info = { .porta_flags = SH_FSI_BRS_INV | - SH_FSI_OUT_SLAVE_MODE | - SH_FSI_IN_SLAVE_MODE | SH_FSI_OFMT(PCM) | SH_FSI_IFMT(PCM), diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 33b662999fc6..037416f346cf 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -724,8 +724,6 @@ static struct platform_device camera_devices[] = { /* FSI */ static struct sh_fsi_platform_info fsi_info = { .portb_flags = SH_FSI_BRS_INV | - SH_FSI_OUT_SLAVE_MODE | - SH_FSI_IN_SLAVE_MODE | SH_FSI_OFMT(I2S) | SH_FSI_IFMT(I2S), }; diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 527679394a25..b4aef05dd8b5 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -287,8 +287,6 @@ static struct platform_device ceu1_device = { /* change J20, J21, J22 pin to 1-2 connection to use slave mode */ static struct sh_fsi_platform_info fsi_info = { .porta_flags = SH_FSI_BRS_INV | - SH_FSI_OUT_SLAVE_MODE | - SH_FSI_IN_SLAVE_MODE | SH_FSI_OFMT(PCM) | SH_FSI_IFMT(PCM), }; diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index d79894192ae3..18e43279f70f 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -17,12 +17,11 @@ /* flags format - * 0xABCDEEFF + * 0xABC0EEFF * * A: channel size for TDM (input) * B: channel size for TDM (ooutput) * C: inversion - * D: mode * E: input format * F: output format */ @@ -46,11 +45,6 @@ #define SH_FSI_LRS_INV (1 << 22) #define SH_FSI_BRS_INV (1 << 23) -/* mode */ -#define SH_FSI_MODE_MASK 0x000F0000 -#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ -#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ - /* DI format */ #define SH_FSI_FMT_MASK 0x000000FF #define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index a722a4c661ff..ce058c749e6a 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -23,15 +23,20 @@ struct fsi_ak4642_data { static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(dai, 0, 11289600, 0); + ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); return ret; } diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index e8df9da92f71..9b24ed466ab1 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -15,11 +15,19 @@ static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; + int ret; - return snd_soc_dai_set_fmt(dai, + ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); + + return ret; } static struct snd_soc_dai_link fsi_da7210_dai = { diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c index a52dd8ec71d3..96d8ce3f3211 100644 --- a/sound/soc/sh/fsi-hdmi.c +++ b/sound/soc/sh/fsi-hdmi.c @@ -12,6 +12,16 @@ #include #include +static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *cpu = rtd->cpu_dai; + int ret; + + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM); + + return ret; +} + static struct snd_soc_dai_link fsi_dai_link = { .name = "HDMI", .stream_name = "HDMI", @@ -19,6 +29,7 @@ static struct snd_soc_dai_link fsi_dai_link = { .codec_dai_name = "sh_mobile_hdmi-hifi", .platform_name = "sh_fsi2", .codec_name = "sh-mobile-hdmi", + .init = fsi_hdmi_dai_init, }; static struct snd_soc_card fsi_soc_card = { diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1d0a16e80919..5f39f364effd 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -78,6 +78,8 @@ /* CKG1 */ #define ACKMD_MASK 0x00007000 #define BPFMD_MASK 0x00000700 +#define DIMD (1 << 4) +#define DOMD (1 << 0) /* A/B MST_CTLR */ #define BP (1 << 4) /* Fix the signal of Biphase output */ @@ -292,21 +294,6 @@ static inline struct fsi_stream *fsi_get_stream(struct fsi_priv *fsi, return is_play ? &fsi->playback : &fsi->capture; } -static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) -{ - u32 mode; - u32 flags = fsi_get_info_flags(fsi); - - mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; - - /* return - * 1 : master mode - * 0 : slave mode - */ - - return (mode & flags) != mode; -} - static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play) { int is_porta = fsi_is_port_a(fsi); @@ -764,19 +751,11 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, u32 fmt; u32 data; int is_play = fsi_is_play(substream); - int is_master; io = fsi_get_stream(fsi, is_play); pm_runtime_get_sync(dai->dev); - /* CKG1 */ - data = is_play ? (1 << 0) : (1 << 4); - is_master = fsi_is_master_mode(fsi, is_play); - if (is_master) - fsi_reg_mask_set(fsi, CKG1, data, data); - else - fsi_reg_mask_set(fsi, CKG1, data, 0); /* clock inversion (CKG2) */ data = 0; @@ -893,6 +872,34 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); + u32 data = 0; + int ret; + + pm_runtime_get_sync(dai->dev); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + data = DIMD | DOMD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + ret = -EINVAL; + goto set_fmt_exit; + } + fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); + ret = 0; + +set_fmt_exit: + pm_runtime_put_sync(dai->dev); + + return ret; +} + static int fsi_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -979,6 +986,7 @@ static struct snd_soc_dai_ops fsi_dai_ops = { .startup = fsi_dai_startup, .shutdown = fsi_dai_shutdown, .trigger = fsi_dai_trigger, + .set_fmt = fsi_dai_set_fmt, .hw_params = fsi_dai_hw_params, }; -- cgit v1.2.3 From 181e055e6bed80afbf8ba2bb5e3ce84fbd3f633c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Jan 2011 14:05:25 +0000 Subject: ASoC: Fix type for snd_soc_volatile_register() We generally refer to registers as unsigned ints (including in the underlying CODEC driver operation). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index c184f84a354c..1355ef029d82 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -267,7 +267,8 @@ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); -int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b0e7689159c1..14861f95f629 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2029,7 +2029,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) * * Boolean function indiciating if a CODEC register is volatile. */ -int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { if (codec->volatile_register) return codec->volatile_register(codec, reg); -- cgit v1.2.3 From 3d23c73fa0a47e8aecd2a4d8f280f45f6f7611a1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Jan 2011 21:51:25 +0000 Subject: ASoC: Remove controls from sequenced PGA arguments We have zero users for PGA controls and the core support for them was removed a while ago so no point in cut'n'pasting them into new macros, even if it's too much hassle to update the existing ones. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6c9ae237814b..6a25e6993859 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -158,11 +158,11 @@ .event = wevent, .event_flags = wflags} /* additional sequencing control within an event type */ -#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, wcontrols, \ - wncontrols, wevent, wflags) \ +#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \ + wevent, wflags) \ { .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ - .event = wevent, .event_flags = wflags, .subseq = wsubseq} + .invert = winvert, .event = wevent, .event_flags = wflags, \ + .subseq = wsubseq} #define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \ wflags) \ { .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ -- cgit v1.2.3 From f17c13ca52d5c5a6a164536244a6debb8cd17983 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jan 2011 10:43:19 +0900 Subject: ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format Current format selection of FSI-codecs depended on platform information for FSI, and chip default settings for codecs. It is not understandable/formal method. This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt. But FSI can use I2S/PCM and SPDIF format today. It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF. So, this patch change FSI platform information to have DAI/SPDIF mode. If platform selects DAI mode (default), FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt, and if it is SPDIF mode, FSI become SPDIF format. Signed-off-by: Kuninori Morimoto Acked-by: Paul Mundt Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- arch/arm/mach-shmobile/board-ag5evm.c | 8 --- arch/arm/mach-shmobile/board-ap4evb.c | 6 +- arch/arm/mach-shmobile/board-mackerel.c | 6 +- arch/sh/boards/mach-ecovec24/setup.c | 4 +- arch/sh/boards/mach-se/7724/setup.c | 4 +- include/sound/sh_fsi.h | 70 ++++++--------------- sound/soc/sh/fsi-ak4642.c | 3 +- sound/soc/sh/fsi-da7210.c | 3 +- sound/soc/sh/fsi.c | 106 +++++++++++++++++--------------- 9 files changed, 84 insertions(+), 126 deletions(-) (limited to 'include') diff --git a/arch/arm/mach-shmobile/board-ag5evm.c b/arch/arm/mach-shmobile/board-ag5evm.c index 9ee55e0fbeb1..343362d02075 100644 --- a/arch/arm/mach-shmobile/board-ag5evm.c +++ b/arch/arm/mach-shmobile/board-ag5evm.c @@ -118,11 +118,6 @@ static struct platform_device keysc_device = { }; /* FSI A */ -static struct sh_fsi_platform_info fsi_info = { - .porta_flags = SH_FSI_OFMT(I2S) | - SH_FSI_IFMT(I2S), -}; - static struct resource fsi_resources[] = { [0] = { .name = "FSI", @@ -141,9 +136,6 @@ static struct platform_device fsi_device = { .id = -1, .num_resources = ARRAY_SIZE(fsi_resources), .resource = fsi_resources, - .dev = { - .platform_data = &fsi_info, - }, }; static struct resource sh_mmcif_resources[] = { diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 920ed81f1c61..17f528a76a1c 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -673,14 +673,12 @@ static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) } static struct sh_fsi_platform_info fsi_info = { - .porta_flags = SH_FSI_BRS_INV | - SH_FSI_OFMT(PCM) | - SH_FSI_IFMT(PCM), + .porta_flags = SH_FSI_BRS_INV, .portb_flags = SH_FSI_BRS_INV | SH_FSI_BRM_INV | SH_FSI_LRS_INV | - SH_FSI_OFMT(SPDIF), + SH_FSI_FMT_SPDIF, .set_rate = fsi_set_rate, }; diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index aa4bcc347044..73b8c90b5072 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -614,14 +614,12 @@ fsi_set_rate_end: } static struct sh_fsi_platform_info fsi_info = { - .porta_flags = SH_FSI_BRS_INV | - SH_FSI_OFMT(PCM) | - SH_FSI_IFMT(PCM), + .porta_flags = SH_FSI_BRS_INV, .portb_flags = SH_FSI_BRS_INV | SH_FSI_BRM_INV | SH_FSI_LRS_INV | - SH_FSI_OFMT(SPDIF), + SH_FSI_FMT_SPDIF, .set_rate = fsi_set_rate, }; diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 037416f346cf..b96b79b970b2 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -723,9 +723,7 @@ static struct