diff options
Diffstat (limited to 'sound')
426 files changed, 23305 insertions, 11791 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index aa5d803..1ca8dc2 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1076,8 +1076,6 @@ static int aaci_remove(struct amba_device *dev) { struct snd_card *card = amba_get_drvdata(dev); - amba_set_drvdata(dev, NULL); - if (card) { struct aaci *aaci = card->private_data; writel(0, aaci->base + AACI_MAINCR); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ec54be4..5066a37 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -14,12 +14,14 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/dmaengine.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include <mach/regs-ac97.h> #include <mach/audio.h> @@ -41,20 +43,20 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_reset, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { - .name = "AC97 PCM out", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(12), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_out_req = 12; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_out_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = { - .name = "AC97 PCM in", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(11), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_in_req = 11; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; @@ -230,7 +232,6 @@ static int pxa2xx_ac97_remove(struct platform_device *dev) if (card) { snd_card_free(card); - platform_set_drvdata(dev, NULL); pxa2xx_ac97_hw_remove(dev); } diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 76e0d56..a61d7a9 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -7,11 +7,13 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/dma-mapping.h> +#include <linux/dmaengine.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include <mach/dma.h> @@ -43,6 +45,35 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, size_t period = params_period_bytes(params); pxa_dma_desc *dma_desc; dma_addr_t dma_buff_phys, next_desc_phys; + u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + + /* temporary transition hack */ + switch (rtd->params->addr_width) { + case DMA_SLAVE_BUSWIDTH_1_BYTE: + dcmd |= DCMD_WIDTH1; + break; + case DMA_SLAVE_BUSWIDTH_2_BYTES: + dcmd |= DCMD_WIDTH2; + break; + case DMA_SLAVE_BUSWIDTH_4_BYTES: + dcmd |= DCMD_WIDTH4; + break; + default: + /* can't happen */ + break; + } + + switch (rtd->params->maxburst) { + case 8: + dcmd |= DCMD_BURST8; + break; + case 16: + dcmd |= DCMD_BURST16; + break; + case 32: + dcmd |= DCMD_BURST32; + break; + } snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = totsize; @@ -55,14 +86,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, dma_desc->ddadr = next_desc_phys; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dma_desc->dsadr = dma_buff_phys; - dma_desc->dtadr = rtd->params->dev_addr; + dma_desc->dtadr = rtd->params->addr; } else { - dma_desc->dsadr = rtd->params->dev_addr; + dma_desc->dsadr = rtd->params->addr; dma_desc->dtadr = dma_buff_phys; } if (period > totsize) period = totsize; - dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN; + dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN; dma_desc++; dma_buff_phys += period; } while (totsize -= period); @@ -76,8 +107,10 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - if (rtd && rtd->params && rtd->params->drcmr) - *rtd->params->drcmr = 0; + if (rtd && rtd->params && rtd->params->filter_data) { + unsigned long req = *(unsigned long *) rtd->params->filter_data; + DRCMR(req) = 0; + } snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -136,6 +169,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_pointer); int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; + unsigned long req; if (!prtd || !prtd->params) return 0; @@ -146,7 +180,8 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) DCSR(prtd->dma_ch) &= ~DCSR_RUN; DCSR(prtd->dma_ch) = 0; DCMD(prtd->dma_ch) = 0; - *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD; + req = *(unsigned long *) prtd->params->filter_data; + DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD; return 0; } @@ -155,7 +190,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_prepare); void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) { struct snd_pcm_substream *substream = dev_id; - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; int dcsr; dcsr = DCSR(dma_ch); @@ -164,9 +198,11 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) if (dcsr & DCSR_ENDINTR) { snd_pcm_period_elapsed(substream); } else { - printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", - rtd->params->name, dma_ch, dcsr); + printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", + dma_ch, dcsr); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 26422a3..69a2455 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -11,8 +11,11 @@ */ #include <linux/module.h> +#include <linux/dmaengine.h> + #include <sound/core.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include "pxa2xx-pcm.h" @@ -40,7 +43,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? client->playback_params : client->capture_params; - ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW, + ret = pxa_request_dma("dma", DMA_PRIO_LOW, pxa2xx_pcm_dma_irq, substream); if (ret < 0) goto err2; diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 65f86b5..2a8fc08 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -13,14 +13,14 @@ struct pxa2xx_runtime_data { int dma_ch; - struct pxa2xx_pcm_dma_params *params; + struct snd_dmaengine_dai_dma_data *params; pxa_dma_desc *dma_desc_array; dma_addr_t dma_desc_array_phys; }; struct pxa2xx_pcm_client { - struct pxa2xx_pcm_dma_params *playback_params; - struct pxa2xx_pcm_dma_params *capture_params; + struct snd_dmaengine_dai_dma_data *playback_params; + struct snd_dmaengine_dai_dma_data *capture_params; int (*startup)(struct snd_pcm_substream *); void (*shutdown)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); diff --git a/sound/core/Kconfig b/sound/core/Kconfig index b413ed0..313f22e 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -6,6 +6,9 @@ config SND_PCM tristate select SND_TIMER +config SND_DMAENGINE_PCM + tristate + config SND_HWDEP tristate @@ -157,6 +160,15 @@ config SND_DYNAMIC_MINORS If you are unsure about this, say N here. +config SND_MAX_CARDS + int "Max number of sound cards" + range 4 256 + default 32 + depends on SND_DYNAMIC_MINORS + help + Specify the max number of sound cards that can be assigned + on a single machine. + config SND_SUPPORT_OLD_API bool "Support old ALSA API" default y diff --git a/sound/core/Makefile b/sound/core/Makefile index 43d4117..5e890cf 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,6 +13,8 @@ snd-$(CONFIG_SND_JACK) += jack.o snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o +snd-pcm-dmaengine-objs := pcm_dmaengine.o + snd-page-alloc-y := memalloc.o snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o @@ -30,6 +32,7 @@ obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o +obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o obj-$(CONFIG_SND_OSSEMUL) += oss/ diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): diff --git a/sound/core/init.c b/sound/core/init.c index 6ef0640..6b90871 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -46,7 +46,8 @@ static LIST_HEAD(shutdown_files); static const struct file_operations snd_shutdown_f_ops; -static unsigned int snd_cards_lock; /* locked for registering/using */ +/* locked for registering/using */ +static DECLARE_BITMAP(snd_cards_lock, SNDRV_CARDS); struct snd_card *snd_cards[SNDRV_CARDS]; EXPORT_SYMBOL(snd_cards); @@ -167,29 +168,35 @@ int snd_card_create(int idx, const char *xid, err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) + for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { /* idx == -1 == 0xffff means: take any free slot */ - if (~snd_cards_lock & idx & 1<<idx2) { + if (idx2 < sizeof(int) && !(idx & (1U << idx2))) + continue; + if (!test_bit(idx2, snd_cards_lock)) { if (module_slot_match(module, idx2)) { idx = idx2; break; } } + } } if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) + for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { /* idx == -1 == 0xffff means: take any free slot */ - if (~snd_cards_lock & idx & 1<<idx2) { + if (idx2 < sizeof(int) && !