diff options
Diffstat (limited to 'sound')
123 files changed, 2628 insertions, 1159 deletions
diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index f84f3e50578..fa8ab2815a9 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -176,7 +176,7 @@ int soundbus_add_one(struct soundbus_dev *dev) return -EINVAL; } - snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount); + dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount); dev->ofdev.dev.bus = &soundbus_bus_type; return of_device_register(&dev->ofdev); } diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index e6beb92c693..b4590df0746 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; } diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 622cd37a011..a0f223c13f6 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -8,7 +8,7 @@ #ifndef __SOUNDBUS_H #define __SOUNDBUS_H -#include <asm/of_device.h> +#include <linux/of_device.h> #include <sound/pcm.h> #include <linux/list.h> diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 99026dfb81e..34c1d94f921 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -50,7 +50,7 @@ unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) mutex_lock(&car_mutex); /* set up primary or secondary codec space */ - if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS) + if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; else reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -90,7 +90,7 @@ void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec space */ - if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS) + if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; else reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -200,7 +200,7 @@ static inline void pxa_ac97_cold_pxa3xx(void) bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { #ifdef CONFIG_PXA25x - if (cpu_is_pxa21x() || cpu_is_pxa25x()) + if (cpu_is_pxa25x()) pxa_ac97_warm_pxa25x(); else #endif @@ -230,7 +230,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { #ifdef CONFIG_PXA25x - if (cpu_is_pxa21x() || cpu_is_pxa25x()) + if (cpu_is_pxa25x()) pxa_ac97_cold_pxa25x(); else #endif @@ -301,7 +301,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) { + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); @@ -325,7 +325,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (ret < 0) goto err; - if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) { + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index cba71d86754..c2635beb4c8 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -44,7 +44,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { .name = "AC97 PCM out", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRTXPCDR, + .drcmr = &DRCMR(12), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -52,7 +52,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = { .name = "AC97 PCM in", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRRXPCDR, + .drcmr = &DRCMR(11), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb77cb9..75a0d746fb6 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = diff --git a/sound/core/control.c b/sound/core/control.c index 6d71f9a7ccb..636b3b52ef8 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -113,7 +113,6 @@ static int snd_ctl_release(struct inode *inode, struct file *file) unsigned int idx; ctl = file->private_data; - fasync_helper(-1, file, 0, &ctl->fasync); file->private_data = NULL; card = ctl->card; write_lock_irqsave(&card->ctl_files_rwlock, flags); @@ -225,8 +224,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.id.iface = ncontrol->iface; kctl.id.device = ncontrol->device; kctl.id.subdevice = ncontrol->subdevice; - if (ncontrol->name) + if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); + if (strcmp(ncontrol->name, kctl.id.name) != 0) + snd_printk(KERN_WARNING + "Control name '%s' truncated to '%s'\n", + ncontrol->name, kctl.id.name); + } kctl.id.index = ncontrol->index; kctl.count = ncontrol->count ? ncontrol->count : 1; access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : diff --git a/sound/core/init.c b/sound/core/init.c index 8af467df924..b47ff8b44be 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -264,8 +264,11 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) } spin_unlock(&shutdown_lock); - if (likely(df)) + if (likely(df)) { + if ((file->f_flags & FASYNC) && df->disconnected_f_op->fasync) + df->disconnected_f_op->fasync(-1, file, 0); return df->disconnected_f_op->release(inode, file); + } panic("%s(%p, %p) failed!", __func__, inode, file); } @@ -549,9 +552,9 @@ int snd_card_register(struct snd_card *card) return -EINVAL; #ifndef CONFIG_SYSFS_DEPRECATED if (!card->card_dev) { - card->card_dev = device_create_drvdata(sound_class, card->dev, - MKDEV(0, 0), NULL, - "card%i", card->number); + card->card_dev = device_create(sound_class, card->dev, + MKDEV(0, 0), NULL, + "card%i", card->number); if (IS_ERR(card->card_dev)) card->card_dev = NULL; } diff --git a/sound/core/jack.c b/sound/core/jack.c index 8133a2b173a..bd2d9e6b55e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -147,6 +147,9 @@ EXPORT_SYMBOL(snd_jack_set_parent); */ void snd_jack_report(struct snd_jack *jack, int status) { + if (!jack) + return; + if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index a7b46ec72f3..1b3534d6768 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -33,9 +33,6 @@ #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> -#ifdef CONFIG_SBUS -#include <asm/sbus.h> -#endif MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>"); @@ -162,39 +159,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, } #endif /* CONFIG_HAS_DMA */ -#ifdef CONFIG_SBUS - -static void *snd_malloc_sbus_pages(struct device *dev, size_t size, - dma_addr_t *dma_addr) -{ - struct sbus_dev *sdev = (struct sbus_dev *)dev; - int pg; - void *res; - - if (WARN_ON(!dma_addr)) - return NULL; - pg = get_order(size); - res = sbus_alloc_consistent(sdev, PAGE_SIZE * (1 << pg), dma_addr); - if (res != NULL) - inc_snd_pages(pg); - return res; -} - -static void snd_free_sbus_pages(struct device *dev, size_t size, - void *ptr, dma_addr_t dma_addr) -{ - struct sbus_dev *sdev = (struct sbus_dev *)dev; - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - dec_snd_pages(pg); - sbus_free_consistent(sdev, PAGE_SIZE * (1 << pg), ptr, dma_addr); -} - -#endif /* CONFIG_SBUS */ - /* * * ALSA generic memory management @@ -231,11 +195,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->area = snd_malloc_pages(size, (unsigned long)device); dmab->addr = 0; break; -#ifdef CONFIG_SBUS - case SNDRV_DMA_TYPE_SBUS: - dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr); - break; -#endif #ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); @@ -306,11 +265,6 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_CONTINUOUS: snd_free_pages(dmab->area, dmab->bytes); break; -#ifdef CONFIG_SBUS - case SNDRV_DMA_TYPE_SBUS: - snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); - break; -#endif #ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); @@ -419,7 +373,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "SBUS" }; + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; mutex_lock(&list_mutex); seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1af62b8b86c..e17836680f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2283,7 +2283,7 @@ static int snd_pcm_oss_open_file(struct file *file, int idx, err; struct snd_pcm_oss_file *pcm_oss_file; struct snd_pcm_substream *substream; - unsigned int f_mode = file->f_mode; + fmode_t f_mode = file->f_mode; if (rpcm_oss_file) *rpcm_oss_file = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6ea5cfb8399..921691080f3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, EXPORT_SYMBOL(snd_pcm_hw_rule_add); /** - * snd_pcm_hw_constraint_mask + * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. */ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask) @@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param } /** - * snd_pcm_hw_constraint_mask64 + * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the 64bit bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. */ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask) @@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par } /** - * snd_pcm_hw_constraint_integer + * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the integer constraint * @@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); /** - * snd_pcm_hw_constraint_minmax + * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the range * @min: the minimal value @@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_constraint_list + * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the list constraint @@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_ratnums + * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint @@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_ratdens + * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint @@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_msbits + * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule * @runtime: PCM runtime instance * @cond: condition bits * @width: sample bits width @@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_step + * snd_pcm_hw_constraint_step - add a hw constraint step rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the step constraint @@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm } /** - * snd_pcm_hw_constraint_pow2 + * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint @@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params) EXPORT_SYMBOL(_snd_pcm_hw_params_any); /** - * snd_pcm_hw_param_value + * snd_pcm_hw_param_value - return @params field @var value * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Return the value for field PAR if it's fixed in configuration space - * defined by PARAMS. Return -EINVAL otherwise + * Return the value for field @var if it's fixed in configuration space + * defined by @params. Return -%EINVAL otherwise. */ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) @@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_param_first + * snd_pcm_hw_param_first - refine config space and return minimum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values > minimum. Reduce configuration space accordingly. * Return the minimum. */ @@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_param_last + * snd_pcm_hw_param_last - refine config space and return maximum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values < maximum. Reduce configuration space accordingly. * Return the maximum. */ @@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, EXPORT_SYMBOL(snd_pcm_hw_param_last); /** - * snd_pcm_hw_param_choose + * snd_pcm_hw_param_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * - * Choose one configuration from configuration space defined by PARAMS + * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 89b7f549beb..ea2bf82c937 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -319,6 +319,7 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width); /** * snd_pcm_format_size - return the byte size of samples on the given format * @format: the format to check + * @samples: sampling rate * * Returns the byte size of the given samples for the format, or a * negative error code if unknown format. diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e61e12506de..a789efc9df3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_start = { }; /** - * snd_pcm_start + * snd_pcm_start - start all linked streams * @substream: the PCM substream instance - * - * Start all linked streams. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_stop = { }; /** - * snd_pcm_stop + * snd_pcm_stop - try to stop all running streams in the substream group * @substream: the PCM substream instance * @state: PCM state after stopping the stream * - * Try to stop all running streams in the substream group. - * The state of each stream is changed to the given value after that unconditionally. + * The state of each stream is then changed to the given state unconditionally. */ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) { @@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) EXPORT_SYMBOL(snd_pcm_stop); /** - * snd_pcm_drain_done + * snd_pcm_drain_done - stop the DMA only when the given stream is playback * @substream: the PCM substream * - * Stop the DMA only when the given stream is playback. - * The state is changed to SETUP. + * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) @@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_suspend = { }; /** - * snd_pcm_suspend + * snd_pcm_suspend - trigger SUSPEND to all linked streams * @substream: the PCM substream * - * Trigger SUSPEND to all linked streams. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend(struct snd_pcm_substream *substream) @@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream) EXPORT_SYMBOL(snd_pcm_suspend); /** - * snd_pcm_suspend_all + * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm * @pcm: the PCM instance * - * Trigger SUSPEND to all substreams in the given pcm. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend_all(struct snd_pcm *pcm) @@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_prepare = { }; /** - * snd_pcm_prepare + * snd_pcm_prepare - prepare the PCM substream to be triggerable * @substream: the PCM substream instance * @file: file to refer f_flags - * - * Prepare the PCM substream to be triggerable. */ static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file) @@ -2177,7 +2169,6 @@ static int snd_pcm_release(struct inode *inode, struct file *file) if (snd_BUG_ON(!substream)) return -ENXIO; pcm = substream->pcm; - fasync_helper(-1, file, 0, &substream->runtime->fasync); mutex_lock(&pcm->open_mutex); snd_pcm_release_substream(substream); kfree(pcm_file); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c4995c9f573..39672f68ce5 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -148,6 +148,8 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream) static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up) { + if (!substream->opened) + return; if (up) { tasklet_hi_schedule(&substream->runtime->tasklet); } else { @@ -158,6 +160,8 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up) { + if (!substream->opened) + return; substream->ops->trigger(substream, up); if (!up && substream->runtime->event) tasklet_kill(&substream->runtime->tasklet); @@ -857,6 +861,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; + if (!substream->opened) + return -EBADFD; if (runtime->buffer == NULL) { snd_printd("snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; @@ -1126,6 +1132,8 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { + if (!substream->opened) + return -EBADFD; count = snd_rawmidi_transmit_peek(substream, buffer, count); if (count < 0) return count; diff --git a/sound/core/sound.c b/sound/core/sound.c index c0685e2f0af..44a69bb8d4f 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -274,9 +274,8 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, return minor; } snd_minors[minor] = preg; - preg->dev = device_create_drvdata(sound_class, device, - MKDEV(major, minor), - private_data, "%s", name); + preg->dev = device_create(sound_class, device, MKDEV(major, minor), + private_data, "%s", name); if (IS_ERR(preg->dev)) { snd_minors[minor] = NULL; mutex_unlock(&sound_mutex); diff --git a/sound/core/timer.c b/sound/core/timer.c index e582face89d..c584408c9f1 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1263,7 +1263,6 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (file->private_data) { tu = file->private_data; file->private_data = NULL; - fasync_helper(-1, file, 0, &tu->fasync); if (tu->timeri) snd_timer_close(tu->timeri); kfree(tu->queue); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index e5e749f3e0e..73be7e14a60 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -51,7 +51,7 @@ static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) if (err < 0) return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); - if (err) < 0) + if (err < 0) return err; return 0; } diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index ecdbeb6d360..7783843ca9a 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, /* get irq */ irq = platform_get_irq(pfdev, 0); if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, - pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", irq); @@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, ml403_ac97cr->irq); irq = platform_get_irq(pfdev, 1); if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, - pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", irq); diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c index cd9b83e7f7d..0444cdeb4be 100644 --- a/sound/drivers/pcsp/pcsp_input.c +++ b/sound/drivers/pcsp/pcsp_input.c @@ -24,13 +24,13 @@ static void pcspkr_do_sound(unsigned int count) spin_lock_irqsave(&i8253_lock, flags); if (count) { - /* enable counter 2 */ - outb_p(inb_p(0x61) | 3, 0x61); /* set command for counter 2, 2 byte write */ outb_p(0xB6, 0x43); /* select desired HZ */ outb_p(count & 0xff, 0x42); outb((count >> 8) & 0xff, 0x42); + /* enable counter 2 */ + outb_p(inb_p(0x61) | 3, 0x61); } else { /* disable counter 2 */ outb(inb_p(0x61) & 0xFC, 0x61); diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index e341f3f83b6..1f42e406311 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -34,7 +34,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->thalf = 0; if (!atomic_read(&chip->timer_active)) return HRTIMER_NORESTART; - hrtimer_forward(&chip->timer, chip->timer.expires, + hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), ktime_set(0, chip->ns_rem)); return HRTIMER_RESTART; } @@ -118,7 +118,8 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->ns_rem = PCSP_PERIOD_NS(); ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); chip->ns_rem -= ns; - hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns)); + hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), + ktime_set(0, ns)); return HRTIMER_RESTART; exit_nr_unlock2: diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 83e90057270..c13a178383b 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -87,8 +87,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) static int snd_tea575x_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long data) { - struct video_device *dev = video_devdata(file); - struct snd_tea575x *tea = video_get_drvdata(dev); + struct snd_tea575x *tea = video_drvdata(file); void __user *arg = (void __user *)data; switch(cmd) { @@ -175,6 +174,21 @@ static void snd_tea575x_release(struct video_device *vfd) { } +static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file) +{ + struct snd_tea575x *tea = video_drvdata(file); + + return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; +} + +static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file) +{ + struct snd_tea575x *tea = video_drvdata(file); + + clear_bit(0, &tea->in_use); + return 0; +} + /* * initialize all the tea575x chips */ @@ -193,9 +207,10 @@ void snd_tea575x_init(struct snd_tea575x *tea) tea->vd.release = snd_tea575x_release; video_set_drvdata(&tea->vd, tea); tea->vd.fops = &tea->fops; + tea->in_use = 0; tea->fops.owner = tea->card->module; - tea->fops.open = video_exclusive_open; - tea->fops.release = video_exclusive_release; + tea->fops.open = snd_tea575x_exclusive_open; + tea->fops.release = snd_tea575x_exclusive_release; tea->fops.ioctl = snd_tea575x_ioctl; if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) { snd_printk(KERN_ERR "unable to register tea575x tuner\n"); diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb41f76..ce0aa044e27 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -211,7 +211,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - select SND_HWDEP + select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index b68d20edc20..223a6c03881 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + dev_err(dev, "please specify irq\n"); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + dev_err(dev, "please specify dma1\n"); return 0; } return 1; diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index efa8c80d05b..374b7177e11 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -36,7 +36,7 @@ static int __devinit snd_adlib_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } return 1; @@ -55,13 +55,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) card = snd_card_new(index[n], id[n], THIS_MODULE, 0); if (!card) { - snd_printk(KERN_ERR "%s: could not create card\n", dev->bus_id); + dev_err(dev, "could not create card\n"); return -EINVAL; } card->private_data = request_region(port[n], 4, CRD_NAME); if (!card->private_data) { - snd_printk(KERN_ERR "%s: could not grab ports\n", dev->bus_id); + dev_err(dev, "could not grab ports\n"); error = -EBUSY; goto out; } @@ -73,13 +73,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3); if (error < 0) { - snd_printk(KERN_ERR "%s: could not create OPL\n", dev->bus_id); + dev_err(dev, "could not create OPL\n"); goto out; } error = snd_opl3_hwdep_new(opl3, 0, 0, NULL); if (error < 0) { - snd_printk(KERN_ERR "%s: could not create FM\n", dev->bus_id); + dev_err(dev, "could not create FM\n"); goto out; } @@ -87,7 +87,7 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) error = snd_card_register(card); if (error < 0) { - snd_printk(KERN_ERR "%s: could not register card\n", dev->bus_id); + dev_err(dev, "could not register card\n"); goto out; } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ddd289120aa..f019d449e2d 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -74,15 +74,15 @@ static int __devinit snd_cs4231_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + dev_err(dev, "please specify irq\n"); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + dev_err(dev, "please specify dma1\n"); return 0; } return 1; @@ -133,7 +133,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) mpu_port[n], 0, mpu_irq[n], mpu_irq[n] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) - printk(KERN_WARNING "%s: MPU401 not detected\n", dev->bus_id); + dev_warn(dev, "MPU401 not detected\n"); } snd_card_set_dev(card, dev); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 91f9c15d3e3..019c9401663 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -488,19 +488,19 @@ static int __devinit snd_cs423x_isa_match(struct device *pdev, return 0; if (port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", pdev->bus_id); + dev_err(pdev, "please specify port\n"); return 0; } if (cport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify cport\n", pdev->bus_id); + dev_err(pdev, "please specify cport\n"); return 0; } if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id); + dev_err(pdev, "please specify irq\n"); return 0; } if (dma1[dev] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", pdev->bus_id); + dev_err(pdev, "please specify dma1\n"); return 0; } return 1; diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index f88639ea64b..b46377139cf 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -88,16 +88,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ\n", - dev->bus_id); + dev_err(dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma8[n] == SNDRV_AUTO_DMA) { dma8[n] = snd_legacy_find_free_dma(possible_dmas); if (dma8[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA\n"); return -EBUSY; } } @@ -147,8 +145,7 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) if (snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - printk(KERN_WARNING "%s: opl3 not detected at 0x%lx\n", - dev->bus_id, chip->port); + dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); else { error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (error < 0) diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 8f914b37bf8..426532a4d73 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -90,24 +90,21 @@ static int __devinit snd_gusclassic_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ\n", - dev->bus_id); + dev_err(dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma1[n] == SNDRV_AUTO_DMA) { dma1[n] = snd_legacy_find_free_dma(possible_dmas); if (dma1[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA1\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2[n] == SNDRV_AUTO_DMA) { dma2[n] = snd_legacy_find_free_dma(possible_dmas); if (dma2[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA2\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA2\n"); return -EBUSY; } } @@ -174,8 +171,8 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n) error = -ENODEV; if (gus->max_flag || gus->ess_flag) { - snd_printk(KERN_ERR "%s: GUS Classic or ACE soundcard was " - "not detected at 0x%lx\n", dev->bus_id, gus->gf1.port); + dev_err(dev, "GUS Classic or ACE soundcard was " + "not detected at 0x%lx\n", gus->gf1.port); goto out; } diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index da13185eb0a..7ad4c3b41a8 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -106,16 +106,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ " - "for ES1688\n", dev->bus_id); + dev_err(dev, "unable to find a free IRQ for ES1688\n"); return -EBUSY; } } if (dma8[n] == SNDRV_AUTO_DMA) { dma8[n] = snd_legacy_find_free_dma(possible_dmas); if (dma8[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA " - "for ES1688\n", dev->bus_id); + dev_err(dev, "unable to find a free DMA for ES1688\n"); return -EBUSY; } } @@ -143,16 +141,14 @@ static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card, if (gf1_irq[n] == SNDRV_AUTO_IRQ) { gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs); if (gf1_irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ " - "for GF1\n", dev->bus_id); + dev_err(dev, "unable to find a free IRQ for GF1\n"); return -EBUSY; } } if (dma1[n] == SNDRV_AUTO_DMA) { dma1[n] = snd_legacy_find_free_dma(possible_dmas); if (dma1[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA " - "for GF1\n", dev->bus_id); + dev_err(dev, "unable to find a free DMA for GF1\n"); return -EBUSY; } } @@ -278,8 +274,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) error = -ENODEV; if (!gus->ess_flag) { - snd_printk(KERN_ERR "%s: GUS Extreme soundcard was not " - "detected at 0x%lx\n", dev->bus_id, gus->gf1.port); + dev_err(dev, "GUS Extreme soundcard was not " + "detected at 0x%lx\n", gus->gf1.port); goto out; } gus->codec_flag = 1; @@ -310,8 +306,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (snd_opl3_create(card, es1688->port, es1688->port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - printk(KERN_ERR "%s: opl3 not detected at 0x%lx\n", - dev->bus_id, es1688->port); + dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port); else { error = snd_opl3_hwdep_new(opl3, 0, 2, NULL); if (error < 0) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 336a3427790..667eccc676a 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -85,11 +85,11 @@ static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev) if (!enable[dev]) return 0; if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id); + dev_err(pdev, "please specify irq\n"); return 0; } if (dma8[dev] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma8\n", pdev->bus_id); + dev_err(pdev, "please specify dma8\n"); return 0; } return 1; diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c index b63839e8f9b..456a1b4d783 100644 --- a/sound/oss/ac97_codec.c +++ b/sound/oss/ac97_codec.c @@ -30,7 +30,7 @@ ************************************************************************** * * History - * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk> * Removed non existant WM9700 * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 * WM9712 and WM9717 diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 23018a7c063..81e1f443d09 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -93,7 +93,7 @@ static struct au1550_state { spinlock_t lock; struct mutex open_mutex; struct mutex sem; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; struct dmabuf { diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index d978b009656..1308d8d3418 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -129,7 +129,7 @@ typedef struct { int (*mixer_ioctl)(u_int, u_long); /* optional */ int (*write_sq_setup)(void); /* optional */ int (*read_sq_setup)(void); /* optional */ - int (*sq_open)(mode_t); /* optional */ + int (*sq_open)(fmode_t); /* optional */ int (*state_info)(char *, size_t); /* optional */ void (*abort_read)(void); /* optional */ int min_dsp_speed; @@ -235,7 +235,7 @@ struct sound_queue { */ int active; wait_queue_head_t action_queue, open_queue, sync_queue; - int open_mode; + int non_blocking; int busy, syncing, xruns, died; }; diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 285239d64b8..4d45bd63718 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -143,7 +143,7 @@ static int AtaMixerIoctl(u_int cmd, u_long arg); static int TTMixerIoctl(u_int cmd, u_long arg); static int FalconMixerIoctl(u_int cmd, u_long arg); static int AtaWriteSqSetup(void); -static int AtaSqOpen(mode_t mode); +static int AtaSqOpen(fmode_t mode); static int TTStateInfo(char *buffer, size_t space); static int FalconStateInfo(char *buffer, size_t space); @@ -1461,7 +1461,7 @@ static int AtaWriteSqSetup(void) return 0 ; } -static int AtaSqOpen(mode_t mode) +static int AtaSqOpen(fmode_t mode) { write_sq_ignore_int = 1; return 0 ; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 95fc5c68175..793b7f47843 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -212,7 +212,7 @@ static int irq_installed; #endif /* MODULE */ /* control over who can modify resources shared between play/record */ -static mode_t shared_resource_owner; +static fmode_t shared_resource_owner; static int shared_resources_initialised; /* @@ -603,7 +603,7 @@ static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft, while (uLeft) { while (write_sq.count >= write_sq.max_active) { sq_play(); - if (write_sq.open_mode & O_NONBLOCK) + if (write_sq.non_blocking) return uWritten > 0 ? uWritten : -EAGAIN; SLEEP(write_sq.action_queue); if (signal_pending(current)) @@ -668,7 +668,7 @@ static inline void sq_init_waitqueue(struct sound_queue *sq) #if 0 /* blocking open() */ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, - mode_t mode) + fmode_t mode) { if (file->f_mode & mode) { sq->busy = 0; /* CHECK: IS THIS OK??? */ @@ -677,7 +677,7 @@ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, } #endif -static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, +static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode, int numbufs, int bufsize) { int rc = 0; @@ -718,7 +718,7 @@ static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, return rc; } - sq->open_mode = file->f_mode; + sq->non_blocking = file->f_flags & O_NONBLOCK; } return rc; } @@ -891,10 +891,10 @@ static int sq_release(struct inode *inode, struct file *file) is the owner - if we have problems. */ -static int shared_resources_are_mine(mode_t md) +static int shared_resources_are_mine(fmode_t md) { if (shared_resource_owner) - return (shared_resource_owner & md ) ; + return (shared_resource_owner & md) != 0; else { shared_resource_owner = md ; return 1 ; diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index eb9bc365530..c180598f171 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -1,7 +1,7 @@ /* * Initialisation code for Cyrix/NatSemi VSA1 softaudio * - * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> * * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. * The older version (VSA1) provides fairly good soundblaster emulation diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h index 61b3955481c..c8be47ec2b7 100644 --- a/sound/oss/msnd.h +++ b/sound/oss/msnd.h @@ -211,7 +211,7 @@ typedef struct multisound_dev { /* State variables */ enum { msndClassic, msndPinnacle } type; - mode_t mode; + fmode_t mode; unsigned long flags; #define F_RESETTING 0 #define F_HAVEDIGITAL 1 diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index b493660deb3..e5d42399491 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -26,7 +26,7 @@ #include <asm/cpu/dac.h> #include <asm/cpu/timer.h> #include <asm/machvec.h> -#include <asm/hp6xx.h> +#include <mach/hp6xx.h> #include <asm/hd64461.h> #define MODNAME "sh_dac_audio" diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 1a00a321061..55271fbe7f4 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -110,24 +110,16 @@ struct channel_info { #define OPEN_WRITE PCM_ENABLE_OUTPUT #define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) -#if OPEN_READ == FMODE_READ && OPEN_WRITE == FMODE_WRITE - -static inline int translate_mode(struct file *file) -{ - return file->f_mode; -} - -#else - static inline int translate_mode(struct file *file) { - return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | - ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); + if (OPEN_READ == (__force int)FMODE_READ && + OPEN_WRITE == (__force int)FMODE_WRITE) + return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE)); + else + return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | + ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); } -#endif - - #include "sound_calls.h" #include "dev_table.h" diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 7d89c081a08..61aaedae6b7 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -560,19 +560,18 @@ static int __init oss_init(void) sound_dmap_flag = (dmabuf > 0 ? 1 : 0); for (i = 0; i < ARRAY_SIZE(dev_list); i++) { - device_create_drvdata(sound_class, NULL, - MKDEV(SOUND_MAJOR, dev_list[i].minor), - NULL, "%s", dev_list[i].name); + device_create(sound_class, NULL, + MKDEV(SOUND_MAJOR, dev_list[i].minor), NULL, + "%s", dev_list[i].name); if (!dev_list[i].num) continue; for (j = 1; j < *dev_list[i].num; j++) - device_create_drvdata(sound_class, NULL, - MKDEV(SOUND_MAJOR, - dev_list[i].minor + (j*0x10)), - NULL, - "%s%d", dev_list[i].name, j); + device_create(sound_class, NULL, + MKDEV(SOUND_MAJOR, + dev_list[i].minor + (j*0x10)), + NULL, "%s%d", dev_list[i].name, j); } if (sound_nblocks >= 1024) diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 044453a4ee5..41562ecde5b 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -295,7 +295,7 @@ struct cs4297a_state { struct mutex open_mutex; struct mutex open_sem_adc; struct mutex open_sem_dac; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; wait_queue_head_t open_wait_adc; wait_queue_head_t open_wait_dac; diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index dcbb3f739e6..78b8acc7c3b 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -1509,7 +1509,7 @@ typedef struct vwsnd_dev { struct mutex open_mutex; struct mutex io_mutex; struct mutex mix_mutex; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; lithium_t lith; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acbca8c..bd510eceff1 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1927,9 +1927,9 @@ static int snd_ac97_dev_register(struct snd_device *device) ac97->dev.bus = &ac97_bus_type; ac97->dev.parent = ac97->bus->card->dev; ac97->dev.release = ac97_device_release; - snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - ac97->bus->card->number, ac97->num, - snd_ac97_get_short_name(ac97)); + dev_set_name(&ac97->dev, "%d-%d:%s", + ac97->bus->card->number, ac97->num, + snd_ac97_get_short_name(ac97)); if ((err = device_register(&ac97->dev)) < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); ac97->dev.bus = NULL; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe98a6..6e831aff1bd 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) } /* - * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk> * removed broken wolfson00 patch. * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a976ef2..a7f38e63303 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card, goto free_and_ret; chip->bar = pci_resource_start(pci, 0); - chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0)); + chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { printk(KERN_ERR PFX "unable to reserve region.\n"); err = -EBUSY; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b8c80..226fe8237d3 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f106306c7f..0e6e5cc1c50 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e49007..a36d4d1fd41 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { printk(KERN_ERR "MMIO area remap failed.\n"); err = -ENOMEM; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d973540..1aa1c040254 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, pci_disable_device(pci); return err; } - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { snd_printk(KERN_ERR "cannot remap io memory\n"); err = -ENOMEM; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index a7d89662acf..88fbf285d2b 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -759,7 +759,6 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); - } #endif return 0; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f7c45..192e7842e18 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + chip->ba0 = pci_ioremap_bar(pci, 0); + chip->ba1 = pci_ioremap_bar(pci, 1); if (!chip->ba0 || !chip->ba1) { snd_cs4281_free(chip); return -ENOMEM; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68e997..6dea5b5cc77 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -2,7 +2,7 @@ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio * * (C) Copyright 2007 Ash Willis <ashwillis@programmer.net> - * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> * * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did * mess with it a bit. The chip seems to have to have trouble with full duplex @@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, } chip->pci_base = pci_resource_start(pci, 0); - mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { kfree(chip); pci_disable_device(pci); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2f283ea6ad9..de5ee8f097f 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1464,6 +1464,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", @@ -1473,6 +1474,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102, .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", @@ -1482,6 +1484,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , /* Audigy 2 */ /* Tested by James@superbug.co.uk 3rd July 2005 */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c04ac38e2b2..35722ec920c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2186,7 +2186,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); err = -ENXIO; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 5b442383fcd..58d7cda03de 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2688,12 +2688,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } err = snd_ice1712_ac97_mixer(ice); if (err < 0) { @@ -2715,12 +2716,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, } } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } if (!c->no_mpu401) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c88d1eace1c..19d3391e229 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2702,6 +2702,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000), + SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000), SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000), { } /* terminator */ }; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce649a6..ae7601f353a 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, } for (i = 0; i < 2; i++) { mgr->mem[i].phys = pci_resource_start(pci, i); - mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys, - pci_resource_len(pci, i)); + mgr->mem[i].virt = pci_ioremap_bar(pci, i); if (!mgr->mem[i].virt) { printk(KERN_ERR "unable to remap resource 0x%lx\n", mgr->mem[i].phys); diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 0e06c6c9fcc..73de6e989b3 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1229,8 +1229,11 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id return -ENOMEM; } - if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) + if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) { + kfree(mgr); + pci_disable_device(pci); return -ENODEV; + } card_name = pcxhr_board_params[pci_id->driver_data].board_name; mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips; mgr->capture_chips = pcxhr_board_params[pci_id->driver_data].capture_chips; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d723543bead..736246f98ac 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4548,11 +4548,20 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne { struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; + int err; switch (cmd) { case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: { struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg; + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + + err = hdsp_check_for_firmware(hdsp, 1); + if (err < 0) + return err; + if (!(hdsp->state & HDSP_FirmwareLoaded)) { snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n"); return -EINVAL; @@ -4572,10 +4581,14 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne unsigned long flags; int i; - if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); - return -EINVAL; - } + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + + err = hdsp_check_for_firmware(hdsp, 1); + if (err < 0) + return err; + spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); @@ -5045,6 +5058,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { #ifdef HDSP_FW_LOADER if ((err = hdsp_request_fw_loader(hdsp)) < 0) @@ -5057,7 +5074,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* init is complete, we return */ return 0; #endif - /* no iobox connected, we defer initialization */ + /* we defer initialization */ snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n"); if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 20d0e328288..8f9e3859c37 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -666,6 +666,7 @@ static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM; card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL; + memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8); ret = snd_ps3_change_avsetting(card); @@ -685,6 +686,7 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) { struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); struct snd_ps3_avsetting_info avs; + int ret; avs = card->avs; @@ -729,19 +731,92 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) return 1; } - if ((card->avs.avs_audio_width != avs.avs_audio_width) || - (card->avs.avs_audio_rate != avs.avs_audio_rate)) { - card->avs = avs; - snd_ps3_change_avsetting(card); + memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8); + if (memcmp(&card->avs, &avs, sizeof(avs))) { pr_debug("%s: after freq=%d width=%d\n", __func__, card->avs.avs_audio_rate, card->avs.avs_audio_width); - return 0; + card->avs = avs; + snd_ps3_change_avsetting(card); + ret = 0; } else + ret = 1; + + /* check CS non-audio bit and mute accordingly */ + if (avs.avs_cs_info[0] & 0x02) + ps3av_audio_mute_analog(1); /* mute if non-audio */ + else + ps3av_audio_mute_analog(0); + + return ret; +} + +/* + * SPDIF status bits controls + */ +static int snd_ps3_spdif_mask_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +/* FIXME: ps3av_set_audio_mode() assumes only consumer mode */ +static int snd_ps3_spdif_cmask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memset(ucontrol->value.iec958.status, 0xff, 8); + return 0; +} + +static int snd_ps3_spdif_pmask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int snd_ps3_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memcpy(ucontrol->value.iec958.status, ps3av_mode_cs_info, 8); + return 0; +} + +static int snd_ps3_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (memcmp(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8)) { + memcpy(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8); return 1; + } + return 0; } +static struct snd_kcontrol_new spdif_ctls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_cmask_get, + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_pmask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_default_get, + .put = snd_ps3_spdif_default_put, + }, +}; static int snd_ps3_map_mmio(void) @@ -842,7 +917,7 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) { - int ret; + int i, ret; u64 lpar_addr, lpar_size; BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1)); @@ -903,6 +978,15 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) strcpy(the_card.card->driver, "PS3"); strcpy(the_card.card->shortname, "PS3"); strcpy(the_card.card->longname, "PS3 sound"); + + /* create control elements */ + for (i = 0; i < ARRAY_SIZE(spdif_ctls); i++) { + ret = snd_ctl_add(the_card.card, + snd_ctl_new1(&spdif_ctls[i], &the_card)); + if (ret < 0) + goto clean_card; + } + /* create PCM devices instance */ /* NOTE:this driver works assuming pcm:substream = 1:1 */ ret = snd_pcm_new(the_card.card, diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h index 4b7e6fbbe50..326fb29e82d 100644 --- a/sound/ppc/snd_ps3.h +++ b/sound/ppc/snd_ps3.h @@ -51,6 +51,7 @@ struct snd_ps3_avsetting_info { uint32_t avs_audio_width; uint32_t avs_audio_format; /* fixed */ uint32_t avs_audio_source; /* fixed */ + unsigned char avs_cs_info[8]; }; /* * PS3 audio 'card' instance diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 98a2d5826a8..b1966e4dfcd 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to SPKOUT */ {"Ext Spk", NULL, "SPKOUTP"}, {"Ext Spk", NULL, "SPKOUTN"}, @@ -312,9 +312,6 @@ static const char *intercon[][3] = { {"Mic Bias", NULL, "Int Mic"}, {"MICN", NULL, "Mic Bias"}, {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, }; @@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) /* * Setup audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + /* always connected pins */ diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 905186502e0..85a883299c2 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -8,20 +8,3 @@ config SND_AT91_SOC config SND_AT91_SOC_SSC tristate - -config SND_AT91_SOC_ETI_B1_WM8731 - tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on WM8731-based - Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. - -config SND_AT91_SOC_ETI_SLAVE - bool "Run codec in slave Mode on Endrelia boards" - depends on SND_AT91_SOC_ETI_B1_WM8731 - default n - help - Say Y if you want to run with the AT91 SSC generating the BCLK - and LRC signals on Endrelia boards. diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile index f23da17cc32..b817f11df28 100644 --- a/sound/soc/at91/Makefile +++ b/sound/soc/at91/Makefile @@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o - -# AT91 Machine Support -snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o - -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index a5b1a79ebff..1b61cc46126 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -5,7 +5,7 @@ * Endrelia Technologies Inc. * * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c deleted file mode 100644 index 684781e4088..00000000000 --- a/sound/soc/at91/eti_b1_wm8731.c +++ /dev/null @@ -1,349 +0,0 @@ -/* - * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 29, 2006 - * - * Based on corgi.c by: - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> - * Richard Purdie <richard@openedhand.com> - * - * 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. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/hardware.h> -#include <mach/gpio.h> - -#include "../codecs/wm8731.h" -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) -#else -#define DBG(x...) -#endif - -static struct clk *pck1_clk; -static struct clk *pllb_clk; - - -static int eti_b1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* cpu clock is the AT91 master clock sent to the SSC */ - ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, - 60000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Start PCK1 clock. */ - clk_enable(pck1_clk); - DBG("pck1 started\n"); - - return 0; -} - -static void eti_b1_shutdown(struct snd_pcm_substream *substream) -{ - /* Stop PCK1 clock. */ - clk_disable(pck1_clk); - DBG("pck1 stopped\n"); -} - -static int eti_b1_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - unsigned int rate; - int cmr_div, period; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ - break; - case 32000: - cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ - period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ - break; - case 48000: - cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ - period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_RCMR_PERIOD, period); - } - if (ret < 0) - return ret; - -#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - /* - * Codec in Master Mode. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - -#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - - return 0; -} - -static struct snd_soc_ops eti_b1_ops = { - .startup = eti_b1_startup, - .hw_params = eti_b1_hw_params, - .shutdown = eti_b1_shutdown, -}; - - -static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route intercon[] = { - - /* speaker connected to LHPOUT */ - {"Ext Spk", NULL, "LHPOUT"}, - - /* mic is connected to Mic Jack, with WM8731 Mic Bias */ - {"MICIN", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Int Mic"}, -}; - -/* - * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. - */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -{ - DBG("eti_b1_wm8731_init() called\n"); - - /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, - ARRAY_SIZE(eti_b1_dapm_widgets)); - - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); - - /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - - /* always connected */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link eti_b1_dai = { - .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai = &at91_ssc_dai[1], - .codec_dai = &wm8731_dai, - .init = eti_b1_wm8731_init, - .ops = &eti_b1_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1 = { - .name = "ETI_B1_WM8731", - .dai_link = &eti_b1_dai, - .num_links = 1, -}; - -static struct wm8731_setup_data eti_b1_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - -static struct snd_soc_device eti_b1_snd_devdata = { - .machine = &snd_soc_machine_eti_b1, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &eti_b1_wm8731_setup, -}; - -static struct platform_device *eti_b1_snd_device; - -static int __init eti_b1_init(void) -{ - int ret; - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) { - DBG("SSC1 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K); - if (!ssc->base) { - DBG("SSC1 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC1; - - eti_b1_snd_device = platform_device_alloc("soc-audio", -1); - if (!eti_b1_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); - eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; - - ret = platform_device_add(eti_b1_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_snd_device); - goto fail_io_unmap; - } - - at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ - at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ - at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ - at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ -/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ - at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ - - /* - * Set PCK1 parent to PLLB and its rate to 12 Mhz. - */ - pllb_clk = clk_get(NULL, "pllb"); - pck1_clk = clk_get(NULL, "pck1"); - - clk_set_parent(pck1_clk, pllb_clk); - clk_set_rate(pck1_clk, 12000000); - - DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); - - /* assign the GPIO pin to PCK1 */ - at91_set_B_periph(AT91_PIN_PA24, 0); - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); -#else - printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); -#endif - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); - return ret; -} - -static void __exit eti_b1_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - clk_put(pck1_clk); - clk_put(pllb_clk); - - platform_device_unregister(eti_b1_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); -} - -module_init(eti_b1_init); -module_exit(eti_b1_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index f98331d099e..dc006206f62 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602 help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_BF5XX_SOC_AD73311 + tristate "SoC AD73311 Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD73311 + help + Say Y if you want to add support for AD73311 codec on Blackfin. + +config SND_BFIN_AD73311_SE + int "PF pin for AD73311L Chip Select" + depends on SND_BF5XX_SOC_AD73311 + default 4 + help + Enter the GPIO used to control AD73311's SE pin. Acceptable + values are 0 to 7 + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN && SND_SOC diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 9ea8bd9e0ba..97bb37a6359 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o - +snd-ad73311-objs := bf5xx-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o +obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907c483..25e50d2ea1e 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { bf5xx_ac97_to_pcm( (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, @@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data) struct snd_pcm_substream *pcm = data; #if defined(CONFIG_SND_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + memset(runtime->dma_area, 0, runtime->buffer_size); snd_pcm_lib_free_pages(substream); return 0; } @@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) * SPORT working in TMD mode(include AC97). */ #if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bf5xx_mmap_copy(substream, runtime->period_size); + snd_pcm_period_elapsed(substream); + sport->tx_delay_pos = 0; sport_tx_start(sport); + } else sport_rx_start(sport); break; @@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) #if defined(CONFIG_SND_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + pr_debug("%s enter\n", __func__); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + + return 0; +} + #ifdef CONFIG_SND_MMAP_SUPPORT static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -272,6 +299,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, + .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e311fd5..5e5aafb6485 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -129,7 +129,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) struct ac97_frame *nextwrite; sport_incfrag(sport, &nextfrag, 1); - sport_incfrag(sport, &nextfrag, 1); nextwrite = (struct ac97_frame *)(sport->tx_buf + \ nextfrag * sport->tx_fragsize); diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c new file mode 100644 index 00000000000..622c9b90953 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -0,0 +1,240 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad73311.c + * Author: Cliff Cai <Cliff.Cai@analog.com> + * + * Created: Thur Sep 25 2008 + * Description: Board driver for ad73311 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad73311.h" +#include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" +#include "bf5xx-i2s.h" + +#if CONFIG_SND_BF5XX_SPORT_NUM == 0 +#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT +#else +#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT +#endif + +#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE + +static struct snd_soc_machine bf5xx_ad73311; + +static int snd_ad73311_startup(void) +{ + pr_debug("%s enter\n", __func__); + + /* Pull up SE pin on AD73311L */ + gpio_set_value(GPIO_SE, 1); + return 0; +} + +static int snd_ad73311_configure(void) +{ + unsigned short ctrl_regs[6]; + unsigned short status = 0; + int count = 0; + + /* DMCLK = MCLK = 16.384 MHz + * SCLK = DMCLK/8 = 2.048 MHz + * Sample Rate = DMCLK/2048 = 8 KHz + */ + ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ + REGB_SCDIV(0) | REGB_DIRATE(0); + ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ + REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; + ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ + REGD_IGS(2); + ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); + ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; + ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; + + local_irq_disable(); + snd_ad73311_startup(); + udelay(1); + + bfin_write_SPORT_TCR1(TFSR); + bfin_write_SPORT_TCR2(0xF); + SSYNC(); + + /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to + * FIFO before enable SPORT to transfer the data + */ + for (count = 0; count < 6; count++) + bfin_write_SPORT_TX16(ctrl_regs[count]); + SSYNC(); + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); + SSYNC(); + + /* When TUVF is set, the data is already send out */ + while (!(status & TUVF) && count++ < 10000) { + udelay(1); + status = bfin_read_SPORT_STAT(); + SSYNC(); + } + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); + SSYNC(); + local_irq_enable(); + + if (count == 10000) { + printk(KERN_ERR "ad73311: failed to configure codec\n"); + return -1; + } + return 0; +} + +static int bf5xx_probe(struct platform_device *pdev) +{ + int err; + if (gpio_request(GPIO_SE, "AD73311_SE")) { + printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); + return -EBUSY; + } + + gpio_direction_output(GPIO_SE, 0); + + err = snd_ad73311_configure(); + if (err < 0) + return -EFAULT; + + return 0; +} + +static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + pr_debug("%s enter\n", __func__); + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + pr_debug("%s rate %d format %x\n", __func__, params_rate(params), + params_format(params)); + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + + +static struct snd_soc_ops bf5xx_ad73311_ops = { + .startup = bf5xx_ad73311_startup, + .hw_params = bf5xx_ad73311_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad73311_dai = { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai = &bf5xx_i2s_dai, + .codec_dai = &ad73311_dai, + .ops = &bf5xx_ad73311_ops, +}; + +static struct snd_soc_machine bf5xx_ad73311 = { + .name = "bf5xx_ad73311", + .probe = bf5xx_probe, + .dai_link = &bf5xx_ad73311_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad73311_snd_devdata = { + .machine = &bf5xx_ad73311, + .platform = &bf5xx_i2s_soc_platform, + .codec_dev = &soc_codec_dev_ad73311, +}; + +static struct platform_device *bf52x_ad73311_snd_device; + +static int __init bf5xx_ad73311_init(void) +{ + int ret; + + pr_debug("%s enter\n", __func__); + bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf52x_ad73311_snd_device) + return -ENOMEM; + + platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; + ret = platform_device_add(bf52x_ad73311_snd_device); + + if (ret) + platform_device_put(bf52x_ad73311_snd_device); + + return ret; +} + +static void __exit bf5xx_ad73311_exit(void) +{ + pr_debug("%s enter\n", __func__); + platform_device_unregister(bf52x_ad73311_snd_device); +} + +module_init(bf5xx_ad73311_init); +module_exit(bf5xx_ad73311_exit); + +/* Module information */ +MODULE_AUTHOR("Cliff Cai"); +MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 43a4092eeb8..e020c160ee4 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,6 +70,25 @@ static struct sport_param sport_params[2] = { } }; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; + static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -78,28 +97,34 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + bf5xx_i2s.tcr1 |= TFSR | TCKFE; + bf5xx_i2s.rcr1 |= RFSR | RCKFE; + bf5xx_i2s.tcr2 |= TSFSE; + bf5xx_i2s.rcr2 |= RSFSE; + break; + case SND_SOC_DAIFMT_DSP_A: + bf5xx_i2s.tcr1 |= TFSR; + bf5xx_i2s.rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } @@ -127,14 +152,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; + sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bf5xx_i2s.tcr2 |= 23; bf5xx_i2s.rcr2 |= 23; + sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: bf5xx_i2s.tcr2 |= 31; bf5xx_i2s.rcr2 |= 31; + sport_handle->wdsize = 4; break; } @@ -145,17 +173,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * need to configure both of them at the time when the first * stream is opened. * - * CPU DAI format:I2S, slave mode. + * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, RFSR | RCKFE, - RSFSE|bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, - TSFSE|bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -174,13 +202,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) static int bf5xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, - }; - pr_debug("%s enter\n", __func__); if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); @@ -198,6 +219,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("%s enter\n", __func__); + peripheral_free_list(&sport_req[sport_num][0]); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct platform_device *dev, struct snd_soc_dai *dai) @@ -263,15 +291,16 @@ struct snd_soc_dai bf5xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, + .