platform_device camera_devices[] = { /* FSI */ static struct sh_fsi_platform_info fsi_info = { - .portb_flags = SH_FSI_BRS_INV | - SH_FSI_OFMT(I2S) | - SH_FSI_IFMT(I2S), + .portb_flags = SH_FSI_BRS_INV, }; static struct resource fsi_resources[] = { diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index b4aef05dd8b5..c8bcf6a19b55 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -286,9 +286,7 @@ static struct platform_device ceu1_device = { /* FSI */ /* change J20, J21, J22 pin to 1-2 connection to use slave mode */ static struct sh_fsi_platform_info fsi_info = { - .porta_flags = SH_FSI_BRS_INV | - SH_FSI_OFMT(PCM) | - SH_FSI_IFMT(PCM), + .porta_flags = SH_FSI_BRS_INV, }; static struct resource fsi_resources[] = { diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 18e43279f70f..9a155f9d0a12 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -15,61 +15,29 @@ #define FSI_PORT_A 0 #define FSI_PORT_B 1 -/* flags format - - * 0xABC0EEFF - * - * A: channel size for TDM (input) - * B: channel size for TDM (ooutput) - * C: inversion - * E: input format - * F: output format - */ - #include #include -/* TDM channel */ -#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) -#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) - -#define SH_FSI_CH_IMASK 0xF0000000 -#define SH_FSI_CH_OMASK 0x0F000000 -#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) -#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) - -/* clock inversion */ -#define SH_FSI_INVERSION_MASK 0x00F00000 -#define SH_FSI_LRM_INV (1 << 20) -#define SH_FSI_BRM_INV (1 << 21) -#define SH_FSI_LRS_INV (1 << 22) -#define SH_FSI_BRS_INV (1 << 23) - -/* DI format */ -#define SH_FSI_FMT_MASK 0x000000FF -#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) -#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) -#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) -#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) - -#define SH_FSI_FMT_MONO 0 -#define SH_FSI_FMT_MONO_DELAY 1 -#define SH_FSI_FMT_PCM 2 -#define SH_FSI_FMT_I2S 3 -#define SH_FSI_FMT_TDM 4 -#define SH_FSI_FMT_TDM_DELAY 5 -#define SH_FSI_FMT_SPDIF 6 - - -#define SH_FSI_IFMT_TDM_CH(x) \ - (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) -#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ - (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) +/* + * flags format + * + * 0x000000BA + * + * A: inversion + * B: format mode + */ -#define SH_FSI_OFMT_TDM_CH(x) \ - (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) -#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ - (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) +/* A: clock inversion */ +#define SH_FSI_INVERSION_MASK 0x0000000F +#define SH_FSI_LRM_INV (1 << 0) +#define SH_FSI_BRM_INV (1 << 1) +#define SH_FSI_LRS_INV (1 << 2) +#define SH_FSI_BRS_INV (1 << 3) + +/* B: format mode */ +#define SH_FSI_FMT_MASK 0x000000F0 +#define SH_FSI_FMT_DAI (0 << 4) +#define SH_FSI_FMT_SPDIF (1 << 4) /* diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index ce058c749e6a..d6f4703b3c07 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -36,7 +36,8 @@ static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBS_CFS); return ret; } diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index 9b24ed466ab1..dbafd7ac5590 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -25,7 +25,8 @@ static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBS_CFS); return ret; } diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c53693d7266..0c9997e2d8c0 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -757,9 +757,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct fsi_master *master = fsi_get_master(fsi); u32 flags = fsi_get_info_flags(fsi); - u32 fmt; u32 data; int is_play = fsi_is_play(substream); @@ -779,54 +777,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, fsi_reg_write(fsi, CKG2, data); - /* do fmt, di fmt */ - data = 0; - fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); - switch (fmt) { - case SH_FSI_FMT_MONO: - data = CR_MONO; - fsi->chan_num = 1; - break; - case SH_FSI_FMT_MONO_DELAY: - data = CR_MONO_D; - fsi->chan_num = 1; - break; - case SH_FSI_FMT_PCM: - data = CR_PCM; - fsi->chan_num = 2; - break; - case SH_FSI_FMT_I2S: - data = CR_I2S; - fsi->chan_num = 2; - break; - case SH_FSI_FMT_TDM: - fsi->chan_num = is_play ? - SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_TDM | (fsi->chan_num - 1); - break; - case SH_FSI_FMT_TDM_DELAY: - fsi->chan_num = is_play ? - SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_TDM_D | (fsi->chan_num - 1); - break; - case SH_FSI_FMT_SPDIF: - if (master->core->ver < 2) { - dev_err(dai->dev, "This FSI can not use SPDIF\n"); - return -EINVAL; - } - data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; - fsi->chan_num = 2; - fsi_spdif_clk_ctrl(fsi, 1); - fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); - break; - default: - dev_err(dai->dev, "unknown format.\n"); - return -EINVAL; - } - is_play ? - fsi_reg_write(fsi, DO_FMT, data) : - fsi_reg_write(fsi, DI_FMT, data); - /* irq clear */ fsi_irq_disable(fsi, is_play); fsi_irq_clear_status(fsi); @@ -881,9 +831,52 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt) +{ + u32 data = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data = CR_I2S; + fsi->chan_num = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + data = CR_PCM; + fsi->chan_num = 2; + break; + default: + return -EINVAL; + } + + fsi_reg_write(fsi, DO_FMT, data); + fsi_reg_write(fsi, DI_FMT, data); + + return 0; +} + +static int fsi_set_fmt_spdif(struct fsi_priv *fsi) +{ + struct fsi_master *master = fsi_get_master(fsi); + u32 data = 0; + + if (master->core->ver < 2) + return -EINVAL; + + data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; + fsi->chan_num = 2; + fsi_spdif_clk_ctrl(fsi, 1); + fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); + + fsi_reg_write(fsi, DO_FMT, data); + fsi_reg_write(fsi, DI_FMT, data); + + return 0; +} + static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); + u32 flags = fsi_get_info_flags(fsi); u32 data = 0; int ret; @@ -901,7 +894,18 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) goto set_fmt_exit; } fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); - ret = 0; + + /* set format */ + switch (flags & SH_FSI_FMT_MASK) { + case SH_FSI_FMT_DAI: + ret = fsi_set_fmt_dai(fsi, fmt & SND_SOC_DAIFMT_FORMAT_MASK); + break; + case SH_FSI_FMT_SPDIF: + ret = fsi_set_fmt_spdif(fsi); + break; + default: + ret = -EINVAL; + } set_fmt_exit: pm_runtime_put_sync(dai->dev); -- cgit v1.2.3 From 70b2ac126a60c87145ae8a8eb1b4dccaa0bf5468 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 14:05:25 +0000 Subject: ASoC: Use card rather than soc-audio device to card PM functions The platform device for the card is tied closely to the soc-audio implementation which we're currently trying to remove in favour of allowing cards to have their own devices. Begin removing it by replacing it with the card in the suspend and resume callbacks we give to cards, also taking the opportunity to remove the legacy suspend types which are currently hard coded anyway. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 8 ++++---- sound/soc/pxa/raumfeld.c | 4 ++-- sound/soc/pxa/zylonite.c | 5 ++--- sound/soc/soc-core.c | 9 ++++----- 4 files changed, 12 insertions(+), 14 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1355ef029d82..4a489ae44a6e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -654,10 +654,10 @@ struct snd_soc_card { /* the pre and post PM functions are used to do any PM work before and * after the codec and DAI's do any PM work. */ - int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); - int (*suspend_post)(struct platform_device *pdev, pm_message_t state); - int (*resume_pre)(struct platform_device *pdev); - int (*resume_post)(struct platform_device *pdev); + int (*suspend_pre)(struct snd_soc_card *card); + int (*suspend_post)(struct snd_soc_card *card); + int (*resume_pre)(struct snd_soc_card *card); + int (*resume_post)(struct snd_soc_card *card); /* callbacks */ int (*set_bias_level)(struct snd_soc_card *, diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 0fd60f423036..db1dd560a585 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +static int raumfeld_line_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct platform_device *pdev) +static int raumfeld_line_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index b222a7d72027..7b729013a728 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -226,8 +226,7 @@ static int zylonite_remove(struct platform_device *pdev) return 0; } -static int zylonite_suspend_post(struct platform_device *pdev, - pm_message_t state) +static int zylonite_suspend_post(struct snd_soc_card *card) { if (clk_pout) clk_disable(pout); @@ -235,7 +234,7 @@ static int zylonite_suspend_post(struct platform_device *pdev, return 0; } -static int zylonite_resume_pre(struct platform_device *pdev) +static int zylonite_resume_pre(struct snd_soc_card *card) { int ret = 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 14861f95f629..446838e7d3ec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1011,7 +1011,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_pre) - card->suspend_pre(pdev, PMSG_SUSPEND); + card->suspend_pre(card); for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; @@ -1078,7 +1078,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_post) - card->suspend_post(pdev, PMSG_SUSPEND); + card->suspend_post(card); return 0; } @@ -1090,7 +1090,6 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); - struct platform_device *pdev = to_platform_device(card->dev); struct snd_soc_codec *codec; int i; @@ -1104,7 +1103,7 @@ static void soc_resume_deferred(struct work_struct *work) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2); if (card->resume_pre) - card->resume_pre(pdev); + card->resume_pre(card); /* resume AC97 DAIs */ for (i = 0; i < card->num_rtd; i++) { @@ -1179,7 +1178,7 @@ static void soc_resume_deferred(struct work_struct *work) } if (card->resume_post) - card->resume_post(pdev); + card->resume_post(card); dev_dbg(card->dev, "resume work completed\n"); -- cgit v1.2.3 From e7361ec4996c170c63c4ac379085896db85ff34d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 14:17:20 +0000 Subject: ASoC: Replace pdev with card in machine driver probe and remove In order to support cards instantiated without using soc-audio remove the use of the platform device in the card probe() and remove() ops. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 4 ++-- sound/soc/fsl/mpc8610_hpcd.c | 6 ++---- sound/soc/fsl/p1022_ds.c | 6 ++---- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/pxa/zylonite.c | 4 ++-- sound/soc/soc-core.c | 8 +++----- 6 files changed, 13 insertions(+), 19 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4a489ae44a6e..2d10090a08c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -649,8 +649,8 @@ struct snd_soc_card { bool instantiated; - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); + int (*probe)(struct snd_soc_card *card); + int (*remove)(struct snd_soc_card *card); /* the pre and post PM functions are used to do any PM work before and * after the codec and DAI's do any PM work. */ diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 7d7847a1e66b..c16c6b2eff95 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -53,9 +53,8 @@ struct mpc8610_hpcd_data { * * Here we program the DMACR and PMUXCR registers. */ -static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); struct ccsr_guts_86xx __iomem *guts; @@ -138,9 +137,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * This function is called to remove the sound device for one SSI. We * de-program the DMACR and PMUXCR register. */ -static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); struct ccsr_guts_86xx __iomem *guts; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 026b756961e0..66e0b68af147 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -85,9 +85,8 @@ struct machine_data { * * Here we program the DMACR and PMUXCR registers. */ -static int p1022_ds_machine_probe(struct platform_device *sound_device) +static int p1022_ds_machine_probe(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct machine_data *mdata = container_of(card, struct machine_data, card); struct ccsr_guts_85xx __iomem *guts; @@ -160,9 +159,8 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream) * This function is called to remove the sound device for one SSI. We * de-program the DMACR and PMUXCR register. */ -static int p1022_ds_machine_remove(struct platform_device *sound_device) +static int p1022_ds_machine_remove(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct machine_data *mdata = container_of(card, struct machine_data, card); struct ccsr_guts_85xx __iomem *guts; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index f75804ef0897..489139a31cf9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -237,7 +237,7 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; -static int tosa_probe(struct platform_device *dev) +static int tosa_probe(struct snd_soc_card *card) { int ret; @@ -251,7 +251,7 @@ static int tosa_probe(struct platform_device *dev) return ret; } -static int tosa_remove(struct platform_device *dev) +static int tosa_remove(struct snd_soc_card *card) { gpio_free(TOSA_GPIO_L_MUTE); return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 7b729013a728..c5858296b48a 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -189,7 +189,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { }, }; -static int zylonite_probe(struct platform_device *pdev) +static int zylonite_probe(struct snd_soc_card *card) { int ret; @@ -216,7 +216,7 @@ static int zylonite_probe(struct platform_device *pdev) return 0; } -static int zylonite_remove(struct platform_device *pdev) +static int zylonite_remove(struct snd_soc_card *card) { if (clk_pout) { clk_disable(pout); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 446838e7d3ec..4bc2365bf1dd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1717,7 +1717,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, static void snd_soc_instantiate_card(struct snd_soc_card *card) { - struct platform_device *pdev = to_platform_device(card->dev); struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; @@ -1781,7 +1780,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) /* initialise the sound card only once */ if (card->probe) { - ret = card->probe(pdev); + ret = card->probe(card); if (ret < 0) goto card_probe_error; } @@ -1842,7 +1841,7 @@ probe_dai_err: card_probe_error: if (card->remove) - card->remove(pdev); + card->remove(card); snd_card_free(card->snd_card); @@ -1888,7 +1887,6 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct platform_device *pdev = to_platform_device(card->dev); int i; /* make sure any delayed work runs */ @@ -1909,7 +1907,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove the card */ if (card->remove) - card->remove(pdev); + card->remove(card); kfree(card->rtd); snd_card_free(card->snd_card); -- cgit v1.2.3 From 6f8ab4ac292f81b9246ddf363bf1c6a2fc7a0629 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 14:59:27 +0000 Subject: ASoC: Export card PM callbacks for use in direct registered cards Allow hookup of cards registered directly with the core to the PM operations by exporting the device power management operations to modules, also exporting the default PM operations since it is expected that most cards will end up using exactly the same setup. Note that the callbacks require that the driver data for the card be the snd_soc_card. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 34 +++++++++++++++++----------------- 2 files changed, 22 insertions(+), 17 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 2d10090a08c0..7e8cf4f318a9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -260,6 +260,9 @@ enum snd_soc_compress_type { int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int snd_soc_suspend(struct device *dev); +int snd_soc_resume(struct device *dev); +int snd_soc_poweroff(struct device *dev); int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); @@ -802,4 +805,6 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) extern struct dentry *snd_soc_debugfs_root; #endif +extern const struct dev_pm_ops snd_soc_pm_ops; + #endif diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4bc2365bf1dd..5dffc7a469c0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -965,12 +965,11 @@ static struct snd_pcm_ops soc_pcm_ops = { .pointer = soc_pcm_pointer, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ -static int soc_suspend(struct device *dev) +int snd_soc_suspend(struct device *dev) { - struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; int i; @@ -1082,6 +1081,7 @@ static int soc_suspend(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_suspend); /* deferred resume work, so resume can complete before we finished * setting our codec back up, which can be very slow on I2C @@ -1187,10 +1187,9 @@ static void soc_resume_deferred(struct work_struct *work) } /* powers up audio subsystem after a suspend */ -static int soc_resume(struct device *dev) +int snd_soc_resume(struct device *dev) { - struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_card *card = dev_get_drvdata(dev); int i; /* AC97 devices might have other drivers hanging off them so @@ -1212,9 +1211,10 @@ static int soc_resume(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_resume); #else -#define soc_suspend NULL -#define soc_resume NULL +#define snd_soc_suspend NULL +#define snd_soc_resume NULL #endif static struct snd_soc_dai_ops null_dai_ops = { @@ -1924,10 +1924,9 @@ static int soc_remove(struct platform_device *pdev) return 0; } -static int soc_poweroff(struct device *dev) +int snd_soc_poweroff(struct device *dev) { - struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_card *card = dev_get_drvdata(dev); int i; if (!card->instantiated) @@ -1944,11 +1943,12 @@ static int soc_poweroff(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_poweroff); -static const struct dev_pm_ops soc_pm_ops = { - .suspend = soc_suspend, - .resume = soc_resume, - .poweroff = soc_poweroff, +const struct dev_pm_ops snd_soc_pm_ops = { + .suspend = snd_soc_suspend, + .resume = snd_soc_resume, + .poweroff = snd_soc_poweroff, }; /* ASoC platform driver */ @@ -1956,7 +1956,7 @@ static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", .owner = THIS_MODULE, - .pm = &soc_pm_ops, + .pm = &snd_soc_pm_ops, }, .probe = soc_probe, .remove = soc_remove, -- cgit v1.2.3 From aaee8ef146111566e1c607bdf368d73fb966be2e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 20:53:50 +0000 Subject: ASoC: Make cache status available via debugfs Could just as well live in sysfs but sysfs doesn't have the simple value export helpers debugfs does. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 5 +++++ 2 files changed, 7 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7e8cf4f318a9..64856d656f15 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -493,14 +493,14 @@ struct snd_soc_codec { struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; unsigned int cache_bypass:1; /* Suppress access to the cache */ - unsigned int cache_only:1; /* Suppress writes to hardware */ - unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ 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 sysfs_registered:1; /* codec has been sysfs registered */ 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 */ void *control_data; /* codec control (i2c/3wire) data */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5dffc7a469c0..8af47f7a8fcb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -235,6 +235,11 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) return; } + debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + &codec->cache_sync); + debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, + &codec->cache_only); + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->debugfs_codec_root, codec, &codec_reg_fops); -- cgit v1.2.3 From f85a9e0d260905f98d4ca6b66f0e64f63a729dba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 21:41:28 +0000 Subject: ASoC: Add subsequence information to seq_notify callbacks Allows drivers to distinguish which subsequence is being notified when they get called back. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- sound/soc/soc-dapm.c | 5 +++-- 3 files changed, 5 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6a25e6993859..979ed84e07d6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -501,7 +501,7 @@ struct snd_soc_dapm_context { struct snd_soc_dapm_update *update; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type); + enum snd_soc_dapm_type, int); struct device *dev; /* from parent - for debug */ struct snd_soc_codec *codec; /* parent codec */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 64856d656f15..7ecdaefd1b63 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -553,7 +553,7 @@ struct snd_soc_codec_driver { enum snd_soc_bias_level level); void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type); + enum snd_soc_dapm_type, int); }; /* SoC platform interface */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 37b376f4c75d..0f94fd057f29 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -899,7 +899,8 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i); + i, + cur_subseq); } INIT_LIST_HEAD(&pending); @@ -968,7 +969,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i); + i, cur_subseq); } } -- cgit v1.2.3 From dddf3e4c257879bc35cda3f542507c43f2648a2a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 28 Jan 2011 13:11:47 +0000 Subject: ASoC: Add card driver data Provide driver data for cards within the card structure. To simplify the implementation of the PM operations we don't use the struct device driver data as this is used by the core to retrieve the card in callbacks from the device model and PM core. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ecdaefd1b63..4b6c0a8c332f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -705,6 +705,8 @@ struct snd_soc_card { struct dentry *debugfs_pop_time; #endif u32 pop_time; + + void *drvdata; }; /* SoC machine DAI configuration, glues a codec and cpu DAI together */ @@ -756,6 +758,17 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec, /* device driver data */ +static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, + void *data) +{ + card->drvdata = data; +} + +static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card) +{ + return card->drvdata; +} + static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, void *data) { -- cgit v1.2.3 From a98a0bc6c92eacd181417a9c0ccd2e8028066622 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Thu, 3 Feb 2011 03:11:45 +0300 Subject: ASoC: CS4271: Move Chip Select control out of the CODEC code. Move Chip Select control out of the CODEC code for CS4271. Signed-off-by: Alexander Sverdlin Reviewed-by: H Hartley Sweeten Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/cs4271.h | 1 - sound/soc/codecs/cs4271.c | 22 +++------------------- 2 files changed, 3 insertions(+), 20 deletions(-) (limited to 'include') diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 16f8d325d3dc..50a059e7d116 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,6 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int gpio_disable; /* GPIO that disable serial bus, if any */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9c5b7db0ce6a..1791796216c8 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -441,22 +441,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int gpio_disable = -EINVAL; codec->control_data = cs4271->control_data; - if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - if (gpio_is_valid(cs4271plat->gpio_disable)) - gpio_disable = cs4271plat->gpio_disable; - } - - if (gpio_disable >= 0) - if (gpio_request(gpio_disable, "CS4271 Disable")) - gpio_disable = -EINVAL; - if (gpio_disable >= 0) - gpio_direction_output(gpio_disable, 0); + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -471,7 +460,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) } cs4271->gpio_nreset = gpio_nreset; - cs4271->gpio_disable = gpio_disable; /* * In case of I2C, chip address specified in board data. @@ -509,10 +497,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset, gpio_disable; + int gpio_nreset; gpio_nreset = cs4271->gpio_nreset; - gpio_disable = cs4271->gpio_disable; if (gpio_is_valid(gpio_nreset)) { /* Set codec to the reset state */ @@ -520,9 +507,6 @@ static int cs4271_remove(struct snd_soc_codec *codec) gpio_free(gpio_nreset); } - if (gpio_is_valid(gpio_disable)) - gpio_free(gpio_disable); - return 0; }; -- cgit v1.2.3 From fa9879edebdaad4cfcd2dbe3eaa2ba0dc4f0a262 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 14:44:17 +0530 Subject: ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Signed-off-by: Mark Brown --- include/sound/soc.h | 23 +++++++++++++++++++++++ sound/soc/soc-jack.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4b6c0a8c332f..4ccf1e4e0dd0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -234,6 +234,7 @@ struct snd_soc_codec; struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_jack; +struct snd_soc_jack_zone; struct snd_soc_jack_pin; struct snd_soc_cache_ops; #include @@ -307,6 +308,9 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones); +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage); #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); @@ -406,6 +410,24 @@ struct snd_soc_jack_pin { bool invert; }; +/** + * struct snd_soc_jack_zone - Describes voltage zones of jack detection + * + * @min_mv: start voltage in mv + * @max_mv: end voltage in mv + * @jack_type: type of jack that is expected for this voltage + * @debounce_time: debounce_time for jack, codec driver should wait for this + * duration before reading the adc for voltages + * @:list: list container + */ +struct snd_soc_jack_zone { + unsigned int min_mv; + unsigned int max_mv; + unsigned int jack_type; + unsigned int debounce_time; + struct list_head list; +}; + /** * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * @@ -435,6 +457,7 @@ struct snd_soc_jack { struct list_head pins; int status; struct blocking_notifier_head notifier; + struct list_head jack_zones; }; /* SoC PCM stream information */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc7375a..99dbaf756b44 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, { jack->codec = codec; INIT_LIST_HEAD(&jack->pins); + INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); @@ -111,6 +112,51 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +/** + * snd_soc_jack_add_zones - Associate voltage zones with jack + * + * @jack: ASoC jack + * @count: Number of zones + * @zone: Array of zones + * + * After this function has been called the zones specified in the + * array will be associated with the jack. + */ +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones) +{ + int i; + + for (i = 0; i < count; i++) { + INIT_LIST_HEAD(&zones[i].list); + list_add(&(zones[i].list), &jack->jack_zones); + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); + +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack from the zones delcared in the jack type + * + * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in + * + * Based on the mic bias value passed, this function helps identify + * the type of jack from the already delcared jack zones + */ +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) +{ + struct snd_soc_jack_zone *zone; + + list_for_each_entry(zone, &jack->jack_zones, list) { + if (micbias_voltage >= zone->min_mv && + micbias_voltage < zone->max_mv) + return zone->jack_type; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * -- cgit v1.2.3 From 7887ab3a274dc5f1d1d94ca0cd41ae495d01f94f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 16:35:55 -0800 Subject: ASoC: Allow GPIO jack detection to be configured as a wake source Some systems wish to use jacks as wake sources. Provide a wake flag in the GPIO configuration which causes the driver to enable the IRQ as a wake source. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 3 +++ sound/soc/soc-jack.c | 8 ++++++++ 2 files changed, 11 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4ccf1e4e0dd0..fb57c33482e5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -436,6 +436,7 @@ struct snd_soc_jack_zone { * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms + * @wake: enable as wake source */ #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio { @@ -444,6 +445,8 @@ struct snd_soc_jack_gpio { int report; int invert; int debounce_time; + bool wake; + struct snd_soc_jack *jack; struct delayed_work work; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4579ee090bbf..1382251ed2a2 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -330,6 +330,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto err; + if (gpios[i].wake) { + ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + if (ret != 0) + printk(KERN_ERR + "Failed to mark GPIO %d as wake source: %d\n", + gpios[i].gpio, ret); + } + #ifdef CONFIG_GPIO_SYSFS /* Expose GPIO value over sysfs for diagnostic purposes */ gpio_export(gpios[i].gpio, false); -- cgit v1.2.3 From fadddc8753ccfab26ee57f3205d6926fe4be1350 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 16:41:42 -0800 Subject: ASoC: Add kerneldoc for jack_status_check callback Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index fb57c33482e5..65d865f7e8c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -437,6 +437,9 @@ struct snd_soc_jack_zone { * @invert: report presence in low state * @debouce_time: debouce time in ms * @wake: enable as wake source + * @jack_status_check: callback function which overrides the detection + * to provide more complex checks (eg, reading an + * ADC). */ #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio { -- cgit v1.2.3 From 9b7c525dfaa9a1b5f01db1f3a1edc50bbb6eb739 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 20:05:44 -0800 Subject: ASoC: Support WM8958 direct microphone detection IRQ Allow direct routing of the WM8958 microphone detection signal to a GPIO to be used, saving the need to demux the interrupt. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/linux/mfd/wm8994/pdata.h | 5 ++++ sound/soc/codecs/wm8994.c | 57 ++++++++++++++++++++++++---------------- 2 files changed, 40 insertions(+), 22 deletions(-) (limited to 'include') diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 9eab263658be..06869466b7f0 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -103,6 +103,11 @@ struct wm8994_pdata { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* IRQ for microphone detection if brought out directly as a + * signal. + */ + int micdet_irq; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b23e91027d64..1ad6e3db7804 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -104,6 +104,7 @@ struct wm8994_priv { void *jack_cb_data; bool jack_is_mic; bool jack_is_video; + int micdet_irq; int revision; struct wm8994_pdata *pdata; @@ -3102,6 +3103,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + if (wm8994->pdata && wm8994->pdata->micdet_irq) + wm8994->micdet_irq = wm8994->pdata->micdet_irq; + else if (wm8994->pdata && wm8994->pdata->irq_base) + wm8994->micdet_irq = wm8994->pdata->irq_base + + WM8994_IRQ_MIC1_DET; + pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); @@ -3150,14 +3157,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8994_mic_irq, "Mic 1 detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic1 detect IRQ: %d\n", - ret); + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8994_mic_irq, + IRQF_TRIGGER_RISING, + "Mic1 detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); + } ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3188,15 +3198,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8958_mic_irq, "Mic detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic detect IRQ: %d\n", - ret); - break; + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8958_mic_irq, + IRQF_TRIGGER_RISING, + "Mic detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic detect IRQ: %d\n", + ret); + } } /* Remember if AIFnLRCLK is configured as a GPIO. This should be @@ -3328,7 +3340,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); err: kfree(wm8994); return ret; @@ -3345,8 +3358,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3356,8 +3369,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) break; case WM8958: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); break; } kfree(wm8994->retune_mobile_texts); -- cgit v1.2.3 From 48e028eccabc9c246bfad175262582a1ce34a316 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 17:11:59 -0800 Subject: ASoC: Support configuration of WM8958 microphone bias analogue parameters The WM8958 has a different microphone bias architecture to WM8994 so needs different configuration to WM8994. Support this in platform data. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/linux/mfd/wm8994/pdata.h | 7 +++++-- include/linux/mfd/wm8994/registers.h | 2 ++ sound/soc/codecs/wm8994.c | 7 +++++++ 3 files changed, 14 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 06869466b7f0..466b1c777aff 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -108,13 +108,16 @@ struct wm8994_pdata { */ int micdet_irq; - /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ + /* WM8994 microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; - /* Jack detect threashold levels, see datasheet for values */ + /* WM8994 jack detect threashold levels, see datasheet for values */ unsigned int jd_scthr:2; unsigned int jd_thr:2; + + /* WM8958 microphone bias configuration */ + int micbias[2]; }; #endif diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index be072faec6f0..f3ee84284670 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -63,6 +63,8 @@ #define WM8994_MICBIAS 0x3A #define WM8994_LDO_1 0x3B #define WM8994_LDO_2 0x3C +#define WM8958_MICBIAS1 0x3D +#define WM8958_MICBIAS2 0x3E #define WM8994_CHARGE_PUMP_1 0x4C #define WM8958_CHARGE_PUMP_2 0x4D #define WM8994_CLASS_W_1 0x51 diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1ad6e3db7804..9b9c15ffb7d2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2855,6 +2855,13 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) else snd_soc_add_controls(wm8994->codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); + + for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { + if (pdata->micbias[i]) { + snd_soc_write(codec, WM8958_MICBIAS1 + i, + pdata->micbias[i] & 0xffff); + } + } } /** -- cgit v1.2.3 From 4a5f7bda8fe9d0ed08ed4c5beb5dc3fa62f09d05 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Mar 2011 20:10:46 +0000 Subject: ASoC: Add platform data for WM9081 IRQ pin configuration The WM9081 IRQ output can be either active high or active low and can support either CMOS or open drain modes. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/wm9081.h | 9 ++++++--- sound/soc/codecs/wm9081.c | 29 +++++++++++++++++++---------- 2 files changed, 25 insertions(+), 13 deletions(-) (limited to 'include') diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h index e173ddbf6bd4..f34b0b1716d8 100644 --- a/include/sound/wm9081.h +++ b/include/sound/wm9081.h @@ -17,9 +17,12 @@ struct wm9081_retune_mobile_setting { u16 config[20]; }; -struct wm9081_retune_mobile_config { - struct wm9081_retune_mobile_setting *configs; - int num_configs; +struct wm9081_pdata { + bool irq_high; /* IRQ is active high */ + bool irq_cmos; /* IRQ is in CMOS mode */ + + struct wm9081_retune_mobile_setting *retune_configs; + int num_retune_configs; }; #endif diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 2103623a0776..7883f3ed797b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -167,7 +167,7 @@ struct wm9081_priv { int fll_fref; int fll_fout; int tdm_width; - struct wm9081_retune_mobile_config *retune; + struct wm9081_pdata pdata; }; static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1082,21 +1082,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, aif4 |= wm9081->bclk / wm9081->fs; /* Apply a ReTune Mobile configuration if it's in use */ - if (wm9081->retune) { - struct wm9081_retune_mobile_config *retune = wm9081->retune; + if (wm9081->pdata.num_retune_configs) { + struct wm9081_pdata *pdata = &wm9081->pdata; struct wm9081_retune_mobile_setting *s; int eq1; best = 0; - best_val = abs(retune->configs[0].rate - wm9081->fs); - for (i = 0; i < retune->num_configs; i++) { - cur_val = abs(retune->configs[i].rate - wm9081->fs); + best_val = abs(pdata->retune_configs[0].rate - wm9081->fs); + for (i = 0; i < pdata->num_retune_configs; i++) { + cur_val = abs(pdata->retune_configs[i].rate - + wm9081->fs); if (cur_val < best_val) { best_val = cur_val; best = i; } } - s = &retune->configs[best]; + s = &pdata->retune_configs[best]; dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", s->name, s->rate); @@ -1255,6 +1256,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ @@ -1266,7 +1275,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); - if (!wm9081->retune) { + if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); snd_soc_add_controls(codec, wm9081_eq_controls, @@ -1343,8 +1352,8 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, wm9081->control_data = i2c; if (dev_get_platdata(&i2c->dev)) - memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev), - sizeof(wm9081->retune)); + memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), + sizeof(wm9081->pdata)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); -- cgit v1.2.3 From e37a4970cd7ab6aec9e848cd3c355fd47fd18afd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:21:57 +0000 Subject: ASoC: Add a per-card DAPM context This means that rather than adding the board specific DAPM widgets to a random CODEC DAPM context they can be added to the card itself which is a bit cleaner. Previously there only was one DAPM context and it was tied to the single supported CODEC. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 16 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 65d865f7e8c0..8064cd130356 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -729,6 +729,9 @@ struct snd_soc_card { struct list_head paths; struct list_head dapm_list; + /* Generic DAPM context for the card */ + struct snd_soc_dapm_context dapm; + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; struct dentry *debugfs_pop_time; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64befac3f9c3..24bfc3ff8e17 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1837,6 +1837,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } card->snd_card->dev = card->dev; + card->dapm.bias_level = SND_SOC_BIAS_OFF; + card->dapm.dev = card->dev; + card->dapm.card = card; + list_add(&card->dapm.list, &card->dapm_list); + #ifdef CONFIG_PM_SLEEP /* deferred resume work */ INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); @@ -1867,6 +1872,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + card->dapm.debugfs_dapm = debugfs_create_dir("dapm", + card->debugfs_card_root); + if (!card->dapm.debugfs_dapm) + printk(KERN_WARNING + "Failed to create card DAPM debugfs directory\n"); + + snd_soc_dapm_debugfs_init(&card->dapm); + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- cgit v1.2.3 From b8ad29debd7401d257da923480d32838172c431a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:35:51 +0000 Subject: ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 7 +++++++ 2 files changed, 15 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 8064cd130356..11d59bd13886 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,14 @@ struct snd_soc_card { struct snd_soc_pcm_runtime *rtd_aux; int num_aux_rtd; + /* + * Card-specific routes and widgets. + */ + struct snd_soc_dapm_widget *dapm_widgets; + int num_dapm_widgets; + struct snd_soc_dapm_route *dapm_routes; + int num_dapm_routes; + struct work_struct deferred_resume_work; /* lists of probed devices belonging to this card */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 24bfc3ff8e17..6a2839c18447 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1872,6 +1872,13 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + if (card->dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, + card->num_dapm_widgets); + if (card->dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, + card->num_dapm_routes); + card->dapm.debugfs_dapm = debugfs_create_dir("dapm", card->debugfs_card_root); if (!card->dapm.debugfs_dapm) -- cgit v1.2.3 From 28e9ad921d3b7defd8940a3e30e8241c8ed734db Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:36:34 +0000 Subject: ASoC: Add a late_probe() callback to cards This is run after the DAPM widgets and routes are added, allowing setup of things like jacks using the routes. The main card probe() is run before anything else so can't be used for this purpose. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 9 +++++++++ 2 files changed, 10 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 11d59bd13886..9c2a6dd170f1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -682,6 +682,7 @@ struct snd_soc_card { bool instantiated; int (*probe)(struct snd_soc_card *card); + int (*late_probe)(struct snd_soc_card *card); int (*remove)(struct snd_soc_card *card); /* the pre and post PM functions are used to do any PM work before and diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6a2839c18447..8926d38fc5a3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1892,6 +1892,15 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->name); + if (card->late_probe) { + ret = card->late_probe(card); + if (ret < 0) { + dev_err(card->dev, "%s late_probe() failed: %d\n", + card->name, ret); + goto probe_aux_dev_err; + } + } + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); -- cgit v1.2.3 From 1d471cd1261a44a3b28350bef7e5113a4609c106 Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Wed, 2 Mar 2011 14:52:32 +0100 Subject: ASoC: Add TI tlv320aic32x4 codec support. This patch adds support for tlv320aic3205 and tlv320aic3254 codecs. It doesn't include miniDSP support for aic3254. Signed-off-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/tlv320aic32x4.h | 31 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4.c | 794 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 143 +++++++ 5 files changed, 974 insertions(+) create mode 100644 include/sound/tlv320aic32x4.h create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h (limited to 'include') diff --git a/include/sound/tlv320aic32x4.h b/include/sound/tlv320aic32x4.h new file mode 100644 index 000000000000..c009f70b4029 --- /dev/null +++ b/include/sound/tlv320aic32x4.h @@ -0,0 +1,31 @@ +/* + * tlv320aic32x4.h -- TLV320AIC32X4 Soc Audio driver platform data + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC32X4_PDATA_H +#define _AIC32X4_PDATA_H + +#define AIC32X4_PWR_MICBIAS_2075_LDOIN 0x00000001 +#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE 0x00000002 +#define AIC32X4_PWR_AIC32X4_LDO_ENABLE 0x00000004 +#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36 0x00000008 +#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED 0x00000010 + +#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K 0x00000001 +#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K 0x00000002 + +struct aic32x4_pdata { + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c04da1871297..82a46309ded6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TVL320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -206,6 +207,9 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI +config SND_SOC_TVL320AIC32X4 + tristate + config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3bbb08c512d0..b43f9d418c9b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -112,6 +113,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c new file mode 100644 index 000000000000..ee82e3896039 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -0,0 +1,794 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic32x4.h" + +struct aic32x4_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; + u8 blck_N; +}; + +struct aic32x4_priv { + u32 sysclk; + s32 master; + u8 page_no; + void *control_data; + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +/* 0dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0); +/* 0dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0); + +static const struct snd_kcontrol_new aic32x4_snd_controls[] = { + SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 7, 0x01, 1), + + SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0), + SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + + SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0), + SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0), + SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1, + 4, 0x07, 0), + SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1, + 0, 0x03, 0), + SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2, + 6, 0x03, 0), + SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2, + 1, 0x1F, 0), + SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3, + 0, 0x7F, 0), + SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6, + 0, 0x1F, 0), + SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7, + 0, 0x0F, 0), +}; + +static const struct aic32x4_rate_divs aic32x4_divs[] = { + /* 8k rate */ + {AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24}, + {AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24}, + {AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24}, + /* 11.025k rate */ + {AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16}, + {AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16}, + /* 16k rate */ + {AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12}, + {AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12}, + {AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12}, + /* 22.05k rate */ + {AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8}, + {AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8}, + {AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8}, + /* 32k rate */ + {AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6}, + {AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6}, + /* 44.1k rate */ + {AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4}, + /* 48k rate */ + {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} +}; + +static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new hpr_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new lol_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new lor_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), + SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0, + &hpl_output_mixer_controls[0], + ARRAY_SIZE(hpl_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0, + &lol_output_mixer_controls[0], + ARRAY_SIZE(lol_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0), + SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0, + &hpr_output_mixer_controls[0], + ARRAY_SIZE(hpr_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0, + &lor_output_mixer_controls[0], + ARRAY_SIZE(lor_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOL"), + SND_SOC_DAPM_OUTPUT("LOR"), + SND_SOC_DAPM_INPUT("IN1_L"), + SND_SOC_DAPM_INPUT("IN1_R"), + SND_SOC_DAPM_INPUT("IN2_L"), + SND_SOC_DAPM_INPUT("IN2_R"), + SND_SOC_DAPM_INPUT("IN3_L"), + SND_SOC_DAPM_INPUT("IN3_R"), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + /* Left Output */ + {"HPL Output Mixer", "L_DAC Switch", "Left DAC"}, + {"HPL Output Mixer", "IN1_L Switch", "IN1_L"}, + + {"HPL Power", NULL, "HPL Output Mixer"}, + {"HPL", NULL, "HPL Power"}, + + {"LOL Output Mixer", "L_DAC Switch", "Left DAC"}, + + {"LOL Power", NULL, "LOL Output Mixer"}, + {"LOL", NULL, "LOL Power"}, + + /* Right Output */ + {"HPR Output Mixer", "R_DAC Switch", "Right DAC"}, + {"HPR Output Mixer", "IN1_R Switch", "IN1_R"}, + + {"HPR Power", NULL, "HPR Output Mixer"}, + {"HPR", NULL, "HPR Power"}, + + {"LOR Output Mixer", "R_DAC Switch", "Right DAC"}, + + {"LOR Power", NULL, "LOR Output Mixer"}, + {"LOR", NULL, "LOR Power"}, + + /* Left input */ + {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, + {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, + {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, + + {"Left ADC", NULL, "Left Input Mixer"}, + + /* Right Input */ + {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, + {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, + {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, + + {"Right ADC", NULL, "Right Input Mixer"}, +}; + +static inline int aic32x4_change_page(struct snd_soc_codec *codec, + unsigned int new_page) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data[2]; + int ret; + + data[0] = 0x00; + data[1] = new_page & 0xff; + + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) { + aic32x4->page_no = new_page; + return 0; + } else { + return ret; + } +} + +static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + u8 data[2]; + int ret; + + /* A write to AIC32X4_PSEL is really a non-explicit page change */ + if (reg == AIC32X4_PSEL) + return aic32x4_change_page(codec, val); + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + + data[0] = fixed_reg & 0xff; + data[1] = val & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + int ret; + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); +} + +static inline int aic32x4_get_divs(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { + if ((aic32x4_divs[i].rate == rate) + && (aic32x4_divs[i].mclk == mclk)) { + return i; + } + } + printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + return -EINVAL; +} + +static int aic32x4_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + ARRAY_SIZE(aic32x4_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case AIC32X4_FREQ_12000000: + case AIC32X4_FREQ_24000000: + case AIC32X4_FREQ_25000000: + aic32x4->sysclk = freq; + return 0; + } + printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + return -EINVAL; +} + +static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 iface_reg_1; + u8 iface_reg_2; + u8 iface_reg_3; + + iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); + iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); + iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); + iface_reg_2 = 0; + iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); + iface_reg_3 = iface_reg_3 & ~(1 << 3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic32x4->master = 1; + iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic32x4->master = 0; + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + iface_reg_2 = 0x01; /* add offset 1 */ + break; + case SND_SOC_DAIFMT_DSP_B: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg_1 |= + (AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg_1 |= + (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1); + snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2); + snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data; + int i; + + i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + if (i < 0) { + printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + return i; + } + + /* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */ + snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN); + snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK); + + /* We will fix R value to 1 and will make P & J=K.D as varialble */ + data = snd_soc_read(codec, AIC32X4_PLLPR); + data &= ~(7 << 4); + snd_soc_write(codec, AIC32X4_PLLPR, + (data | (aic32x4_divs[i].p_val << 4) | 0x01)); + + snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j); + + snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8)); + snd_soc_write(codec, AIC32X4_PLLDLSB, + (aic32x4_divs[i].pll_d & 0xff)); + + /* NDAC divider value */ + data = snd_soc_read(codec, AIC32X4_NDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac); + + /* MDAC divider value */ + data = snd_soc_read(codec, AIC32X4_MDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac); + + /* DOSR MSB & LSB values */ + snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); + snd_soc_write(codec, AIC32X4_DOSRLSB, + (aic32x4_divs[i].dosr & 0xff)); + + /* NADC divider value */ + data = snd_soc_read(codec, AIC32X4_NADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc); + + /* MADC divider value */ + data = snd_soc_read(codec, AIC32X4_MADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc); + + /* AOSR value */ + snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr); + + /* BCLK N divider */ + data = snd_soc_read(codec, AIC32X4_BCLKN); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N); + + data = snd_soc_read(codec, AIC32X4_IFACE1); + data = data & ~(3 << 4); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); + break; + } + snd_soc_write(codec, AIC32X4_IFACE1, data); + + return 0; +} + +static int aic32x4_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dac_reg; + + dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON; + if (mute) + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON); + else + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg); + return 0; +} + +static int aic32x4_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 value; + + switch (level) { + case SND_SOC_BIAS_ON: + if (aic32x4->master) { + /* Switch on PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value | AIC32X4_PLLEN)); + + /* Switch on NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value | AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value | AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value | AIC32X4_MDACEN); + + /* Switch on MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value | AIC32X4_MDACEN); + + /* Switch on BCLK_N Divider */ + value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_write(codec, AIC32X4_BCLKN, + value | AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (aic32x4->master) { + /* Switch off PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value & ~AIC32X4_PLLEN)); + + /* Switch off NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value & ~AIC32X4_NDACEN); + + /* Switch off MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value & ~AIC32X4_MDACEN); + + /* Switch off NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value & ~AIC32X4_NDACEN); + + /* Switch off MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value & ~AIC32X4_MDACEN); + value = snd_soc_read(codec, AIC32X4_BCLKN); + + /* Switch off BCLK_N Divider */ + snd_soc_write(codec, AIC32X4_BCLKN, + value & ~AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops aic32x4_ops = { + .hw_params = aic32x4_hw_params, + .digital_mute = aic32x4_mute, + .set_fmt = aic32x4_set_dai_fmt, + .set_sysclk = aic32x4_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver aic32x4_dai = { + .name = "tlv320aic32x4-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .ops = &aic32x4_ops, + .symmetric_rates = 1, +}; + +static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic32x4_resume(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic32x4_probe(struct snd_soc_codec *codec) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u32 tmp_reg; + + codec->hw_write = (hw_write_t) i2c_master_send; + codec->control_data = aic32x4->control_data; + + snd_soc_write(codec, AIC32X4_RESET, 0x01); + + /* Power platform configuration */ + if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { + snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) { + snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN); + } + tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + tmp_reg |= AIC32X4_LDOIN_18_36; + } + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + tmp_reg |= AIC32X4_LDOIN2HP; + } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); + + /* Do DACs need to be swapped? */ + if (aic32x4->swapdacs) { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN); + } else { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN); + } + + /* Mic PGA routing */ + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + } + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + } + + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_add_controls(codec, aic32x4_snd_controls, + ARRAY_SIZE(aic32x4_snd_controls)); + aic32x4_add_widgets(codec); + + return 0; +} + +static int aic32x4_remove(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { + .read = aic32x4_read, + .write = aic32x4_write, + .probe = aic32x4_probe, + .remove = aic32x4_remove, + .suspend = aic32x4_suspend, + .resume = aic32x4_resume, + .set_bias_level = aic32x4_set_bias_level, +}; + +static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic32x4_pdata *pdata = i2c->dev.platform_data; + struct aic32x4_priv *aic32x4; + int ret; + + aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->control_data = i2c; + i2c_set_clientdata(i2c, aic32x4); + + if (pdata) { + aic32x4->power_cfg = pdata->power_cfg; + aic32x4->swapdacs = pdata->swapdacs; + aic32x4->micpga_routing = pdata->micpga_routing; + } else { + aic32x4->power_cfg = 0; + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret < 0) + kfree(aic32x4); + return ret; +} + +static __devexit int aic32x4_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + }, + .probe = aic32x4_i2c_probe, + .remove = __devexit_p(aic32x4_i2c_remove), + .id_table = aic32x4_i2c_id, +}; + +static int __init aic32x4_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", + ret); + } + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ + i2c_del_driver(&aic32x4_i2c_driver); +} +module_exit(aic32x4_exit); + +MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); +MODULE_AUTHOR("Javier Martin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h new file mode 100644 index 000000000000..aae2b2440398 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -0,0 +1,143 @@ +/* + * tlv320aic32x4.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + + +#ifndef _TLV320AIC32X4_H +#define _TLV320AIC32X4_H + +/* tlv320aic32x4 register space (in decimal to match datasheet) */ + +#define AIC32X4_PAGE1 128 + +#define AIC32X4_PSEL 0 +#define AIC32X4_RESET 1 +#define AIC32X4_CLKMUX 4 +#define AIC32X4_PLLPR 5 +#define AIC32X4_PLLJ 6 +#define AIC32X4_PLLDMSB 7 +#define AIC32X4_PLLDLSB 8 +#define AIC32X4_NDAC 11 +#define AIC32X4_MDAC 12 +#define AIC32X4_DOSRMSB 13 +#define AIC32X4_DOSRLSB 14 +#define AIC32X4_NADC 18 +#define AIC32X4_MADC 19 +#define AIC32X4_AOSR 20 +#define AIC32X4_CLKMUX2 25 +#define AIC32X4_CLKOUTM 26 +#define AIC32X4_IFACE1 27 +#define AIC32X4_IFACE2 28 +#define AIC32X4_IFACE3 29 +#define AIC32X4_BCLKN 30 +#define AIC32X4_IFACE4 31 +#define AIC32X4_IFACE5 32 +#define AIC32X4_IFACE6 33 +#define AIC32X4_DOUTCTL 53 +#define AIC32X4_DINCTL 54 +#define AIC32X4_DACSPB 60 +#define AIC32X4_ADCSPB 61 +#define AIC32X4_DACSETUP 63 +#define AIC32X4_DACMUTE 64 +#define AIC32X4_LDACVOL 65 +#define AIC32X4_RDACVOL 66 +#define AIC32X4_ADCSETUP 81 +#define AIC32X4_ADCFGA 82 +#define AIC32X4_LADCVOL 83 +#define AIC32X4_RADCVOL 84 +#define AIC32X4_LAGC1 86 +#define AIC32X4_LAGC2 87 +#define AIC32X4_LAGC3 88 +#define AIC32X4_LAGC4 89 +#define AIC32X4_LAGC5 90 +#define AIC32X4_LAGC6 91 +#define AIC32X4_LAGC7 92 +#define AIC32X4_RAGC1 94 +#define AIC32X4_RAGC2 95 +#define AIC32X4_RAGC3 96 +#define AIC32X4_RAGC4 97 +#define AIC32X4_RAGC5 98 +#define AIC32X4_RAGC6 99 +#define AIC32X4_RAGC7 100 +#define AIC32X4_PWRCFG (AIC32X4_PAGE1 + 1) +#define AIC32X4_LDOCTL (AIC32X4_PAGE1 + 2) +#define AIC32X4_OUTPWRCTL (AIC32X4_PAGE1 + 9) +#define AIC32X4_CMMODE (AIC32X4_PAGE1 + 10) +#define AIC32X4_HPLROUTE (AIC32X4_PAGE1 + 12) +#define AIC32X4_HPRROUTE (AIC32X4_PAGE1 + 13) +#define AIC32X4_LOLROUTE (AIC32X4_PAGE1 + 14) +#define AIC32X4_LORROUTE (AIC32X4_PAGE1 + 15) +#define AIC32X4_HPLGAIN (AIC32X4_PAGE1 + 16) +#define AIC32X4_HPRGAIN (AIC32X4_PAGE1 + 17) +#define AIC32X4_LOLGAIN (AIC32X4_PAGE1 + 18) +#define AIC32X4_LORGAIN (AIC32X4_PAGE1 + 19) +#define AIC32X4_HEADSTART (AIC32X4_PAGE1 + 20) +#define AIC32X4_MICBIAS (AIC32X4_PAGE1 + 51) +#define AIC32X4_LMICPGAPIN (AIC32X4_PAGE1 + 52) +#define AIC32X4_LMICPGANIN (AIC32X4_PAGE1 + 54) +#define AIC32X4_RMICPGAPIN (AIC32X4_PAGE1 + 55) +#define AIC32X4_RMICPGANIN (AIC32X4_PAGE1 + 57) +#define AIC32X4_FLOATINGINPUT (AIC32X4_PAGE1 + 58) +#define AIC32X4_LMICPGAVOL (AIC32X4_PAGE1 + 59) +#define AIC32X4_RMICPGAVOL (AIC32X4_PAGE1 + 60) + +#define AIC32X4_FREQ_12000000 12000000 +#define AIC32X4_FREQ_24000000 24000000 +#define AIC32X4_FREQ_25000000 25000000 + +#define AIC32X4_WORD_LEN_16BITS 0x00 +#define AIC32X4_WORD_LEN_20BITS 0x01 +#define AIC32X4_WORD_LEN_24BITS 0x02 +#define AIC32X4_WORD_LEN_32BITS 0x03 + +#define AIC32X4_I2S_MODE 0x00 +#define AIC32X4_DSP_MODE 0x01 +#define AIC32X4_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC32X4_LEFT_JUSTIFIED_MODE 0x03 + +#define AIC32X4_AVDDWEAKDISABLE 0x08 +#define AIC32X4_LDOCTLEN 0x01 + +#define AIC32X4_LDOIN_18_36 0x01 +#define AIC32X4_LDOIN2HP 0x02 + +#define AIC32X4_DACSPBLOCK_MASK 0x1f +#define AIC32X4_ADCSPBLOCK_MASK 0x1f + +#define AIC32X4_PLLJ_SHIFT 6 +#define AIC32X4_DOSRMSB_SHIFT 4 + +#define AIC32X4_PLLCLKIN 0x03 + +#define AIC32X4_MICBIAS_LDOIN 0x08 +#define AIC32X4_MICBIAS_2075V 0x60 + +#define AIC32X4_LMICPGANIN_IN2R_10K 0x10 +#define AIC32X4_RMICPGANIN_IN1L_10K 0x10 + +#define AIC32X4_LMICPGAVOL_NOGAIN 0x80 +#define AIC32X4_RMICPGAVOL_NOGAIN 0x80 + +#define AIC32X4_BCLKMASTER 0x08 +#define AIC32X4_WCLKMASTER 0x04 +#define AIC32X4_PLLEN (0x01 << 7) +#define AIC32X4_NDACEN (0x01 << 7) +#define AIC32X4_MDACEN (0x01 << 7) +#define AIC32X4_NADCEN (0x01 << 7) +#define AIC32X4_MADCEN (0x01 << 7) +#define AIC32X4_BCLKEN (0x01 << 7) +#define AIC32X4_DACEN (0x03 << 6) +#define AIC32X4_RDAC2LCHN (0x02 << 2) +#define AIC32X4_LDAC2RCHN (0x02 << 4) +#define AIC32X4_LDAC2LCHN (0x01 << 4) +#define AIC32X4_RDAC2RCHN (0x01 << 2) + +#define AIC32X4_SSTEP2WCLK 0x01 +#define AIC32X4_MUTEON 0x0C +#define AIC32X4_DACMOD2BCLK 0x01 + +#endif /* _TLV320AIC32X4_H */ -- cgit v1.2.3 From 89b95ac09e408b5d88a8b3792dc76c863e45fb31 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 16:38:44 +0000 Subject: ASoC: Add DAPM widget and path data to CODEC driver structure Allow a slight simplification of CODEC drivers by allowing DAPM routes and widgets to be provided in a table. They will be instantiated at the end of CODEC probe. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 12 ++++++++++-- 2 files changed, 16 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9c2a6dd170f1..6f197589b6d7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -562,6 +562,12 @@ struct snd_soc_codec_driver { pm_message_t state); int (*resume)(struct snd_soc_codec *); + /* Default DAPM setup, added after probe() is run */ + const struct snd_soc_dapm_widget *dapm_widgets; + int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + int num_dapm_routes; + /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be34f6b95386..c12f2bd23a4e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1464,6 +1464,7 @@ static int soc_probe_codec(struct snd_soc_card *card, struct snd_soc_codec *codec) { int ret = 0; + const struct snd_soc_codec_driver *driver = codec->driver; codec->card = card; codec->dapm.card = card; @@ -1472,8 +1473,8 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - if (codec->driver->probe) { - ret = codec->driver->probe(codec); + if (driver->probe) { + ret = driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", @@ -1482,6 +1483,13 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, + driver->num_dapm_widgets); + if (driver->dapm_routes) + snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, + driver->num_dapm_routes); + soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ -- cgit v1.2.3 From ec4ee52a8f5fb5b8e235ae9f02589d60d54740cc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 20:58:11 +0000 Subject: ASoC: Provide CODEC clocking operations and API calls When multi component systems use DAIless amplifiers which require clocking configuration it is at best hard to use the current clocking API as this requires a DAI even though the device may not even have one. Address this by adding set_sysclk() and set_pll() operations and APIs for CODECs. In order to avoid issues with devices which could be used either with or without DAIs make the DAI variants call through to their CODEC counterparts if there is no DAI specific operation. Converting over entirely would create problems for multi-DAI devices which offer per-DAI clocking setup. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 11 +++++++++++ sound/soc/soc-core.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 57 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6f197589b6d7..14f601f3e189 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -259,6 +259,11 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir); +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out); + int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); @@ -568,6 +573,12 @@ struct snd_soc_codec_driver { const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + /* codec wide operations */ + int (*set_sysclk)(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out); + /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c12f2bd23a4e..c2ec6cb05631 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3064,11 +3064,33 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, { if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); + else if (dai->codec && dai->codec->driver->set_sysclk) + return dai->codec->driver->set_sysclk(dai->codec, clk_id, + freq, dir); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); +/** + * snd_soc_codec_set_sysclk - configure CODEC system or master clock. + * @codec: CODEC + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + if (codec->driver->set_sysclk) + return codec->driver->set_sysclk(codec, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); + /** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI @@ -3105,11 +3127,35 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, if (dai->driver && dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); + else if (dai->codec && dai->codec->driver->set_pll) + return dai->codec->driver->set_pll(dai->codec, pll_id, source, + freq_in, freq_out); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); +/* + * snd_soc_codec_set_pll - configure codec PLL. + * @codec: CODEC + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + if (codec->driver->set_pll) + return codec->driver->set_pll(codec, pll_id, source, + freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); + /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI -- cgit v1.2.3 From efb7ac3f9c28fcb379c51f987b63174f727b7453 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 17:23:24 +0000 Subject: ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew() Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++++++++-- 3 files changed, 47 insertions(+), 17 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 14f601f3e189..bfa4836ea107 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -340,7 +340,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); *Controls */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name); + void *data, char *long_name, + const char *prefix); int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index db3075dd11fe..17efacdb248a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2344,22 +2344,45 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); * @_template: control template * @data: control private data * @long_name: control long name + * @prefix: control name prefix * * Create a new mixer control from a template control. * * Returns 0 for success, else error. */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name) + void *data, char *long_name, + const char *prefix) { struct snd_kcontrol_new template; + struct snd_kcontrol *kcontrol; + char *name = NULL; + int name_len; memcpy(&template, _template, sizeof(template)); - if (long_name) - template.name = long_name; template.index = 0; - return snd_ctl_new1(&template, data); + if (!long_name) + long_name = template.name; + + if (prefix) { + name_len = strlen(long_name) + strlen(prefix) + 2; + name = kmalloc(name_len, GFP_ATOMIC); + if (!name) + return NULL; + + snprintf(name, name_len, "%s %s", prefix, long_name); + + template.name = name; + } else { + template.name = long_name; + } + + kcontrol = snd_ctl_new1(&template, data); + + kfree(name); + + return kcontrol; } EXPORT_SYMBOL_GPL(snd_soc_cnew); @@ -2378,22 +2401,16 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; - char prefixed_name[44], *name; int err, i; for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - if (codec->name_prefix) { - snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", - codec->name_prefix, control->name); - name = prefixed_name; - } else { - name = control->name; - } - err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); + err = snd_ctl_add(card, snd_soc_cnew(control, codec, + control->name, + codec->name_prefix)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, name, err); + codec->name, control->name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 570db8819d9b..a6fb85d46416 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -369,6 +369,12 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, size_t name_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; + const char *prefix; + + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -409,7 +415,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, - path->long_name); + path->long_name, prefix); ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { dev_err(dapm->dev, @@ -431,6 +437,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; + const char *prefix; int ret = 0; if (!w->num_kcontrols) { @@ -438,7 +445,12 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; + + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) -- cgit v1.2.3 From 3cbdd7533148f00444013700af89548b8cf32646 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Aug 2008 16:09:01 +0200 Subject: ALSA: Add snd_ctl_activate_id() Added a new API function snd_ctl_activate_id() for activate / inactivate the control element dynamically. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/control.h | 2 ++ sound/core/control.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+) (limited to 'include') diff --git a/include/sound/control.h b/include/sound/control.h index 7715e6f00d38..e67db2869360 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -115,6 +115,8 @@ int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); +int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, + int active); struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid); struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id); diff --git a/sound/core/control.c b/sound/core/control.c index 9ce00ed20fba..db51e4e64984 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -465,6 +465,52 @@ error: return ret; } +/** + * snd_ctl_activate_id - activate/inactivate the control of the given id + * @card: the card instance + * @id: the control id to activate/inactivate + * @active: non-zero to activate + * + * Finds the control instance with the given id, and activate or + * inactivate the control together with notification, if changed. + * + * Returns 0 if unchanged, 1 if changed, or a negative error code on failure. + */ +int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, + int active) +{ + struct snd_kcontrol *kctl; + struct snd_kcontrol_volatile *vd; + unsigned int index_offset; + int ret; + + down_write(&card->controls_rwsem); + kctl = snd_ctl_find_id(card, id); + if (kctl == NULL) { + ret = -ENOENT; + goto unlock; + } + index_offset = snd_ctl_get_ioff(kctl, &kctl->id); + vd = &kctl->vd[index_offset]; + ret = 0; + if (active) { + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)) + goto unlock; + vd->access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } else { + if (vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) + goto unlock; + vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } + ret = 1; + unlock: + up_write(&card->controls_rwsem); + if (ret > 0) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id); + return ret; +} +EXPORT_SYMBOL_GPL(snd_ctl_activate_id); + /** * snd_ctl_rename_id - replace the id of a control on the card * @card: the card instance -- cgit v1.2.3