(idx & (1U << idx2))) + continue; + if (!test_bit(idx2, snd_cards_lock)) { if (!slots[idx2] || !*slots[idx2]) { idx = idx2; break; } } + } } if (idx < 0) err = -ENODEV; else if (idx < snd_ecards_limit) { - if (snd_cards_lock & (1 << idx)) + if (test_bit(idx, snd_cards_lock)) err = -EBUSY; /* invalid */ } else if (idx >= SNDRV_CARDS) err = -ENODEV; @@ -199,7 +206,7 @@ int snd_card_create(int idx, const char *xid, idx, snd_ecards_limit - 1, err); goto __error; } - snd_cards_lock |= 1 << idx; /* lock it */ + set_bit(idx, snd_cards_lock); /* lock it */ if (idx >= snd_ecards_limit) snd_ecards_limit = idx + 1; /* increase the limit */ mutex_unlock(&snd_card_mutex); @@ -249,7 +256,7 @@ int snd_card_locked(int card) int locked; mutex_lock(&snd_card_mutex); - locked = snd_cards_lock & (1 << card); + locked = test_bit(card, snd_cards_lock); mutex_unlock(&snd_card_mutex); return locked; } @@ -361,7 +368,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 1: disable fops (user space) operations for ALSA API */ mutex_lock(&snd_card_mutex); snd_cards[card->number] = NULL; - snd_cards_lock &= ~(1 << card->number); + clear_bit(card->number, snd_cards_lock); mutex_unlock(&snd_card_mutex); /* phase 2: replace file->f_op with special dummy operations */ @@ -549,7 +556,6 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, const char *nid) { int len, loops; - bool with_suffix; bool is_default = false; char *id; @@ -565,26 +571,23 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, is_default = true; } - with_suffix = false; + len = strlen(id); for (loops = 0; loops < SNDRV_CARDS; loops++) { + char *spos; + char sfxstr[5]; /* "_012" */ + int sfxlen; + if (card_id_ok(card, id)) return; /* OK */ - len = strlen(id); - if (!with_suffix) { - /* add the "_X" suffix */ - char *spos = id + len; - if (len > sizeof(card->id) - 3) - spos = id + sizeof(card->id) - 3; - strcpy(spos, "_1"); - with_suffix = true; - } else { - /* modify the existing suffix */ - if (id[len - 1] != '9') - id[len - 1]++; - else - id[len - 1] = 'A'; - } + /* Add _XYZ suffix */ + sprintf(sfxstr, "_%X", loops + 1); + sfxlen = strlen(sfxstr); + if (len + sfxlen >= sizeof(card->id)) + spos = id + sizeof(card->id) - sfxlen - 1; + else + spos = id + len; + strcpy(spos, sfxstr); } /* fallback to the default id */ if (!is_default) { diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/core/pcm_dmaengine.c index aa924d9..aa924d9 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/core/pcm_dmaengine.c diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 41b3dfe..6e03b46 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ xrun_log_show(substream); \ - if (printk_ratelimit()) { \ + if (snd_printd_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ dump_stack_on_xrun(substream); \ @@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return -EPIPE; } if (pos >= runtime->buffer_size) { - if (printk_ratelimit()) { + if (snd_printd_ratelimit()) { char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); @@ -568,7 +568,8 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) * * Sets the given PCM operators to the pcm instance. */ -void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, struct snd_pcm_ops *ops) +void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, + const struct snd_pcm_ops *ops) { struct snd_pcm_str *stream = &pcm->streams[direction]; struct snd_pcm_substream *substream; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f928181..a68d4c6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1589,29 +1589,16 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream) } -/* WARNING: Don't forget to fput back the file */ -static struct file *snd_pcm_file_fd(int fd, int *fput_needed) +static bool is_pcm_file(struct file *file) { - struct file *file; - struct inode *inode; + struct inode *inode = file_inode(file); unsigned int minor; - file = fget_light(fd, fput_needed); - if (!file) - return NULL; - inode = file_inode(file); - if (!S_ISCHR(inode->i_mode) || - imajor(inode) != snd_major) { - fput_light(file, *fput_needed); - return NULL; - } + if (!S_ISCHR(inode->i_mode) || imajor(inode) != snd_major) + return false; minor = iminor(inode); - if (!snd_lookup_minor_data(minor, SNDRV_DEVICE_TYPE_PCM_PLAYBACK) && - !snd_lookup_minor_data(minor, SNDRV_DEVICE_TYPE_PCM_CAPTURE)) { - fput_light(file, *fput_needed); - return NULL; - } - return file; + return snd_lookup_minor_data(minor, SNDRV_DEVICE_TYPE_PCM_PLAYBACK) || + snd_lookup_minor_data(minor, SNDRV_DEVICE_TYPE_PCM_CAPTURE); } /* @@ -1620,16 +1607,18 @@ static struct file *snd_pcm_file_fd(int fd, int *fput_needed) static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) { int res = 0; - struct file *file; struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream1; struct snd_pcm_group *group; - int fput_needed; + struct fd f = fdget(fd); - file = snd_pcm_file_fd(fd, &fput_needed); - if (!file) + if (!f.file) return -EBADFD; - pcm_file = file->private_data; + if (!is_pcm_file(f.file)) { + res = -EBADFD; + goto _badf; + } + pcm_file = f.file->private_data; substream1 = pcm_file->substream; group = kmalloc(sizeof(*group), GFP_KERNEL); if (!group) { @@ -1663,8 +1652,9 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) up_write(&snd_pcm_link_rwsem); _nolock: snd_card_unref(substream1->pcm->card); - fput_light(file, fput_needed); kfree(group); + _badf: + fdput(f); return res; } diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index e3cb46f..b3f39b5 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -31,6 +31,7 @@ #include <linux/export.h> #include <linux/moduleparam.h> #include <linux/slab.h> +#include <linux/workqueue.h> /* * common variables @@ -60,6 +61,14 @@ static void free_devinfo(void *private); #define call_ctl(type,rec) snd_seq_kernel_client_ctl(system_client, type, rec) +/* call snd_seq_oss_midi_lookup_ports() asynchronously */ +static void async_call_lookup_ports(struct work_struct *work) +{ + snd_seq_oss_midi_lookup_ports(system_client); +} + +static DECLARE_WORK(async_lookup_work, async_call_lookup_ports); + /* * create sequencer client for OSS sequencer */ @@ -85,9 +94,6 @@ snd_seq_oss_create_client(void) system_client = rc; debug_printk(("new client = %d\n", rc)); - /* look up midi devices */ - snd_seq_oss_midi_lookup_ports(system_client); - /* create annoucement receiver port */ memset(port, 0, sizeof(*port)); strcpy(port->name, "Receiver"); @@ -115,6 +121,9 @@ snd_seq_oss_create_client(void) } rc = 0; + /* look up midi devices */ + schedule_work(&async_lookup_work); + __error: kfree(port); return rc; @@ -160,6 +169,7 @@ receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic int snd_seq_oss_delete_client(void) { + cancel_work_sync(&async_lookup_work); if (system_client >= 0) snd_seq_delete_kernel_client(system_client); diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 677dc84..862d8489 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -72,7 +72,7 @@ static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, * look up the existing ports * this looks a very exhausting job. */ -int __init +int snd_seq_oss_midi_lookup_ports(int client) { struct snd_seq_client_info *clinfo; diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 02f90b4..842a97d 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -310,20 +310,10 @@ static int master_get(struct snd_kcontrol *kcontrol, return 0; } -static int master_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int sync_slaves(struct link_master *master, int old_val, int new_val) { - struct link_master *master = snd_kcontrol_chip(kcontrol); struct link_slave *slave; struct snd_ctl_elem_value *uval; - int err, old_val; - - err = master_init(master); - if (err < 0) - return err; - old_val = master->val; - if (ucontrol->value.integer.value[0] == old_val) - return 0; uval = kmalloc(sizeof(*uval), GFP_KERNEL); if (!uval) @@ -332,11 +322,33 @@ static int master_put(struct snd_kcontrol *kcontrol, master->val = old_val; uval->id = slave->slave.id; slave_get_val(slave, uval); - master->val = ucontrol->value.integer.value[0]; + master->val = new_val; slave_put_val(slave, uval); } kfree(uval); - if (master->hook && !err) + return 0; +} + +static int master_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + int err, new_val, old_val; + bool first_init; + + err = master_init(master); + if (err < 0) + return err; + first_init = err; + old_val = master->val; + new_val = ucontrol->value.integer.value[0]; + if (new_val == old_val) + return 0; + + err = sync_slaves(master, old_val, new_val); + if (err < 0) + return err; + if (master->hook && !first_init) master->hook(master->hook_private_data, master->val); return 1; } @@ -442,20 +454,33 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook); /** - * snd_ctl_sync_vmaster_hook - Sync the vmaster hook + * snd_ctl_sync_vmaster - Sync the vmaster slaves and hook * @kcontrol: vmaster kctl element + * @hook_only: sync only the hook * - * Call the hook function to synchronize with the current value of the given - * vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't - * exist. + * Forcibly call the put callback of each slave and call the hook function + * to synchronize with the current value of the given vmaster element. + * NOP when NULL is passed to @kcontrol. */ -void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol) +void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only) { struct link_master *master; + bool first_init = false; + if (!kcontrol) return; master = snd_kcontrol_chip(kcontrol); - if (master->hook) + if (!hook_only) { + int err = master_init(master); + if (err < 0) + return; + first_init = err; + err = sync_slaves(master, master->val, master->val); + if (err < 0) + return; + } + + if (master->hook && !first_init) master->hook(master->hook_private_data, master->val); } -EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook); +EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 6f78de9..f758992 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1183,7 +1183,6 @@ static int loopback_probe(struct platform_device *devptr) static int loopback_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index fd798f7..915b4d7 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry, if (i >= ARRAY_SIZE(fields)) continue; snd_info_get_str(item, ptr, sizeof(item)); - if (strict_strtoull(item, 0, &val)) + if (kstrtoull(item, 0, &val)) continue; if (fields[i].size == sizeof(int)) *get_dummy_int_ptr(dummy, fields[i].offset) = val; @@ -1129,7 +1129,6 @@ static int snd_dummy_probe(struct platform_device *devptr) static int snd_dummy_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 8125a7e..95ea4a1 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1325,7 +1325,6 @@ static int snd_ml403_ac97cr_probe(struct platform_device *pfdev) static int snd_ml403_ac97cr_remove(struct platform_device *pfdev) { snd_card_free(platform_get_drvdata(pfdev)); - platform_set_drvdata(pfdev, NULL); return 0; } diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index da1a29b..90a3a7b 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -129,7 +129,6 @@ static int snd_mpu401_probe(struct platform_device *devptr) static int snd_mpu401_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 9f1815b..e5ec7eb 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -749,7 +749,6 @@ static int snd_mtpav_probe(struct platform_device *dev) static int snd_mtpav_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 7a5fdb9..1c19cd7 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -189,7 +189,6 @@ static int pcsp_remove(struct platform_device *dev) struct snd_pcsp *chip = platform_get_drvdata(dev); alsa_card_pcsp_exit(chip); pcspkr_input_remove(chip->input_dev); - platform_set_drvdata(dev, NULL); return 0; } diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 7425dd8..e0bf5e7 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -985,7 +985,6 @@ static int snd_serial_probe(struct platform_device *devptr) static int snd_serial_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index cc4be88..ace3879 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -132,7 +132,6 @@ static int snd_virmidi_probe(struct platform_device *devptr) static int snd_virmidi_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index c39961c..8359689 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -205,7 +205,7 @@ static int vx_read_status(struct vx_core *chip, struct vx_rmh *rmh) if (size < 1) return 0; - if (snd_BUG_ON(size > SIZE_MAX_STATUS)) + if (snd_BUG_ON(size >= SIZE_MAX_STATUS)) return -EINVAL; for (i = 1; i <= size; i++) { diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index b680c5e..f6103d6 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -3,7 +3,6 @@ #include <linux/interrupt.h> #include <linux/mutex.h> -#include <linux/spinlock.h> #include "packets-buffer.h" /** diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index d428ffe..58a5afe 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -626,9 +626,9 @@ static u64 get_unit_base(struct fw_unit *unit) return 0; } -static int isight_probe(struct device *unit_dev) +static int isight_probe(struct fw_unit *unit, + const struct ieee1394_device_id *id) { - struct fw_unit *unit = fw_unit(unit_dev); struct fw_device *fw_dev = fw_parent_device(unit); struct snd_card *card; struct isight *isight; @@ -637,7 +637,7 @@ static int isight_probe(struct device *unit_dev) err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*isight), &card); if (err < 0) return err; - snd_card_set_dev(card, unit_dev); + snd_card_set_dev(card, &unit->device); isight = card->private_data; isight->card = card; @@ -674,7 +674,7 @@ static int isight_probe(struct device *unit_dev) if (err < 0) goto error; - dev_set_drvdata(unit_dev, isight); + dev_set_drvdata(&unit->device, isight); return 0; @@ -686,23 +686,6 @@ error: return err; } -static int isight_remove(struct device *dev) -{ - struct isight *isight = dev_get_drvdata(dev); - - isight_pcm_abort(isight); - - snd_card_disconnect(isight->card); - - mutex_lock(&isight->mutex); - isight_stop_streaming(isight); - mutex_unlock(&isight->mutex); - - snd_card_free_when_closed(isight->card); - - return 0; -} - static void isight_bus_reset(struct fw_unit *unit) { struct isight *isight = dev_get_drvdata(&unit->device); @@ -716,6 +699,21 @@ static void isight_bus_reset(struct fw_unit *unit) } } +static void isight_remove(struct fw_unit *unit) +{ + struct isight *isight = dev_get_drvdata(&unit->device); + + isight_pcm_abort(isight); + + snd_card_disconnect(isight->card); + + mutex_lock(&isight->mutex); + isight_stop_streaming(isight); + mutex_unlock(&isight->mutex); + + snd_card_free_when_closed(isight->card); +} + static const struct ieee1394_device_id isight_id_table[] = { { .match_flags = IEEE1394_MATCH_SPECIFIER_ID | @@ -732,10 +730,10 @@ static struct fw_driver isight_driver = { .owner = THIS_MODULE, .name = KBUILD_MODNAME, .bus = &fw_bus_type, - .probe = isight_probe, - .remove = isight_remove, }, + .probe = isight_probe, .update = isight_bus_reset, + .remove = isight_remove, .id_table = isight_id_table, }; diff --git a/sound/firewire/scs1x.c b/sound/firewire/scs1x.c index 844a555..505fc81 100644 --- a/sound/firewire/scs1x.c +++ b/sound/firewire/scs1x.c @@ -384,9 +384,8 @@ static void scs_card_free(struct snd_card *card) kfree(scs->buffer); } -static int scs_probe(struct device *unit_dev) +static int scs_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { - struct fw_unit *unit = fw_unit(unit_dev); struct fw_device *fw_dev = fw_parent_device(unit); struct snd_card *card; struct scs *scs; @@ -395,7 +394,7 @@ static int scs_probe(struct device *unit_dev) err = snd_card_create(-16, NULL, THIS_MODULE, sizeof(*scs), &card); if (err < 0) return err; - snd_card_set_dev(card, unit_dev); + snd_card_set_dev(card, &unit->device); scs = card->private_data; scs->card = card; @@ -405,8 +404,10 @@ static int scs_probe(struct device *unit_dev) scs->output_idle = true; scs->buffer = kmalloc(HSS1394_MAX_PACKET_SIZE, GFP_KERNEL); - if (!scs->buffer) + if (!scs->buffer) { + err = -ENOMEM; goto err_card; + } scs->hss_handler.length = HSS1394_MAX_PACKET_SIZE; scs->hss_handler.address_callback = handle_hss; @@ -440,7 +441,7 @@ static int scs_probe(struct device *unit_dev) if (err < 0) goto