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c163454bbf..fcadcc081f7 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -123,6 +123,8 @@ struct sport_device { int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0b9869df0f..38a0e3b620a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS depends on I2C select SPI select SPI_MASTER + select SND_SOC_AD73311 select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1980 tristate +config SND_SOC_AD73311 + tristate + config SND_SOC_AK4535 tristate @@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" - depends on SND_SOC && SPI + tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE + depends on SPI config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f977978a340..90f0a585fc7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,8 +1,10 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o +snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(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_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96ca7bc..bd1ebdc6c86 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f2c06..1397b8e06c0 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c new file mode 100644 index 00000000000..37af8607b00 --- /dev/null +++ b/sound/soc/codecs/ad73311.c @@ -0,0 +1,107 @@ +/* + * ad73311.c -- ALSA Soc AD73311 codec support + * + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * 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. + * + * Revision history + * 25th Sep 2008 Initial version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad73311.h" + +struct snd_soc_dai ad73311_dai = { + .name = "AD73311", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +EXPORT_SYMBOL_GPL(ad73311_dai); + +static int ad73311_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "AD73311"; + codec->owner = THIS_MODULE; + codec->dai = &ad73311_dai; + codec->num_dai = 1; + socdev->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int ad73311_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad73311 = { + .probe = ad73311_soc_probe, + .remove = ad73311_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); + +MODULE_DESCRIPTION("ASoC ad73311 driver"); +MODULE_AUTHOR("Cliff Cai "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h new file mode 100644 index 00000000000..507ce0c30ed --- /dev/null +++ b/sound/soc/codecs/ad73311.h @@ -0,0 +1,90 @@ +/* + * File: sound/soc/codec/ad73311.h + * Based on: + * Author: Cliff Cai <cliff.cai@analog.com> + * + * Created: Thur Sep 25, 2008 + * Description: definitions for AD73311 registers + * + * + * Modified: + * Copyright 2006 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD73311_H__ +#define __AD73311_H__ + +#define AD_CONTROL 0x8000 +#define AD_DATA 0x0000 +#define AD_READ 0x4000 +#define AD_WRITE 0x0000 + +/* Control register A */ +#define CTRL_REG_A (0 << 8) + +#define REGA_MODE_PRO 0x00 +#define REGA_MODE_DATA 0x01 +#define REGA_MODE_MIXED 0x03 +#define REGA_DLB 0x04 +#define REGA_SLB 0x08 +#define REGA_DEVC(x) ((x & 0x7) << 4) +#define REGA_RESET 0x80 + +/* Control register B */ +#define CTRL_REG_B (1 << 8) + +#define REGB_DIRATE(x) (x & 0x3) +#define REGB_SCDIV(x) ((x & 0x3) << 2) +#define REGB_MCDIV(x) ((x & 0x7) << 4) +#define REGB_CEE (1 << 7) + +/* Control register C */ +#define CTRL_REG_C (2 << 8) + +#define REGC_PUDEV (1 << 0) +#define REGC_PUADC (1 << 3) +#define REGC_PUDAC (1 << 4) +#define REGC_PUREF (1 << 5) +#define REGC_REFUSE (1 << 6) + +/* Control register D */ +#define CTRL_REG_D (3 << 8) + +#define REGD_IGS(x) (x & 0x7) +#define REGD_RMOD (1 << 3) +#define REGD_OGS(x) ((x & 0x7) << 4) +#define REGD_MUTE (x << 7) + +/* Control register E */ +#define CTRL_REG_E (4 << 8) + +#define REGE_DA(x) (x & 0x1f) +#define REGE_IBYP (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 8) + +#define REGF_SEEN (1 << 5) +#define REGF_INV (1 << 6) +#define REGF_ALB (1 << 7) + +extern struct snd_soc_dai ad73311_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad73311; +#endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 088cf992772..2a89b5888e1 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -28,7 +28,6 @@ #include "ak4535.h" -#define AUDIO_NAME "ak4535" #define AK4535_VERSION "0.3" struct snd_soc_codec_device soc_codec_dev_ak4535; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 940ce1c3522..44ef0dacd56 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -42,7 +42,6 @@ #include "ssm2602.h" -#define AUDIO_NAME "ssm2602" #define SSM2602_VERSION "0.1" struct snd_soc_codec_device soc_codec_dev_ssm2602; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 00000000000..44308dac9e1 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,714 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * 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. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AIC23_VERSION "0.1" + +struct tlv320aic23_srate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data[2]; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data[0] = (reg << 1) | (value >> 8 & 0x01); + data[1] = value & 0xff; + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); + +static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val, reg; + + val = (ucontrol->value.integer.value[0] & 0x07); + + /* linear conversion to userspace + * 000 = -6db + * 001 = -9db + * 010 = -12db + * 011 = -18db (Min) + * 100 = 0db (Max) + */ + val = (val >= 4) ? 4 : (3 - val); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); + tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + + return 0; +} + +static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val; + + val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = val >> 6; + val = (val >= 4) ? 4 : (3 - val); + ucontrol->value.integer.value[0] = val; + return 0; + +} + +#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ + .put = snd_soc_tlv320aic23_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, + 6, 4, 0, sidetone_vol_tlv), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((srate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(srate_reg_info))) { + count++; + } + + data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | + (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | + TLV320AIC23_USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_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; + + switch (freq) { + case 12000000: + return 0; + } + return -EINVAL; +} + +static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); + tlv320aic23_write(codec, TLV320AIC23_LINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + (TLV320AIC23_LRS_ENABLED)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + TLV320AIC23_LRS_ENABLED); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); + tlv320aic23_write(codec, TLV320AIC23_ANLG, + (reg) & (~TLV320AIC23_BYPASS_ON) & + (~TLV320AIC23_MICM_MUTED)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 00000000000..79d1faf8e57 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * 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 _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 566a427c928..cff276ee261 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -48,7 +48,6 @@ #include "tlv320aic3x.h" -#define AUDIO_NAME "aic3x" #define AIC3X_VERSION "0.2" /* codec private data */ @@ -864,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: @@ -991,7 +994,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) struct snd_soc_dai aic3x_dai = { - .name = "aic3x", + .name = "tlv320aic3x", .playback = { .stream_name = "Playback", .channels_min = 1, @@ -1055,7 +1058,7 @@ static int aic3x_init(struct snd_soc_device *socdev) struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; - codec->name = "aic3x"; + codec->name = "tlv320aic3x"; codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index d206d7f892b..a69ee72a7af 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -36,7 +36,6 @@ #include "uda1380.h" #define UDA1380_VERSION "0.6" -#define AUDIO_NAME "uda1380" /* * uda1380 register cache diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d95ed..d8ca2da8d63 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,7 +28,6 @@ #include "wm8510.h" -#define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" struct snd_soc_codec_device soc_codec_dev_wm8510; @@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0001, }; +#define WM8510_POWER1_BIASEN 0x08 +#define WM8510_POWER1_BUFIOEN 0x10 + /* * read wm8510 register cache */ @@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, - &wm8510_micpga_controls[0], - ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, &wm8510_boost_controls[0], ARRAY_SIZE(wm8510_boost_controls)), @@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: - wm8510_write(codec, WM8510_POWER1, 0x1ff); - wm8510_write(codec, WM8510_POWER2, 0x1ff); - wm8510_write(codec, WM8510_POWER3, 0x1ff); - break; case SND_SOC_BIAS_PREPARE: + power1 |= 0x1; /* VMID 50k */ + wm8510_write(codec, WM8510_POWER1, power1); + break; + case SND_SOC_BIAS_STANDBY: + power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + wm8510_write(codec, WM8510_POWER1, power1); break; + case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - wm8510_write(codec, WM8510_POWER1, 0x0); - wm8510_write(codec, WM8510_POWER2, 0x0); - wm8510_write(codec, WM8510_POWER3, 0x0); + wm8510_write(codec, WM8510_POWER1, 0); + wm8510_write(codec, WM8510_POWER2, 0); + wm8510_write(codec, WM8510_POWER3, 0); break; } + codec->bias_level = level; return 0; } @@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev) } /* power on device */ + codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); @@ -747,6 +763,62 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8510_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8510_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8510\n"); + + return ret; +} + +static int __devexit wm8510_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8510_spi_driver = { + .driver = { + .name = "wm8510", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8510_spi_probe, + .remove = __devexit_p(wm8510_spi_remove), +}; + +static int wm8510_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + static int wm8510_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8510_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8510_spi_write; + ret = spi_register_driver(&wm8510_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) @@ -796,6 +874,9 @@ static int wm8510_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8510_spi_driver); +#endif kfree(codec); return 0; diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c5368396045..bdefcf5c69f 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe305b..627ebfb4209 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -36,7 +35,6 @@ #include "wm8580.h" -#define AUDIO_NAME "wm8580" #define WM8580_VERSION "0.1" struct pll_state { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7b64d9a7ff7..7f8a7e36b33 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,7 +29,6 @@ #include "wm8731.h" -#define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" struct snd_soc_codec_device soc_codec_dev_wm8731; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4892e398a59..9b7296ee5b0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,7 +29,6 @@ #include "wm8750.h" -#define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" /* codec private data */ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44f334..d426eaa2218 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -40,6 +39,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -51,7 +51,6 @@ #include "wm8753.h" -#define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" static int caps_charge = 2000; @@ -1719,6 +1718,63 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8753_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8753_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8753\n"); + + return ret; +} + +static int __devexit wm8753_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; + +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif + + static int wm8753_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8753_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8753_spi_write; + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) { @@ -1798,6 +1860,9 @@ static int wm8753_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8753_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); +#endif kfree(codec->private_data); kfree(codec); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde069f1..f55704ce931 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -79,6 +78,7 @@ #define WM8753_ADCTL2 0x3f struct wm8753_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d38b48..3b326c9b558 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3f54ec4226..ce40d787760 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0, 0, 31, 0), SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1, 6, 1, 0), SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 974a4cd0f3f..f41a578ddd4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,7 +29,6 @@ #include "wm8971.h" -#define AUDIO_NAME "wm8971" #define WM8971_VERSION "0.9" #define WM8971_REG_COUNT 43 diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 63410d7b5ef..572d22b0880 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,7 +30,6 @@ #include "wm8990.h" -#define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" /* codec private data */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b1d55..ffb471e420e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d0580db1..945b32ed988 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -141,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), -SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), SOC_ENUM("ALC Function", wm9713_enum[6]), SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 86923299bc1..94a02eaa482 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -277,7 +277,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; u16 imr; u8 psc_cmd; - long flags; + unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_i2s->capture; @@ -699,9 +699,11 @@ static ssize_t psc_i2s_stat_store(struct device *dev, return count; } -DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); -DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); +static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); +static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -819,8 +821,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, /* Register the SYSFS files */ rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_i2s->dev, "error creating sysfs files\n"); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e70043..8b7766b998d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on osk5912. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58075e..e09d1f297f6 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-osk5912-objs := osk5912.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d166b6b2a60..fae3ad36e0b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); - snd_soc_dapm_disable_pin(codec, "HPLCOM"); - snd_soc_dapm_disable_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); + snd_soc_dapm_nc_pin(codec, "HPLCOM"); + snd_soc_dapm_nc_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e16d7f..8485a8a9d0f 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -43,6 +43,7 @@ struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; /* * Flags indicating is the bus already activated and configured by * another substream @@ -59,12 +60,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = { -{ - { .name = "I2S PCM Stereo out", }, - { .name = "I2S PCM Stereo in", }, -}, -}; +static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) static const int omap1_dma_reqs[][2] = { @@ -84,11 +80,22 @@ static const unsigned long omap1_mcbsp_port[][2] = { static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2420) -static const int omap2420_dma_reqs[][2] = { + +#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX) +static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) + { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, + { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, + { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, +#endif }; +#else +static const int omap24xx_dma_reqs[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, @@ -96,10 +103,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, }; #else -static const int omap2420_dma_reqs[][2] = {}; static const unsigned long omap2420_mcbsp_port[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP2430) +static const unsigned long omap2430_mcbsp_port[][2] = { + { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap2430_mcbsp_port[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP34XX) +static const unsigned long omap34xx_mcbsp_port[][2] = { + { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap34xx_mcbsp_port[][2] = {}; +#endif + static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -161,20 +201,26 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + int wlen; unsigned long port; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; port = omap1_mcbsp_port[bus_id][substream->stream]; } else if (cpu_is_omap2420()) { - dma = omap2420_dma_reqs[bus_id][substream->stream]; + dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap2420_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap2430()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap2430_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap343x()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap34xx_mcbsp_port[bus_id][substream->stream]; } else { - /* - * TODO: Add support for 2430 and 3430 - */ return -ENODEV; } + omap_mcbsp_dai_dma_params[id][substream->stream].name = + substream->stream ? "Audio Capture" : "Audio Playback"; omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; @@ -200,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ + wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); - /* Set FS period and length in terms of bit clock periods */ - regs->srgr2 |= FPER(16 * 2 - 1); - regs->srgr1 |= FWID(16 - 1); break; default: /* Unsupported PCM format */ return -EINVAL; } + /* Set FS period and length in terms of bit clock periods */ + switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen - 1); + break; + case SND_SOC_DAIFMT_DSP_A: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen * 2 - 2); + break; + } + omap_mcbsp_config(bus_id, &mcbsp_data->regs); mcbsp_data->configured = 1; @@ -232,6 +288,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, if (mcbsp_data->configured) return 0; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -245,6 +302,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; @@ -310,7 +372,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, int clk_id) { int sel_bit; - u16 reg; + u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1; if (cpu_class_is_omap1()) { /* OMAP1's can use only external source clock */ @@ -320,6 +382,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } + if (cpu_is_omap2420() && mcbsp_data->bus_id > 1) + return -EINVAL; + + if (cpu_is_omap343x()) + reg_devconf1 = OMAP343X_CONTROL_DEVCONF1; + switch (mcbsp_data->bus_id) { case 0: reg = OMAP2_CONTROL_DEVCONF0; @@ -329,20 +397,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, reg = OMAP2_CONTROL_DEVCONF0; sel_bit = 6; break; - /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */ + case 2: + reg = reg_devconf1; + sel_bit = 0; + break; + case 3: + reg = reg_devconf1; + sel_bit = 2; + break; + case 4: + reg = reg_devconf1; + sel_bit = 4; + break; default: return -EINVAL; } - if (cpu_class_is_omap2()) { - if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) { - omap_ctrl_writel(omap_ctrl_readl(reg) & - ~(1 << sel_bit), reg); - } else { - omap_ctrl_writel(omap_ctrl_readl(reg) | - (1 << sel_bit), reg); - } - } + if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) + omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg); + else + omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg); return 0; } @@ -376,37 +450,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } -struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { -{ - .name = "omap-mcbsp-dai", - .id = 0, - .type = SND_SOC_DAI_I2S, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = { - .startup = omap_mcbsp_dai_startup, - .shutdown = omap_mcbsp_dai_shutdown, - .trigger = omap_mcbsp_dai_trigger, - .hw_params = omap_mcbsp_dai_hw_params, - }, - .dai_ops = { - .set_fmt = omap_mcbsp_dai_set_dai_fmt, - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, - }, - .private_data = &mcbsp_data[0].bus_id, -}, +#define OMAP_MCBSP_DAI_BUILDER(link_id) \ +{ \ + .name = "omap-mcbsp-dai-(link_id)", \ + .id = (link_id), \ + .type = SND_SOC_DAI_I2S, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .ops = { \ + .startup = omap_mcbsp_dai_startup, \ + .shutdown = omap_mcbsp_dai_shutdown, \ + .trigger = omap_mcbsp_dai_trigger, \ + .hw_params = omap_mcbsp_dai_hw_params, \ + }, \ + .dai_ops = { \ + .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ + }, \ + .private_data = &mcbsp_data[(link_id)].bus_id, \ +} + +struct snd_soc_dai omap_mcbsp_dai[] = { + OMAP_MCBSP_DAI_BUILDER(0), + OMAP_MCBSP_DAI_BUILDER(1), +#if NUM_LINKS >= 3 + OMAP_MCBSP_DAI_BUILDER(2), +#endif +#if NUM_LINKS == 5 + OMAP_MCBSP_DAI_BUILDER(3), + OMAP_MCBSP_DAI_BUILDER(4), +#endif }; + EXPORT_SYMBOL_GPL(omap_mcbsp_dai); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ed8afb55067..df7ad13ba73 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -38,11 +38,17 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -/* - * REVISIT: Preparation for the ASoC v2. Let the number of available links to - * be same than number of McBSP ports found in OMAP(s) we are compiling for. - */ -#define NUM_LINKS 1 +#if defined(CONFIG_ARCH_OMAP2420) +#define NUM_LINKS 2 +#endif +#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) +#undef NUM_LINKS +#define NUM_LINKS 3 +#endif +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#undef NUM_LINKS +#define NUM_LINKS 5 +#endif extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfeaec4a..e9084fdd208 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!cpu_is_omap1510()) { + if (!err & !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c new file mode 100644 index 00000000000..0fe73379689 --- /dev/null +++ b/sound/soc/omap/osk5912.c @@ -0,0 +1,232 @@ +/* + * osk5912.c -- SoC audio for OSK 5912 + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks@mistralsolutions.com> + * + * 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. + * + * 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 St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "OSK5912", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC OSK 5912"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 72b7a5140bf..2718eaf7895 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -18,13 +18,13 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/hardware/scoop.h> #include <mach/pxa-regs.h> #include <mach/hardware.h> #include <mach/corgi.h> @@ -54,8 +54,8 @@ static void corgi_ext_control(struct snd_soc_codec *codec) switch (corgi_jack_func) { case CORGI_HP: /* set = unmute headphone */ - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 1); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headphone Jack"); @@ -63,24 +63,24 @@ static void corgi_ext_control(struct snd_soc_codec *codec) break; case CORGI_MIC: /* reset = mute headphone */ - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 0); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 0); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_enable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); @@ -114,8 +114,8 @@ static int corgi_shutdown(struct snd_pcm_substream *substream) struct snd_soc_codec *codec = rtd->socdev->codec; /* set = unmute headphone */ - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 1); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); return 0; } @@ -218,22 +218,14 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol, static int corgi_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); - else - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); - + gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); return 0; } static int corgi_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); - else - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); - + gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -289,8 +281,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index d9c3f7b28be..e6ff6929ab4 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -9,7 +9,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index f84f7d8db09..4d9930c5278 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a80ae074b09..a7a3a9c5c6f 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -49,7 +49,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRTXPCDR, + .drcmr = &DRCMR(12), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -57,7 +57,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { .name = "AC97 PCM Stereo in", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRRXPCDR, + .drcmr = &DRCMR(11), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -65,7 +65,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { .name = "AC97 Aux PCM (Slot 5) Mono out", .dev_addr = __PREG(MODR), - .drcmr = &DRCMRTXMODR, + .drcmr = &DRCMR(10), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -73,7 +73,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { .name = "AC97 Aux PCM (Slot 5) Mono in", .dev_addr = __PREG(MODR), - .drcmr = &DRCMRRXMODR, + .drcmr = &DRCMR(9), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -81,7 +81,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { .name = "AC97 Mic PCM (Slot 6) Mono in", .dev_addr = __PREG(MCDR), - .drcmr = &DRCMRRXMCDR, + .drcmr = &DRCMR(8), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 39d19212f6d..e758034db5c 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -3,7 +3,7 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * lrg@slimlogic.co.uk * * 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 @@ -39,6 +39,45 @@ struct pxa2xx_gpio { u32 frm; }; +/* + * I2S Controller Register and Bit Definitions + */ +#define SACR0 __REG(0x40400000) /* Global Control Register */ +#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */ +#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */ +#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */ +#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */ +#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */ +#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */ + +#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */ +#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */ +#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */ +#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ +#define SACR0_ENB (1 << 0) /* Enable I2S Link */ +#define SACR1_ENLBF (1 << 5) /* Enable Loopback */ +#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ +#define SACR1_DREC (1 << 3) /* Disable Recording Function */ +#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */ + +#define SASR0_I2SOFF (1 << 7) /* Controller Status */ +#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */ +#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */ +#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */ +#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */ +#define SASR0_BSY (1 << 2) /* I2S Busy */ +#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */ +#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ + +#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */ +#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */ + +#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */ +#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */ +#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */ +#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */ struct pxa_i2s_port { u32 sadiv; @@ -54,7 +93,7 @@ static struct clk *clk_i2s; static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { .name = "I2S PCM Stereo out", .dev_addr = __PREG(SADR), - .drcmr = &DRCMRTXSADR, + .drcmr = &DRCMR(3), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -62,7 +101,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { .name = "I2S PCM Stereo in", .dev_addr = __PREG(SADR), - .drcmr = &DRCMRRXSADR, + .drcmr = &DRCMR(2), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -366,6 +405,6 @@ module_init(pxa2xx_i2s_init); module_exit(pxa2xx_i2s_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 