err_card; - dev_set_drvdata(unit_dev, scs); + dev_set_drvdata(&unit->device, scs); return 0; @@ -451,9 +452,20 @@ err_card: return err; } -static int scs_remove(struct device *dev) +static void scs_update(struct fw_unit *unit) { - struct scs *scs = dev_get_drvdata(dev); + struct scs *scs = dev_get_drvdata(&unit->device); + __be64 data; + + data = cpu_to_be64(((u64)HSS1394_TAG_CHANGE_ADDRESS << 56) | + scs->hss_handler.offset); + snd_fw_transaction(scs->unit, TCODE_WRITE_BLOCK_REQUEST, + HSS1394_ADDRESS, &data, 8); +} + +static void scs_remove(struct fw_unit *unit) +{ + struct scs *scs = dev_get_drvdata(&unit->device); snd_card_disconnect(scs->card); @@ -465,19 +477,6 @@ static int scs_remove(struct device *dev) tasklet_kill(&scs->tasklet); snd_card_free_when_closed(scs->card); - - return 0; -} - -static void scs_update(struct fw_unit *unit) -{ - struct scs *scs = dev_get_drvdata(&unit->device); - __be64 data; - - data = cpu_to_be64(((u64)HSS1394_TAG_CHANGE_ADDRESS << 56) | - scs->hss_handler.offset); - snd_fw_transaction(scs->unit, TCODE_WRITE_BLOCK_REQUEST, - HSS1394_ADDRESS, &data, 8); } static const struct ieee1394_device_id scs_id_table[] = { @@ -506,10 +505,10 @@ static struct fw_driver scs_driver = { .owner = THIS_MODULE, .name = KBUILD_MODNAME, .bus = &fw_bus_type, - .probe = scs_probe, - .remove = scs_remove, }, + .probe = scs_probe, .update = scs_update, + .remove = scs_remove, .id_table = scs_id_table, }; diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index d684655..fe9e6e2 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -49,7 +49,6 @@ struct fwspk { struct snd_card *card; struct fw_unit *unit; const struct device_info *device_info; - struct snd_pcm_substream *pcm; struct mutex mutex; struct cmp_connection connection; struct amdtp_out_stream stream; @@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk) return err; pcm->private_data = fwspk; strcpy(pcm->name, fwspk->device_info->short_name); - fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - fwspk->pcm->ops = &ops; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); return 0; } @@ -663,45 +661,9 @@ static void fwspk_card_free(struct snd_card *card) mutex_destroy(&fwspk->mutex); } -static const struct device_info *fwspk_detect(struct fw_device *dev) -{ - static const struct device_info griffin_firewave = { - .driver_name = "FireWave", - .short_name = "FireWave", - .long_name = "Griffin FireWave Surround", - .pcm_constraints = firewave_constraints, - .mixer_channels = 6, - .mute_fb_id = 0x01, - .volume_fb_id = 0x02, - }; - static const struct device_info lacie_speakers = { - .driver_name = "FWSpeakers", - .short_name = "FireWire Speakers", - .long_name = "LaCie FireWire Speakers", - .pcm_constraints = lacie_speakers_constraints, - .mixer_channels = 1, - .mute_fb_id = 0x01, - .volume_fb_id = 0x01, - }; - struct fw_csr_iterator i; - int key, value; - - fw_csr_iterator_init(&i, dev->config_rom); - while (fw_csr_iterator_next(&i, &key, &value)) - if (key == CSR_VENDOR) - switch (value) { - case VENDOR_GRIFFIN: - return &griffin_firewave; - case VENDOR_LACIE: - return &lacie_speakers; - } - - return NULL; -} - -static int fwspk_probe(struct device *unit_dev) +static int fwspk_probe(struct fw_unit *unit, + const struct ieee1394_device_id *id) { - struct fw_unit *unit = fw_unit(unit_dev); struct fw_device *fw_dev = fw_parent_device(unit); struct snd_card *card; struct fwspk *fwspk; @@ -711,17 +673,13 @@ static int fwspk_probe(struct device *unit_dev) err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*fwspk), &card); if (err < 0) return err; - snd_card_set_dev(card, unit_dev); + snd_card_set_dev(card, &unit->device); fwspk = card->private_data; fwspk->card = card; mutex_init(&fwspk->mutex); fwspk->unit = fw_unit_get(unit); - fwspk->device_info = fwspk_detect(fw_dev); - if (!fwspk->device_info) { - err = -ENODEV; - goto err_unit; - } + fwspk->device_info = (const struct device_info *)id->driver_data; err = cmp_connection_init(&fwspk->connection, unit, 0); if (err < 0) @@ -756,7 +714,7 @@ static int fwspk_probe(struct device *unit_dev) if (err < 0) goto error; - dev_set_drvdata(unit_dev, fwspk); + dev_set_drvdata(&unit->device, fwspk); return 0; @@ -770,22 +728,6 @@ error: return err; } -static int fwspk_remove(struct device *dev) -{ - struct fwspk *fwspk = dev_get_drvdata(dev); - - amdtp_out_stream_pcm_abort(&fwspk->stream); - snd_card_disconnect(fwspk->card); - - mutex_lock(&fwspk->mutex); - fwspk_stop_stream(fwspk); - mutex_unlock(&fwspk->mutex); - - snd_card_free_when_closed(fwspk->card); - - return 0; -} - static void fwspk_bus_reset(struct fw_unit *unit) { struct fwspk *fwspk = dev_get_drvdata(&unit->device); @@ -803,6 +745,40 @@ static void fwspk_bus_reset(struct fw_unit *unit) amdtp_out_stream_update(&fwspk->stream); } +static void fwspk_remove(struct fw_unit *unit) +{ + struct fwspk *fwspk = dev_get_drvdata(&unit->device); + + amdtp_out_stream_pcm_abort(&fwspk->stream); + snd_card_disconnect(fwspk->card); + + mutex_lock(&fwspk->mutex); + fwspk_stop_stream(fwspk); + mutex_unlock(&fwspk->mutex); + + snd_card_free_when_closed(fwspk->card); +} + +static const struct device_info griffin_firewave = { + .driver_name = "FireWave", + .short_name = "FireWave", + .long_name = "Griffin FireWave Surround", + .pcm_constraints = firewave_constraints, + .mixer_channels = 6, + .mute_fb_id = 0x01, + .volume_fb_id = 0x02, +}; + +static const struct device_info lacie_speakers = { + .driver_name = "FWSpeakers", + .short_name = "FireWire Speakers", + .long_name = "LaCie FireWire Speakers", + .pcm_constraints = lacie_speakers_constraints, + .mixer_channels = 1, + .mute_fb_id = 0x01, + .volume_fb_id = 0x01, +}; + static const struct ieee1394_device_id fwspk_id_table[] = { { .match_flags = IEEE1394_MATCH_VENDOR_ID | @@ -813,6 +789,7 @@ static const struct ieee1394_device_id fwspk_id_table[] = { .model_id = 0x00f970, .specifier_id = SPECIFIER_1394TA, .version = VERSION_AVC, + .driver_data = (kernel_ulong_t)&griffin_firewave, }, { .match_flags = IEEE1394_MATCH_VENDOR_ID | @@ -823,6 +800,7 @@ static const struct ieee1394_device_id fwspk_id_table[] = { .model_id = 0x00f970, .specifier_id = SPECIFIER_1394TA, .version = VERSION_AVC, + .driver_data = (kernel_ulong_t)&lacie_speakers, }, { } }; @@ -833,10 +811,10 @@ static struct fw_driver fwspk_driver = { .owner = THIS_MODULE, .name = KBUILD_MODNAME, .bus = &fw_bus_type, - .probe = fwspk_probe, - .remove = fwspk_remove, }, + .probe = fwspk_probe, .update = fwspk_bus_reset, + .remove = fwspk_remove, .id_table = fwspk_id_table, }; diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index c95d8f1..5526b03 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -8,10 +8,8 @@ snd-ak4117-objs := ak4117.o snd-ak4113-objs := ak4113.o snd-ak4xxx-adda-objs := ak4xxx-adda.o snd-pt2258-objs := pt2258.o -snd-tea575x-tuner-objs := tea575x-tuner.o # Module Dependency obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4113.o snd-ak4xxx-adda.o snd-pt2258.o -obj-$(CONFIG_SND_TEA575X) += snd-tea575x-tuner.o diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index cef813d..ed726d1 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -571,7 +571,7 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); int mixer_ch = AK_GET_SHIFT(kcontrol->private_value); const char **input_names; - int num_names, idx; + unsigned int num_names, idx; num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); if (!num_names) diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c deleted file mode 100644 index 8a36a1d..0000000 --- a/sound/i2c/other/tea575x-tuner.