3d4738c06e7..d307b6757e9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -19,16 +19,15 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/hardware/scoop.h> #include <mach/pxa-regs.h> #include <mach/hardware.h> -#include <mach/akita.h> #include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-pcm.h" @@ -63,8 +62,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 1); + gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -72,8 +71,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -81,8 +80,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_enable_pin(codec, "Line Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -90,8 +89,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headset Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_HP_OFF: @@ -100,8 +99,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; } snd_soc_dapm_sync(codec); @@ -215,23 +214,14 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (machine_is_borzoi() || machine_is_spitz()) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&spitzscoop2_device.dev, - SPITZ_SCP2_MIC_BIAS); - else - reset_scoop_gpio(&spitzscoop2_device.dev, - SPITZ_SCP2_MIC_BIAS); - } + if (machine_is_borzoi() || machine_is_spitz()) + gpio_set_value(SPITZ_GPIO_MIC_BIAS, + SND_SOC_DAPM_EVENT_ON(event)); + + if (machine_is_akita()) + gpio_set_value(AKITA_GPIO_MIC_BIAS, + SND_SOC_DAPM_EVENT_ON(event)); - if (machine_is_akita()) { - if (SND_SOC_DAPM_EVENT_ON(event)) - akita_set_ioexp(&akitaioexp_device.dev, - AKITA_IOEXP_MIC_BIAS); - else - akita_reset_ioexp(&akitaioexp_device.dev, - AKITA_IOEXP_MIC_BIAS); - } return 0; } @@ -291,13 +281,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_disable_pin(codec, "RINPUT1"); - snd_soc_dapm_disable_pin(codec, "LINPUT2"); - snd_soc_dapm_disable_pin(codec, "RINPUT2"); - snd_soc_dapm_disable_pin(codec, "LINPUT3"); - snd_soc_dapm_disable_pin(codec, "RINPUT3"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2baaa750f12..afefe41b8c4 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONOOUT"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 73a50e93a9a..87ddfefcc2f 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -511,21 +511,20 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_disable_pin(codec, "LOUT2"); - snd_soc_dapm_disable_pin(codec, "ROUT2"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "OUT4"); - snd_soc_dapm_disable_pin(codec, "LINE1"); - snd_soc_dapm_disable_pin(codec, "LINE2"); - - - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_nc_pin(codec, "LOUT2"); + snd_soc_dapm_nc_pin(codec, "ROUT2"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); /* Add neo1973 specific widgets */ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, ARRAY_SIZE(wm8753_dapm_widgets)); + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { err = snd_ctl_add(codec->card, @@ -603,6 +602,8 @@ static int lm4857_i2c_probe(struct i2c_client *client, { DBG("Entered %s\n", __func__); + i2c = client; + lm4857_write_regs(); return 0; } @@ -611,6 +612,8 @@ static int lm4857_i2c_remove(struct i2c_client *client) { DBG("Entered %s\n", __func__); + i2c = NULL; + return 0; } @@ -650,7 +653,7 @@ static void lm4857_shutdown(struct i2c_client *dev) } static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 } + { "neo1973_lm4857", 0 }, { } }; @@ -668,48 +671,6 @@ static struct i2c_driver lm4857_i2c_driver = { }; static struct platform_device *neo1973_snd_device; -static struct i2c_client *lm4857_client; - -static int __init neo1973_add_lm4857_device(struct platform_device *pdev, - int i2c_bus, - unsigned short i2c_address) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add lm4857 driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = i2c_address; - strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - lm4857_client = client; - return 0; - -err_driver: - i2c_del_driver(&lm4857_i2c_driver); - return -ENODEV; -} static int __init neo1973_init(void) { @@ -736,8 +697,8 @@ static int __init neo1973_init(void) return ret; } - ret = neo1973_add_lm4857_device(neo1973_snd_device, - neo1973_wm8753_setup, 0x7C); + ret = i2c_add_driver(&lm4857_i2c_driver); + if (ret != 0) platform_device_unregister(neo1973_snd_device); @@ -748,7 +709,6 @@ static void __exit neo1973_exit(void) { DBG("Entered %s\n", __func__); - i2c_unregister_device(lm4857_client); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad381138fc2..16c7453f494 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4,8 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * with code, comments and ideas from :- * Richard Purdie <richard@openedhand.com> * @@ -96,8 +95,8 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.parent = NULL; codec->ac97->dev.release = soc_ac97_device_release; - snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - codec->card->number, 0, codec->name); + dev_set_name(&codec->ac97->dev, "%d-%d:%s", + codec->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); @@ -1463,7 +1462,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - unsigned int shift = mc->min; + unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; if (max == 1) @@ -1886,7 +1885,7 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9ca9c08610f..7351db9606e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2,8 +2,7 @@ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -832,7 +831,7 @@ int snd_soc_dapm_sys_add(struct device *dev) return ret; asoc_debugfs = debugfs_create_dir("asoc", NULL); - if (!IS_ERR(asoc_debugfs)) + if (!IS_ERR(asoc_debugfs) && asoc_debugfs) debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, &pop_time); else @@ -1484,6 +1483,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** + * snd_soc_dapm_nc_pin - permanently disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); + +/** * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec * @pin: audio signal pin endpoint (or start point) @@ -1521,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); diff --git a/sound/sound_core.c b/sound/sound_core.c index 4ae07e236b3..a75b289a5d7 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -57,7 +57,7 @@ module_exit(cleanup_soundcore); /* * OSS sound core handling. Breaks out sound functions to submodules * - * Author: Alan Cox <alan.cox@linux.org> + * Author: Alan Cox <alan@lxorguk.ukuu.org.uk> * * Fixes: * @@ -220,9 +220,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati else sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP); - device_create_drvdata(sound_class, dev, - MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, s->name+6); + device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), + NULL, s->name+6); return r; fail: diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 49acee0c484..f87933e4881 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1,6 +1,6 @@ /* * Driver for AMD7930 sound chips found on Sparcs. - * Copyright (C) 2002 David S. Miller <davem@redhat.com> + * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net> * * Based entirely upon drivers/sbus/audio/amd7930.c which is: * Copyright (C) 1996,1997 Thomas K. Dyas (tdyas@eden.rutgers.edu) @@ -35,6 +35,8 @@ #include <linux/init.h> #include <linux/interrupt.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -44,7 +46,6 @@ #include <asm/io.h> #include <asm/irq.h> -#include <asm/sbus.h> #include <asm/prom.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ @@ -335,8 +336,8 @@ struct snd_amd7930 { int pgain; int mgain; + struct of_device *op; unsigned int irq; - unsigned int regs_size; struct snd_amd7930 *next; }; @@ -905,13 +906,16 @@ static int __devinit snd_amd7930_mixer(struct snd_amd7930 *amd) static int snd_amd7930_free(struct snd_amd7930 *amd) { + struct of_device *op = amd->op; + amd7930_idle(amd); if (amd->irq) free_irq(amd->irq, amd); if (amd->regs) - sbus_iounmap(amd->regs, amd->regs_size); + of_iounmap(&op->resource[0], amd->regs, + resource_size(&op->resource[0])); kfree(amd); @@ -930,13 +934,12 @@ static struct snd_device_ops snd_amd7930_dev_ops = { }; static int __devinit snd_amd7930_create(struct snd_card *card, - struct resource *rp, - unsigned int reg_size, + struct of_device *op, int irq, int dev, struct snd_amd7930 **ramd) { - unsigned long flags; struct snd_amd7930 *amd; + unsigned long flags; int err; *ramd = NULL; @@ -946,9 +949,10 @@ static int __devinit snd_amd7930_create(struct snd_card *card, spin_lock_init(&amd->lock); amd->card = card; - amd->regs_size = reg_size; + amd->op = op; - amd->regs = sbus_ioremap(rp, 0, amd->regs_size, "amd7930"); + amd->regs = of_ioremap(&op->resource[0], 0, + resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { snd_printk("amd7930-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -997,12 +1001,15 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_attach_common(struct resource *rp, int irq) +static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_device_id *match) { + struct resource *rp = &op->resource[0]; static int dev_num; struct snd_card *card; struct snd_amd7930 *amd; - int err; + int err, irq; + + irq = op->irqs[0]; if (dev_num >= SNDRV_CARDS) return -ENODEV; @@ -1023,8 +1030,7 @@ static int __devinit amd7930_attach_common(struct resource *rp, int irq) (unsigned long long)rp->start, irq); - if ((err = snd_amd7930_create(card, rp, - (rp->end - rp->start) + 1, + if ((err = snd_amd7930_create(card, op, irq, dev_num, &amd)) < 0) goto out_err; @@ -1049,43 +1055,7 @@ out_err: return err; } -static int __devinit amd7930_obio_attach(struct device_node *dp) -{ - const struct linux_prom_registers *regs; - const struct linux_prom_irqs *irqp; - struct resource res, *rp; - int len; - - irqp = of_get_property(dp, "intr", &len); - if (!irqp) { - snd_printk("%s: Firmware node lacks IRQ property.\n", - dp->full_name); - return -ENODEV; - } - - regs = of_get_property(dp, "reg", &len); - if (!regs) { - snd_printk("%s: Firmware node lacks register property.\n", - dp->full_name); - return -ENODEV; - } - - rp = &res; - rp->start = regs->phys_addr; - rp->end = rp->start + regs->reg_size - 1; - rp->flags = IORESOURCE_IO | (regs->which_io & 0xff); - - return amd7930_attach_common(rp, irqp->pri); -} - -static int __devinit amd7930_sbus_probe(struct of_device *dev, const struct of_device_id *match) -{ - struct sbus_dev *sdev = to_sbus_device(&dev->dev); - - return amd7930_attach_common(&sdev->resource[0], sdev->irqs[0]); -} - -static struct of_device_id amd7930_match[] = { +static const struct of_device_id amd7930_match[] = { { .name = "audio", }, @@ -1100,20 +1070,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - struct device_node *dp; - - /* Try to find the sun4c "audio" node first. */ - dp = of_find_node_by_path("/"); - dp = dp->child; - while (dp) { - if (!strcmp(dp->name, "audio")) - amd7930_obio_attach(dp); - - dp = dp->sibling; - } - - /* Probe each SBUS for amd7930 chips. */ - return of_register_driver(&amd7930_sbus_driver, &sbus_bus_type); + return of_register_driver(&amd7930_sbus_driver, &of_bus_type); } static void __exit amd7930_exit(void) diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 791d2fb821d..d44bf98e965 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1,6 +1,6 @@ /* * Driver for CS4231 sound chips found on Sparcs. - * Copyright (C) 2002 David S. Miller <davem@redhat.com> + * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net> * * Based entirely upon drivers/sbus/audio/cs4231.c which is: * Copyright (C) 1996, 1997, 1998 Derrick J Brashear (shadow@andrew.cmu.edu) @@ -17,7 +17,8 @@ #include <linux/moduleparam.h> #include <linux/irq.h> #include <linux/io.h> - +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,13 +30,12 @@ #ifdef CONFIG_SBUS #define SBUS_SUPPORT -#include <asm/sbus.h> #endif #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64) #define EBUS_SUPPORT #include <linux/pci.h> -#include <asm/ebus.h> +#include <asm/ebus_dma.h> #endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ @@ -70,8 +70,6 @@ struct cs4231_dma_control { int (*request)(struct cs4231_dma_control *dma_cont, dma_addr_t bus_addr, size_t len); unsigned int (*address)(struct cs4231_dma_control *dma_cont); - void (*preallocate)(struct snd_cs4231 *chip, - struct snd_pcm *pcm); #ifdef EBUS_SUPPORT struct ebus_dma_info ebus_info; #endif @@ -114,21 +112,12 @@ struct snd_cs4231 { struct mutex mce_mutex; /* mutex for mce register */ struct mutex open_mutex; /* mutex for ALSA open/close */ - union { -#ifdef SBUS_SUPPORT - struct sbus_dev *sdev; -#endif -#ifdef EBUS_SUPPORT - struct pci_dev *pdev; -#endif - } dev_u; + struct of_device *op; unsigned int irq[2]; unsigned int regs_size; struct snd_cs4231 *next; }; -static struct snd_cs4231 *cs4231_list; - /* Eventually we can use sound/isa/cs423x/cs4231_lib.c directly, but for * now.... -DaveM */ @@ -267,27 +256,19 @@ static unsigned char snd_cs4231_original_image[32] = static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr) { -#ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) return readb(reg_addr); else -#endif -#ifdef SBUS_SUPPORT return sbus_readb(reg_addr); -#endif } static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, void __iomem *reg_addr) { -#ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) return writeb(val, reg_addr); else -#endif -#ifdef SBUS_SUPPORT return sbus_writeb(val, reg_addr); -#endif } /* @@ -1258,7 +1239,9 @@ static int __init snd_cs4231_pcm(struct snd_card *card) pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; strcpy(pcm->name, "CS4231"); - chip->p_dma.preallocate(chip, pcm); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->op->dev, + 64 * 1024, 128 * 1024); chip->pcm = pcm; @@ -1627,8 +1610,7 @@ static int __init cs4231_attach_finish(struct snd_card *card) if (err < 0) goto out_err; - chip->next = cs4231_list; - cs4231_list = chip; + dev_set_drvdata(&chip->op->dev, chip); dev++; return 0; @@ -1783,24 +1765,19 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont) return sbus_readl(base->regs + base->dir + APCVA); } -static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_SBUS, - snd_dma_sbus_data(chip->dev_u.sdev), - 64 * 1024, 128 * 1024); -} - /* * Init and exit routines */ static int snd_cs4231_sbus_free(struct snd_cs4231 *chip) { + struct of_device *op = chip->op; + if (chip->irq[0]) free_irq(chip->irq[0], chip); if (chip->port) - sbus_iounmap(chip->port, chip->regs_size); + of_iounmap(&op->resource[0], chip->port, chip->regs_size); return 0; } @@ -1817,7 +1794,7 @@ static struct snd_device_ops snd_cs4231_sbus_dev_ops = { }; static int __init snd_cs4231_sbus_create(struct snd_card *card, - struct sbus_dev *sdev, + struct of_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -1828,13 +1805,13 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, spin_lock_init(&chip->p_dma.sbus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - chip->dev_u.sdev = sdev; - chip->regs_size = sdev->reg_addrs[0].reg_size; + chip->op = op; + chip->regs_size = resource_size(&op->resource[0]); memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - chip->port = sbus_ioremap(&sdev->resource[0], 0, - chip->regs_size, "cs4231"); + chip->port = of_ioremap(&op->resource[0], 0, + chip->regs_size, "cs4231"); if (!chip->port) { snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -1849,22 +1826,20 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, chip->p_dma.enable = sbus_dma_enable; chip->p_dma.request = sbus_dma_request; chip->p_dma.address = sbus_dma_addr; - chip->p_dma.preallocate = sbus_dma_preallocate; chip->c_dma.prepare = sbus_dma_prepare; chip->c_dma.enable = sbus_dma_enable; chip->c_dma.request = sbus_dma_request; chip->c_dma.address = sbus_dma_addr; - chip->c_dma.preallocate = sbus_dma_preallocate; - if (request_irq(sdev->irqs[0], snd_cs4231_sbus_interrupt, + if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt, IRQF_SHARED, "cs4231", chip)) { snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n", - dev, sdev->irqs[0]); + dev, op->irqs[0]); snd_cs4231_sbus_free(chip); return -EBUSY; } - chip->irq[0] = sdev->irqs[0]; + chip->irq[0] = op->irqs[0]; if (snd_cs4231_probe(chip) < 0) { snd_cs4231_sbus_free(chip); @@ -1881,9 +1856,9 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __init cs4231_sbus_attach(struct sbus_dev *sdev) +static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) { - struct resource *rp = &sdev->resource[0]; + struct resource *rp = &op->resource[0]; struct snd_card *card; int err; @@ -1895,9 +1870,9 @@ static int __init cs4231_sbus_attach(struct sbus_dev *sdev) card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, - sdev->irqs[0]); + op->irqs[0]); - err = snd_cs4231_sbus_create(card, sdev, dev); + err = snd_cs4231_sbus_create(card, op, dev); if (err < 0) { snd_card_free(card); return err; @@ -1950,30 +1925,25 @@ static unsigned int _ebus_dma_addr(struct cs4231_dma_control *dma_cont) return ebus_dma_addr(&dma_cont->ebus_info); } -static void _ebus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->dev_u.pdev), - 64*1024, 128*1024); -} - /* * Init and exit routines */ static int snd_cs4231_ebus_free(struct snd_cs4231 *chip) { + struct of_device *op = chip->op; + if (chip->c_dma.ebus_info.regs) { ebus_dma_unregister(&chip->c_dma.ebus_info); - iounmap(chip->c_dma.ebus_info.regs); + of_iounmap(&op->resource[2], chip->c_dma.ebus_info.regs, 0x10); } if (chip->p_dma.ebus_info.regs) { ebus_dma_unregister(&chip->p_dma.ebus_info); - iounmap(chip->p_dma.ebus_info.regs); + of_iounmap(&op->resource[1], chip->p_dma.ebus_info.regs, 0x10); } if (chip->port) - iounmap(chip->port); + of_iounmap(&op->resource[0], chip->port, 0x10); return 0; } @@ -1990,7 +1960,7 @@ static struct snd_device_ops snd_cs4231_ebus_dev_ops = { }; static int __init snd_cs4231_ebus_create(struct snd_card *card, - struct linux_ebus_device *edev, + struct of_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -2002,35 +1972,35 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); chip->flags |= CS4231_FLAG_EBUS; - chip->dev_u.pdev = edev->bus->self; + chip->op = op; memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)"); chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; chip->c_dma.ebus_info.client_cookie = chip; - chip->c_dma.ebus_info.irq = edev->irqs[0]; + chip->c_dma.ebus_info.irq = op->irqs[0]; strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; chip->p_dma.ebus_info.client_cookie = chip; - chip->p_dma.ebus_info.irq = edev->irqs[1]; + chip->p_dma.ebus_info.irq = op->irqs[1]; chip->p_dma.prepare = _ebus_dma_prepare; chip->p_dma.enable = _ebus_dma_enable; chip->p_dma.request = _ebus_dma_request; chip->p_dma.address = _ebus_dma_addr; - chip->p_dma.preallocate = _ebus_dma_preallocate; chip->c_dma.prepare = _ebus_dma_prepare; chip->c_dma.enable = _ebus_dma_enable; chip->c_dma.request = _ebus_dma_request; chip->c_dma.address = _ebus_dma_addr; - chip->c_dma.preallocate = _ebus_dma_preallocate; - chip->port = ioremap(edev->resource[0].start, 0x10); - chip->p_dma.ebus_info.regs = ioremap(edev->resource[1].start, 0x10); - chip->c_dma.ebus_info.regs = ioremap(edev->resource[2].start, 0x10); + chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231"); + chip->p_dma.ebus_info.regs = + of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma"); + chip->c_dma.ebus_info.regs = + of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma"); if (!chip->port || !chip->p_dma.ebus_info.regs || !chip->c_dma.ebus_info.regs) { snd_cs4231_ebus_free(chip); @@ -2078,7 +2048,7 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) +static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) { struct snd_card *card; int err; @@ -2089,10 +2059,10 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, - edev->resource[0].start, - edev->irqs[0]); + op->resource[0].start, + op->irqs[0]); - err = snd_cs4231_ebus_create(card, edev, dev); + err = snd_cs4231_ebus_create(card, op, dev); if (err < 0) { snd_card_free(card); return err; @@ -2102,68 +2072,57 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) } #endif -static int __init cs4231_init(void) +static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) { -#ifdef SBUS_SUPPORT - struct sbus_bus *sbus; - struct sbus_dev *sdev; -#endif #ifdef EBUS_SUPPORT - struct linux_ebus *ebus; - struct linux_ebus_device *edev; + if (!strcmp(op->node->parent->name, "ebus")) + return cs4231_ebus_probe(op, match); #endif - int found; - - found = 0; - #ifdef SBUS_SUPPORT - for_all_sbusdev(sdev, sbus) { - if (!strcmp(sdev->prom_name, "SUNW,CS4231")) { - if (cs4231_sbus_attach(sdev) == 0) - found++; - } - } + if (!strcmp(op->node->parent->name, "sbus") || + !strcmp(op->node->parent->name, "sbi")) + return cs4231_sbus_probe(op, match); #endif -#ifdef EBUS_SUPPORT - for_each_ebus(ebus) { - for_each_ebusdev(edev, ebus) { - int match = 0; - - if (!strcmp(edev->prom_node->name, "SUNW,CS4231")) { - match = 1; - } else if (!strcmp(edev->prom_node->name, "audio")) { - const char *compat; - - compat = of_get_property(edev->prom_node, - "compatible", NULL); - if (compat && !strcmp(compat, "SUNW,CS4231")) - match = 1; - } + return -ENODEV; +} - if (match && - cs4231_ebus_attach(edev) == 0) - found++; - } - } -#endif +static int __devexit cs4231_remove(struct of_device *op) +{ + struct snd_cs4231 *chip = dev_get_drvdata(&op->dev); + snd_card_free(chip->card); - return (found > 0) ? 0 : -EIO; + return 0; } -static void __exit cs4231_exit(void) -{ - struct snd_cs4231 *p = cs4231_list; +static const struct of_device_id cs4231_match[] = { + { + .name = "SUNW,CS4231", + }, + { + .name = "audio", + .compatible = "SUNW,CS4231", + }, + {}, +}; - while (p != NULL) { - struct snd_cs4231 *next = p->next; +MODULE_DEVICE_TABLE(of, cs4231_match); - snd_card_free(p->card); +static struct of_platform_driver cs4231_driver = { + .name = "audio", + .match_table = cs4231_match, + .probe = cs4231_probe, + .remove = __devexit_p(cs4231_remove), +}; - p = next; - } +static int __init cs4231_init(void) +{ + return of_register_driver(&cs4231_driver, &of_bus_type); +} - cs4231_list = NULL; +static void __exit cs4231_exit(void) +{ + of_unregister_driver(&cs4231_driver); } module_init(cs4231_init); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index c534a2a849f..23ed6f04a71 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -57,6 +57,7 @@ #include <linux/delay.h> #include <linux/irq.h> #include <linux/io.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -66,7 +67,7 @@ #include <sound/initval.h> #include <linux/of.h> -#include <asm/sbus.h> +#include <linux/of_device.h> #include <asm/atomic.h> MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets"); @@ -297,7 +298,7 @@ struct dbri_streaminfo { /* This structure holds the information for both chips (DBRI & CS4215) */ struct snd_dbri { int regs_size, irq; /* Needed for unload */ - struct sbus_dev *sdev; /* SBUS device info */ + struct of_device *op; /* OF device info */ spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ @@ -2093,14 +2094,15 @@ static int snd_dbri_hw_params(struct snd_pcm_substream *substream, */ if (info->dvma_buffer == 0) { if (DBRI_STREAMNO(substream) == DBRI_PLAY) - direction = SBUS_DMA_TODEVICE; + direction = DMA_TO_DEVICE; else - direction = SBUS_DMA_FROMDEVICE; + direction = DMA_FROM_DEVICE; - info->dvma_buffer = sbus_map_single(dbri->sdev, - runtime->dma_area, - params_buffer_bytes(hw_params), - direction); + info->dvma_buffer = + dma_map_single(&dbri->op->dev, + runtime->dma_area, + params_buffer_bytes(hw_params), + direction); } direction = params_buffer_bytes(hw_params); @@ -2121,12 +2123,12 @@ static int snd_dbri_hw_free(struct snd_pcm_substream *substream) */ if (info->dvma_buffer) { if (DBRI_STREAMNO(substream) == DBRI_PLAY) - direction = SBUS_DMA_TODEVICE; + direction = DMA_TO_DEVICE; else - direction = SBUS_DMA_FROMDEVICE; + direction = DMA_FROM_DEVICE; - sbus_unmap_single(dbri->sdev, info->dvma_buffer, - substream->runtime->buffer_size, direction); + dma_unmap_single(&dbri->op->dev, info->dvma_buffer, + substream->runtime->buffer_size, direction); info->dvma_buffer = 0; } if (info->pipe != -1) { @@ -2519,31 +2521,34 @@ static void __devinit snd_dbri_proc(struct snd_card *card) static void snd_dbri_free(struct snd_dbri *dbri); static int __devinit snd_dbri_create(struct snd_card *card, - struct sbus_dev *sdev, - int irq, int dev) + struct of_device *op, + int irq, int dev) { struct snd_dbri *dbri = card->private_data; int err; spin_lock_init(&dbri->lock); - dbri->sdev = sdev; + dbri->op = op; dbri->irq = irq; - dbri->dma = sbus_alloc_consistent(sdev, sizeof(struct dbri_dma), - &dbri->dma_dvma); + dbri->dma = dma_alloc_coherent(&op->dev, + sizeof(struct dbri_dma), + &dbri->dma_dvma, GFP_ATOMIC); + if (!dbri->dma) + return -ENOMEM; memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", dbri->dma, dbri->dma_dvma); /* Map the registers into memory. */ - dbri->regs_size = sdev->reg_addrs[0].reg_size; - dbri->regs = sbus_ioremap(&sdev->resource[0], 0, - dbri->regs_size, "DBRI Registers"); + dbri->regs_size = resource_size(&op->resource[0]); + dbri->regs = of_ioremap(&op->resource[0], 0, + dbri->regs_size, "DBRI Registers"); if (!dbri->regs) { printk(KERN_ERR "DBRI: could not allocate registers\n"); - sbus_free_consistent(sdev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&op->dev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); return -EIO; } @@ -2551,9 +2556,9 @@ static int __devinit snd_dbri_create(struct snd_card *card, "DBRI audio", dbri); if (err) { printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq); - sbus_iounmap(dbri->regs, dbri->regs_size); - sbus_free_consistent(sdev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + of_iounmap(&op->resource[0], dbri->regs, dbri->regs_size); + dma_free_coherent(&op->dev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); return err; } @@ -2577,27 +2582,23 @@ static void snd_dbri_free(struct snd_dbri *dbri) free_irq(dbri->irq, dbri); if (dbri->regs) - sbus_iounmap(dbri->regs, dbri->regs_size); + of_iounmap(&dbri->op->resource[0], dbri->regs, dbri->regs_size); if (dbri->dma) - sbus_free_consistent(dbri->sdev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&dbri->op->dev, + sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct of_device *of_dev, - const struct of_device_id *match) +static int __devinit dbri_probe(struct of_device *op, const struct of_device_id *match) { - struct sbus_dev *sdev = to_sbus_device(&of_dev->dev); struct snd_dbri *dbri; - int irq; struct resource *rp; struct snd_card *card; static int dev = 0; + int irq; int err; - dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n", - sdev->prom_name, sdev->slot); - if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2605,7 +2606,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, return -ENOENT; } - irq = sdev->irqs[0]; + irq = op->irqs[0]; if (irq <= 0) { printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev); return -ENODEV; @@ -2618,12 +2619,12 @@ static int __devinit dbri_probe(struct of_device *of_dev, strcpy(card->driver, "DBRI"); strcpy(card->shortname, "Sun DBRI"); - rp = &sdev->resource[0]; + rp = &op->resource[0]; sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d", card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, irq); - err = snd_dbri_create(card, sdev, irq, dev); + err = snd_dbri_create(card, op, irq, dev); if (err < 0) { snd_card_free(card); return err; @@ -2640,7 +2641,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, /* /proc file handling */ snd_dbri_proc(card); - dev_set_drvdata(&of_dev->dev, card); + dev_set_drvdata(&op->dev, card); err = snd_card_register(card); if (err < 0) @@ -2648,7 +2649,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n", dev, dbri->regs, - dbri->irq, sdev->prom_name[9], dbri->mm.version); + dbri->irq, op->node->name[9], dbri->mm.version); dev++; return 0; @@ -2659,19 +2660,19 @@ _err: return err; } -static int __devexit dbri_remove(struct of_device *dev) +static int __devexit dbri_remove(struct of_device *op) { - struct snd_card *card = dev_get_drvdata(&dev->dev); + struct snd_card *card = dev_get_drvdata(&op->dev); snd_dbri_free(card->private_data); snd_card_free(card); - dev_set_drvdata(&dev->dev, NULL); + dev_set_drvdata(&op->dev, NULL); return 0; } -static struct of_device_id dbri_match[] = { +static const struct of_device_id dbri_match[] = { { .name = "SUNW,DBRIe", }, @@ -2693,7 +2694,7 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_driver(&dbri_sbus_driver, &sbus_bus_type); + return of_register_driver(&dbri_sbus_driver, &of_bus_type); } static void __exit dbri_exit(void) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 69689e79bf7..92115755d98 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1480,6 +1480,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Advanced modes of the Edirol UA-25EX. + * For the standard mode, UA-25EX has ID 0582:00e7, which + * offers only 16-bit PCM at 44.1 kHz and no MIDI. + */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-25EX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index b441fe2cd19..c2515b680f9 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -118,12 +118,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, void *vaddr; struct us122l *us122l = area->vm_private_data; struct usb_stream *s; - int vm_f = VM_FAULT_SIGBUS; mutex_lock(&us122l->mutex); s = us122l->sk.s; if (!s) - goto out; + goto unlock; offset = vmf->pgoff << PAGE_SHIFT; if (offset < PAGE_ALIGN(s->read_size)) @@ -131,7 +130,7 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, else { offset -= PAGE_ALIGN(s->read_size); if (offset >= PAGE_ALIGN(s->write_size)) - goto out; + goto unlock; vaddr = us122l->sk.write_page + offset; } @@ -141,9 +140,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, mutex_unlock(&us122l->mutex); vmf->page = page; - vm_f = 0; -out: - return vm_f; + + return 0; +unlock: + mutex_unlock(&us122l->mutex); + return VM_FAULT_SIGBUS; } static void usb_stream_hwdep_vm_close(struct vm_area_struct *area) |