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * ALSA driver for TEA5757/5759 Philips AM/FM radio tuner chips - * - * Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz> - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include <asm/io.h> -#include <linux/delay.h> -#include <linux/module.h> -#include <linux/init.h> -#include <linux/slab.h> -#include <linux/sched.h> -#include <media/v4l2-device.h> -#include <media/v4l2-dev.h> -#include <media/v4l2-fh.h> -#include <media/v4l2-ioctl.h> -#include <media/v4l2-event.h> -#include <sound/tea575x-tuner.h> - -MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); -MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); -MODULE_LICENSE("GPL"); - -/* - * definitions - */ - -#define TEA575X_BIT_SEARCH (1<<24) /* 1 = search action, 0 = tuned */ -#define TEA575X_BIT_UPDOWN (1<<23) /* 0 = search down, 1 = search up */ -#define TEA575X_BIT_MONO (1<<22) /* 0 = stereo, 1 = mono */ -#define TEA575X_BIT_BAND_MASK (3<<20) -#define TEA575X_BIT_BAND_FM (0<<20) -#define TEA575X_BIT_BAND_MW (1<<20) -#define TEA575X_BIT_BAND_LW (2<<20) -#define TEA575X_BIT_BAND_SW (3<<20) -#define TEA575X_BIT_PORT_0 (1<<19) /* user bit */ -#define TEA575X_BIT_PORT_1 (1<<18) /* user bit */ -#define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */ -#define TEA575X_BIT_SEARCH_5_28 (0<<16) /* FM >5uV, AM >28uV */ -#define TEA575X_BIT_SEARCH_10_40 (1<<16) /* FM >10uV, AM > 40uV */ -#define TEA575X_BIT_SEARCH_30_63 (2<<16) /* FM >30uV, AM > 63uV */ -#define TEA575X_BIT_SEARCH_150_1000 (3<<16) /* FM > 150uV, AM > 1000uV */ -#define TEA575X_BIT_DUMMY (1<<15) /* buffer */ -#define TEA575X_BIT_FREQ_MASK 0x7fff - -enum { BAND_FM, BAND_FM_JAPAN, BAND_AM }; - -static const struct v4l2_frequency_band bands[] = { - { - .type = V4L2_TUNER_RADIO, - .index = 0, - .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | - V4L2_TUNER_CAP_FREQ_BANDS, - .rangelow = 87500 * 16, - .rangehigh = 108000 * 16, - .modulation = V4L2_BAND_MODULATION_FM, - }, - { - .type = V4L2_TUNER_RADIO, - .index = 0, - .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | - V4L2_TUNER_CAP_FREQ_BANDS, - .rangelow = 76000 * 16, - .rangehigh = 91000 * 16, - .modulation = V4L2_BAND_MODULATION_FM, - }, - { - .type = V4L2_TUNER_RADIO, - .index = 1, - .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS, - .rangelow = 530 * 16, - .rangehigh = 1710 * 16, - .modulation = V4L2_BAND_MODULATION_AM, - }, -}; - -/* - * lowlevel part - */ - -static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) -{ - u16 l; - u8 data; - - if (tea->ops->write_val) - return tea->ops->write_val(tea, val); - - tea->ops->set_direction(tea, 1); - udelay(16); - - for (l = 25; l > 0; l--) { - data = (val >> 24) & TEA575X_DATA; - val <<= 1; /* shift data */ - tea->ops->set_pins(tea, data | TEA575X_WREN); - udelay(2); - tea->ops->set_pins(tea, data | TEA575X_WREN | TEA575X_CLK); - udelay(2); - tea->ops->set_pins(tea, data | TEA575X_WREN); - udelay(2); - } - - if (!tea->mute) - tea->ops->set_pins(tea, 0); -} - -static u32 snd_tea575x_read(struct snd_tea575x *tea) -{ - u16 l, rdata; - u32 data = 0; - - if (tea->ops->read_val) - return tea->ops->read_val(tea); - - tea->ops->set_direction(tea, 0); - tea->ops->set_pins(tea, 0); - udelay(16); - - for (l = 24; l--;) { - tea->ops->set_pins(tea, TEA575X_CLK); - udelay(2); - if (!l) - tea->tuned = tea->ops->get_pins(tea) & TEA575X_MOST ? 0 : 1; - tea->ops->set_pins(tea, 0); - udelay(2); - data <<= 1; /* shift data */ - rdata = tea->ops->get_pins(tea); - if (!l) - tea->stereo = (rdata & TEA575X_MOST) ? 0 : 1; - if (rdata & TEA575X_DATA) - data++; - udelay(2); - } - - if (tea->mute) - tea->ops->set_pins(tea, TEA575X_WREN); - - return data; -} - -static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val) -{ - u32 freq = val & TEA575X_BIT_FREQ_MASK; - - if (freq == 0) - return freq; - - switch (tea->band) { - case BAND_FM: - /* freq *= 12.5 */ - freq *= 125; - freq /= 10; - /* crystal fixup */ - freq -= TEA575X_FMIF; - break; - case BAND_FM_JAPAN: - /* freq *= 12.5 */ - freq *= 125; - freq /= 10; - /* crystal fixup */ - freq += TEA575X_FMIF; - break; - case BAND_AM: - /* crystal fixup */ - freq -= TEA575X_AMIF; - break; - } - - return clamp(freq * 16, bands[tea->band].rangelow, - bands[tea->band].rangehigh); /* from kHz */ -} - -static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) -{ - return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea)); -} - -void snd_tea575x_set_freq(struct snd_tea575x *tea) -{ - u32 freq = tea->freq / 16; /* to kHz */ - u32 band = 0; - - switch (tea->band) { - case BAND_FM: - band = TEA575X_BIT_BAND_FM; - /* crystal fixup */ - freq += TEA575X_FMIF; - /* freq /= 12.5 */ - freq *= 10; - freq /= 125; - break; - case BAND_FM_JAPAN: - band = TEA575X_BIT_BAND_FM; - /* crystal fixup */ - freq -= TEA575X_FMIF; - /* freq /= 12.5 */ - freq *= 10; - freq /= 125; - break; - case BAND_AM: - band = TEA575X_BIT_BAND_MW; - /* crystal fixup */ - freq += TEA575X_AMIF; - break; - } - - tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK); - tea->val |= band; - tea->val |= freq & TEA575X_BIT_FREQ_MASK; - snd_tea575x_write(tea, tea->val); - tea->freq = snd_tea575x_val_to_freq(tea, tea->val); -} - -/* - * Linux Video interface - */ - -static int vidioc_querycap(struct file *file, void *priv, - struct v4l2_capability *v) -{ - struct snd_tea575x *tea = video_drvdata(file); - - strlcpy(v->driver, tea->v4l2_dev->name, sizeof(v->driver)); - strlcpy(v->card, tea->card, sizeof(v->card)); - strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card)); - strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info)); - v->device_caps = V4L2_CAP_TUNER | V4L2_CAP_RADIO; - if (!tea->cannot_read_data) - v->device_caps |= V4L2_CAP_HW_FREQ_SEEK; - v->capabilities = v->device_caps | V4L2_CAP_DEVICE_CAPS; - return 0; -} - -static int vidioc_enum_freq_bands(struct file *file, void *priv, - struct v4l2_frequency_band *band) -{ - struct snd_tea575x *tea = video_drvdata(file); - int index; - - if (band->tuner != 0) - return -EINVAL; - - switch (band->index) { - case 0: - if (tea->tea5759) - index = BAND_FM_JAPAN; - else - index = BAND_FM; - break; - case 1: - if (tea->has_am) { - index = BAND_AM; - break; - } - /* Fall through */ - default: - return -EINVAL; - } - - *band = bands[index]; - if (!tea->cannot_read_data) - band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; - - return 0; -} - -static int vidioc_g_tuner(struct file *file, void *priv, - struct v4l2_tuner *v) -{ - struct snd_tea575x *tea = video_drvdata(file); - struct v4l2_frequency_band band_fm = { 0, }; - - if (v->index > 0) - return -EINVAL; - - snd_tea575x_read(tea); - vidioc_enum_freq_bands(file, priv, &band_fm); - - memset(v, 0, sizeof(*v)); - strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name)); - v->type = V4L2_TUNER_RADIO; - v->capability = band_fm.capability; - v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow; - v->rangehigh = band_fm.rangehigh; - v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO; - v->audmode = (tea->val & TEA575X_BIT_MONO) ? - V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO; - v->signal = tea->tuned ? 0xffff : 0; - return 0; -} - -static int vidioc_s_tuner(struct file *file, void *priv, - const struct v4l2_tuner *v) -{ - struct snd_tea575x *tea = video_drvdata(file); - u32 orig_val = tea->val; - - if (v->index) - return -EINVAL; - tea->val &= ~TEA575X_BIT_MONO; - if (v->audmode == V4L2_TUNER_MODE_MONO) - tea->val |= TEA575X_BIT_MONO; - /* Only apply changes if currently tuning FM */ - if (tea->band != BAND_AM && tea->val != orig_val) - snd_tea575x_set_freq(tea); - - return 0; -} - -static int vidioc_g_frequency(struct file *file, void *priv, - struct v4l2_frequency *f) -{ - struct snd_tea575x *tea = video_drvdata(file); - - if (f->tuner != 0) - return -EINVAL; - f->type = V4L2_TUNER_RADIO; - f->frequency = tea->freq; - return 0; -} - -static int vidioc_s_frequency(struct file *file, void *priv, - const struct v4l2_frequency *f) -{ - struct snd_tea575x *tea = video_drvdata(file); - - if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) - return -EINVAL; - - if (tea->has_am && f->frequency < (20000 * 16)) - tea->band = BAND_AM; - else if (tea->tea5759) - tea->band = BAND_FM_JAPAN; - else - tea->band = BAND_FM; - - tea->freq = clamp_t(u32, f->frequency, bands[tea->band].rangelow, - bands[tea->band].rangehigh); - snd_tea575x_set_freq(tea); - return 0; -} - -static int vidioc_s_hw_freq_seek(struct file *file, void *fh, - const struct v4l2_hw_freq_seek *a) -{ - struct snd_tea575x *tea = video_drvdata(file); - unsigned long timeout; - int i, spacing; - - if (tea->cannot_read_data) - return -ENOTTY; - if (a->tuner || a->wrap_around) - return -EINVAL; - - if (file->f_flags & O_NONBLOCK) - return -EWOULDBLOCK; - - if (a->rangelow || a->rangehigh) { - for (i = 0; i < ARRAY_SIZE(bands); i++) { - if ((i == BAND_FM && tea->tea5759) || - (i == BAND_FM_JAPAN && !tea->tea5759) || - (i == BAND_AM && !tea->has_am)) - continue; - if (bands[i].rangelow == a->rangelow && - bands[i].rangehigh == a->rangehigh) - break; - } - if (i == ARRAY_SIZE(bands)) - return -EINVAL; /* No matching band found */ - if (i != tea->band) { - tea->band = i; - tea->freq = clamp(tea->freq, bands[i].rangelow, - bands[i].rangehigh); - snd_tea575x_set_freq(tea); - } - } - - spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */ - - /* clear the frequency, HW will fill it in */ - tea->val &= ~TEA575X_BIT_FREQ_MASK; - tea->val |= TEA575X_BIT_SEARCH; - if (a->seek_upward) - tea->val |= TEA575X_BIT_UPDOWN; - else - tea->val &= ~TEA575X_BIT_UPDOWN; - snd_tea575x_write(tea, tea->val); - timeout = jiffies + msecs_to_jiffies(10000); - for (;;) { - if (time_after(jiffies, timeout)) - break; - if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { - /* some signal arrived, stop search */ - tea->val &= ~TEA575X_BIT_SEARCH; - snd_tea575x_set_freq(tea); - return -ERESTARTSYS; - } - if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { - u32 freq; - - /* Found a frequency, wait until it can be read */ - for (i = 0; i < 100; i++) { - msleep(10); - freq = snd_tea575x_get_freq(tea); - if (freq) /* available */ - break; - } - if (freq == 0) /* shouldn't happen */ - break; - /* - * if we moved by less than the spacing, or in the - * wrong direction, continue seeking - */ - if (abs(tea->freq - freq) < 16 * spacing || - (a->seek_upward && freq < tea->freq) || - (!a->seek_upward && freq > tea->freq)) { - snd_tea575x_write(tea, tea->val); - continue; - } - tea->freq = freq; - tea->val &= ~TEA575X_BIT_SEARCH; - return 0; - } - } - tea->val &= ~TEA575X_BIT_SEARCH; - snd_tea575x_set_freq(tea); - return -ENODATA; -} - -static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) -{ - struct snd_tea575x *tea = container_of(ctrl->handler, struct snd_tea575x, ctrl_handler); - - switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - tea->mute = ctrl->val; - snd_tea575x_set_freq(tea); - return 0; - } - - return -EINVAL; -} - -static const struct v4l2_file_operations tea575x_fops = { - .unlocked_ioctl = video_ioctl2, - .open = v4l2_fh_open, - .release = v4l2_fh_release, - .poll = v4l2_ctrl_poll, -}; - -static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { - .vidioc_querycap = vidioc_querycap, - .vidioc_g_tuner = vidioc_g_tuner, - .vidioc_s_tuner = vidioc_s_tuner, - .vidioc_g_frequency = vidioc_g_frequency, - .vidioc_s_frequency = vidioc_s_frequency, - .vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek, - .vidioc_enum_freq_bands = vidioc_enum_freq_bands, - .vidioc_log_status = v4l2_ctrl_log_status, - .vidioc_subscribe_event = v4l2_ctrl_subscribe_event, - .vidioc_unsubscribe_event = v4l2_event_unsubscribe, -}; - -static const struct video_device tea575x_radio = { - .ioctl_ops = &tea575x_ioctl_ops, - .release = video_device_release_empty, -}; - -static const struct v4l2_ctrl_ops tea575x_ctrl_ops = { - .s_ctrl = tea575x_s_ctrl, -}; - -/* - * initialize all the tea575x chips - */ -int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner) -{ - int retval; - - tea->mute = true; - - /* Not all devices can or know how to read the data back. - Such devices can set cannot_read_data to true. */ - if (!tea->cannot_read_data) { - snd_tea575x_write(tea, 0x55AA); - if (snd_tea575x_read(tea) != 0x55AA) - return -ENODEV; - } - - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; - tea->freq = 90500 * 16; /* 90.5Mhz default */ - snd_tea575x_set_freq(tea); - - tea->vd = tea575x_radio; - video_set_drvdata(&tea->vd, tea); - mutex_init(&tea->mutex); - strlcpy(tea->vd.name, tea->v4l2_dev->name, sizeof(tea->vd.name)); - tea->vd.lock = &tea->mutex; - tea->vd.v4l2_dev = tea->v4l2_dev; - tea->fops = tea575x_fops; - tea->fops.owner = owner; - tea->vd.fops = &tea->fops; - set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags); - /* disable hw_freq_seek if we can't use it */ - if (tea->cannot_read_data) - v4l2_disable_ioctl(&tea->vd, VIDIOC_S_HW_FREQ_SEEK); - - if (!tea->cannot_mute) { - tea->vd.ctrl_handler = &tea->ctrl_handler; - v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); - v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, - V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1); - retval = tea->ctrl_handler.error; - if (retval) { - v4l2_err(tea->v4l2_dev, "can't initialize controls\n"); - v4l2_ctrl_handler_free(&tea->ctrl_handler); - return retval; - } - - if (tea->ext_init) { - retval = tea->ext_init(tea); - if (retval) { - v4l2_ctrl_handler_free(&tea->ctrl_handler); - return retval; - } - } - - v4l2_ctrl_handler_setup(&tea->ctrl_handler); - } - - retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->radio_nr); - if (retval) { - v4l2_err(tea->v4l2_dev, "can't register video device!\n"); - v4l2_ctrl_handler_free(tea->vd.ctrl_handler); - return retval; - } - - return 0; -} - -void snd_tea575x_exit(struct snd_tea575x *tea) -{ - video_unregister_device(&tea->vd); - v4l2_ctrl_handler_free(tea->vd.ctrl_handler); -} - -static int __init alsa_tea575x_module_init(void) -{ - return 0; -} - -static void __exit alsa_tea575x_module_exit(void) -{ -} - -module_init(alsa_tea575x_module_init) -module_exit(alsa_tea575x_module_exit) - -EXPORT_SYMBOL(snd_tea575x_init); -EXPORT_SYMBOL(snd_tea575x_exit); -EXPORT_SYMBOL(snd_tea575x_set_freq); diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index c214ecf..e3f455b 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -135,7 +135,6 @@ out: snd_card_free(card); static int snd_ad1848_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index d265455..3565921 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -101,7 +101,6 @@ out: snd_card_free(card); static int snd_adlib_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index a7369fe..f84f073 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -418,7 +418,6 @@ static int snd_cmi8328_remove(struct device *pdev, unsigned int dev) snd_cmi8328_cfg_write(cmi->port, CFG2, 0); snd_cmi8328_cfg_write(cmi->port, CFG3, 0); snd_card_free(card); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index c707c52..270b965 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -651,7 +651,6 @@ static int snd_cmi8330_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index aa7a5d8..ba9a74e 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -151,7 +151,6 @@ out: snd_card_free(card); static int snd_cs4231_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 252e9fb..69614ac 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -504,7 +504,6 @@ static int snd_cs423x_isa_remove(struct device *pdev, unsigned int dev) { snd_card_free(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } @@ -600,7 +599,6 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, static void snd_cs423x_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); - pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 102874a..cdcfb57 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -213,7 +213,6 @@ out: static int snd_es1688_isa_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 24380ef..12978b8 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2235,7 +2235,6 @@ static int snd_es18xx_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } @@ -2305,7 +2304,6 @@ static int snd_audiodrive_pnp_detect(struct pnp_dev *pdev, static void snd_audiodrive_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); - pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index 672184e..81244e7 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -623,7 +623,6 @@ error: static int snd_galaxy_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 16bca4e..1adc1b9 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -215,7 +215,6 @@ out: snd_card_free(card); static int snd_gusclassic_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 0b9c242..38e1e32 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -344,7 +344,6 @@ out: snd_card_free(card); static int snd_gusextreme_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index c309a5d..652d5d8 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -357,7 +357,6 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) static int snd_gusmax_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 78bc574..afef0d7 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); #ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, - 8, iwave); + printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); #endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ @@ -849,7 +848,6 @@ static int snd_interwave_isa_probe(struct device *pdev, static int snd_interwave_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index ddabb40..81aeb93 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -1064,7 +1064,6 @@ cfg_error: static int snd_msnd_isa_remove(struct device *pdev, unsigned int dev) { snd_msnd_unload(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 075777a..cc01c41 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -757,7 +757,6 @@ static int snd_opl3sa2_pnp_detect(struct pnp_dev *pdev, static void snd_opl3sa2_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); - pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM @@ -900,7 +899,6 @@ static int snd_opl3sa2_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c3da1df..619753d 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1495,7 +1495,6 @@ static int snd_miro_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index b41ed86..6effe99 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -173,11 +173,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids); #endif /* CONFIG_PNP */ -#ifdef OPTi93X -#define DEV_NAME "opti93x" -#else -#define DEV_NAME "opti92x" -#endif +#define DEV_NAME KBUILD_MODNAME static char * snd_opti9xx_names[] = { "unknown", @@ -1035,7 +1031,6 @@ static int snd_opti9xx_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } @@ -1168,7 +1163,7 @@ static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard) static struct pnp_card_driver opti9xx_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, - .name = "opti9xx", + .name = DEV_NAME, .id_table = snd_opti9xx_pnpids, .probe = snd_opti9xx_pnp_probe, .remove = snd_opti9xx_pnp_remove, diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 4961da4..356a630 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -345,7 +345,6 @@ static int snd_jazz16_remove(struct device *devptr, unsigned int dev) { struct snd_card *card = dev_get_drvdata(devptr); - dev_set_drvdata(devptr, NULL); snd_card_free(card); return 0; } diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 50dbec4..a413099 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -566,7 +566,6 @@ static int snd_sb16_isa_probe(struct device *pdev, unsigned int dev) static int snd_sb16_isa_remove(struct device *pdev, unsigned int dev) { snd_card_free(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 237d964..a806ae9 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -208,7 +208,6 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) static int snd_sb8_remove(struct device *pdev, unsigned int dev) { snd_card_free(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 5376ebf..09d481b 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -698,7 +698,6 @@ static int snd_sc6000_remove(struct device *devptr, unsigned int dev) release_region(port[dev], 0x10); release_region(mss_port[dev], 4); - dev_set_drvdata(devptr, NULL); snd_card_free(card); return 0; } diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 42a0097..57b3389 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1200,7 +1200,6 @@ _release_card: static int snd_sscape_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index fe5dd98..82dd769 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -581,7 +581,6 @@ static int snd_wavefront_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index a59c888..461d94c 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) unsigned long flags; int err = 0, n = 0; struct dma_buffparms *dmap = adev->dmap_in; - int go; if (!(adev->open_mode & OPEN_READ)) return -EIO; @@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) spin_unlock_irqrestore(&dmap->lock,flags); return -EAGAIN; } - if ((go = adev->go)) + if (adev->go) timeout = dmabuf_timeout(dmap); spin_unlock_irqrestore(&dmap->lock,flags); diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 2a44cc1..12be1fb 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -178,7 +178,6 @@ static int probe_one(struct pci_dev *pdev, const struct pci_device_id *ent) return 0; err_out_free: - pci_set_drvdata(pdev, NULL); kfree(hw_config); return 1; } @@ -187,7 +186,6 @@ static void remove_one(struct pci_dev *pdev) { struct address_info *hw_config = pci_get_drvdata(pdev); sb_dsp_unload(hw_config, 0); - pci_set_drvdata(pdev, NULL); kfree(hw_config); } diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 7e814a5..4bbcc0f 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -149,17 +149,19 @@ #include <linux/interrupt.h> #include <linux/mutex.h> #include <linux/slab.h> +#include <linux/delay.h> #include <asm/visws/cobalt.h> #include "sound_config.h" +static DEFINE_MUTEX(vwsnd_mutex); + /*****************************************************************************/ /* debug stuff */ #ifdef VWSND_DEBUG -static DEFINE_MUTEX(vwsnd_mutex); static int shut_up = 1; /* diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 0e66ba4..67f56a2 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -902,8 +902,6 @@ snd_harmony_free(struct snd_harmony *h) if (h->iobase) iounmap(h->iobase); - parisc_set_drvdata(h->dev, NULL); - kfree(h); return 0; } @@ -1016,7 +1014,6 @@ static int snd_harmony_remove(struct parisc_device *padev) { snd_card_free(parisc_get_drvdata(padev)); - parisc_set_drvdata(padev, NULL); return 0; } diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fe6fa93..46ed9e8 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,10 +1,5 @@ # ALSA PCI drivers -config SND_TEA575X - tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO || RADIO_SHARK - default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO || RADIO_SHARK - menuconfig SND_PCI bool "PCI sound devices" depends on PCI @@ -542,7 +537,11 @@ config SND_ES1968_INPUT config SND_ES1968_RADIO bool "Enable TEA5757 radio tuner support for es1968" depends on SND_ES1968 + depends on MEDIA_RADIO_SUPPORT depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_ES1968 + select RADIO_ADAPTERS + select RADIO_TEA575X + help Say Y here to include support for TEA5757 radio tuner integrated on some MediaForte cards (e.g. SF64-PCE2). @@ -562,7 +561,10 @@ config SND_FM801 config SND_FM801_TEA575X_BOOL bool "ForteMedia FM801 + TEA5757 tuner" depends on SND_FM801 + depends on MEDIA_RADIO_SUPPORT depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_FM801 + select RADIO_ADAPTERS + select RADIO_TEA575X help Say Y here to include support for soundcards based on the ForteMedia FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index d37c683..445ca48 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1296,7 +1296,7 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, struct snd_ac97 *ac97) { int err; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; unsigned char lo_max, hi_max; if (! snd_ac97_valid_reg(ac97, reg)) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index ad8a311..d2b9d61 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1046,7 +1046,6 @@ static void snd_ad1889_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 53754f5..3dfa12b 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2298,7 +2298,6 @@ static int snd_ali_probe(struct pci_dev *pci, static void snd_ali_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver ali5451_driver = { diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 864c431..591efb6 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -282,7 +282,6 @@ static void snd_als300_remove(struct pci_dev *pci) { snd_als300_dbgcallenter(); snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); snd_als300_dbgcallleave(); } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 61efda2..ffc821b 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -984,7 +984,6 @@ out: static void snd_card_als4000_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index fbc1720..dc632cd 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,7 +769,10 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { + unsigned long flags; + snd_pcm_stream_lock_irqsave(s, flags); snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(s, flags); continue; } } else { @@ -1278,7 +1281,7 @@ struct hpi_control { u16 dst_node_type; u16 dst_node_index; u16 band; - char name[44]; /* copied to snd_ctl_elem_id.name[44]; */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* copied to snd_ctl_elem_id.name[44]; */ }; static const char * const asihpi_tuner_band_names[] = { diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index ef5019f..7f02720 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -445,7 +445,6 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) if (pa->p_buffer) vfree(pa->p_buffer); - pci_set_drvdata(pci_dev, NULL); if (1) dev_info(&pci_dev->dev, "remove %04x:%04x,%04x:%04x,%04x, HPI index %d\n", diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 6e78c67..f6dec3e 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -689,7 +689,9 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* @@ -1714,7 +1716,6 @@ static int snd_atiixp_probe(struct pci_dev *pci, static void snd_atiixp_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver atiixp_driver = { diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index d0bec7b..289563e 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,7 +638,9 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* @@ -1334,7 +1336,6 @@ static int snd_atiixp_probe(struct pci_dev *pci, static void snd_atiixp_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver atiixp_modem_driver = { diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index b157e1f..7059dd6 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -371,7 +371,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_vortex_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } // pci_driver definition diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 08e9a47..2925220 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -392,7 +392,6 @@ static int snd_aw2_probe(struct pci_dev *pci, static void snd_aw2_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* open callback */ diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 1204a0f..c8e1216 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2725,7 +2725,6 @@ snd_azf3328_remove(struct pci_dev *pci) { snd_azf3328_dbgcallenter(); snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); snd_azf3328_dbgcallleave(); } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 9febe55..1880203 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -953,7 +953,6 @@ _error: static void snd_bt87x_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* default entries for all Bt87x cards - it's not exported */ diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 1610a57..f4db558 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1896,7 +1896,6 @@ static int snd_ca0106_probe(struct pci_dev *pci, static void snd_ca0106_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index c617435..2755ec5 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3317,7 +3317,6 @@ static int snd_cmipci_probe(struct pci_dev *pci, static void snd_cmipci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 6a86950..1dc793e 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1312,7 +1312,7 @@ static int snd_cs4281_free(struct cs4281 *chip) /* Sound System Power Management - Turn Everything OFF */ snd_cs4281_pokeBA0(chip, BA0_SSPM, 0); /* PCI interface - D3 state */ - pci_set_power_state(chip->pci, 3); + pci_set_power_state(chip->pci, PCI_D3hot); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1971,7 +1971,6 @@ static int snd_cs4281_probe(struct pci_dev *pci, static void snd_cs4281_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 6b0d8b5..b034983 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -158,7 +158,6 @@ static int snd_card_cs46xx_probe(struct pci_dev *pci, static void snd_card_cs46xx_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver cs46xx_driver = { diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index dace827..c6b82c8 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -91,7 +91,6 @@ static int snd_cs5530_dev_free(struct snd_device *device) static void snd_cs5530_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static u8 snd_cs5530_mixer_read(unsigned long io, u8 reg) diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 7e4b13e..902bebd 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -391,7 +391,6 @@ static void snd_cs5535audio_remove(struct pci_dev *pci) { olpc_quirks_cleanup(); snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver cs5535audio_driver = { diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index d01ffcb..d464ad2 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -122,7 +122,6 @@ error: static void ct_card_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 760cbff..05cfe55 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2323,7 +2323,6 @@ static void snd_echo_remove(struct pci_dev *pci) chip = pci_get_drvdata(pci); if (chip) snd_card_free(chip->card); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 8c5010f..9e1bd0c 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -202,7 +202,6 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, static void snd_card_emu10k1_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index cdff11d..56ad9d6 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1623,7 +1623,6 @@ static int snd_emu10k1x_probe(struct pci_dev *pci, static void snd_emu10k1x_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } // PCI IDs diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index db2dc83..61262f3 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1842,7 +1842,7 @@ static int snd_ensoniq_create_gameport(struct ensoniq *ensoniq, int dev) default: if (!request_region(io_port, 8, "ens137x: gameport")) { - printk(KERN_WARNING "ens137x: gameport io port 0x%#x in use\n", + printk(KERN_WARNING "ens137x: gameport io port %#x in use\n", io_port); return -EBUSY; } @@ -1939,7 +1939,7 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) #endif if (ensoniq->irq >= 0) synchronize_irq(ensoniq->irq); - pci_set_power_state(ensoniq->pci, 3); + pci_set_power_state(ensoniq->pci, PCI_D3hot); __hw_end: #ifdef CHIP1370 if (ensoniq->dma_bug.area) @@ -2497,7 +2497,6 @@ static int snd_audiopci_probe(struct pci_dev *pci, static void snd_audiopci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver ens137x_driver = { diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 8423403..9213fb3 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1881,7 +1881,6 @@ static int snd_es1938_probe(struct pci_dev *pci, static void snd_es1938_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver es1938_driver = { diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..b0e3d92 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -113,7 +113,7 @@ #include <sound/initval.h> #ifdef CONFIG_SND_ES1968_RADIO -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> #endif #define CARD_NAME "ESS Maestro1/2" @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; @@ -2909,7 +2936,6 @@ static int snd_es1968_probe(struct pci_dev *pci, static void snd_es1968_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver es1968_driver = { diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4f07fda..45bc8a9 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -37,7 +37,7 @@ #include <asm/io.h> #ifdef CONFIG_SND_FM801_TEA575X_BOOL -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> #endif MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); @@ -1370,7 +1370,6 @@ static int snd_card_fm801_probe(struct pci_dev *pci, static void snd_card_fm801_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 80a7d44..8de66cc 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -140,7 +140,6 @@ config SND_HDA_CODEC_VIA config SND_HDA_CODEC_HDMI bool "Build HDMI/DisplayPort HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include HDMI and DisplayPort HD-audio codec @@ -152,6 +151,11 @@ config SND_HDA_CODEC_HDMI snd-hda-codec-hdmi. This module is automatically loaded at probing. +config SND_HDA_I915 + bool + default y + depends on DRM_I915 + config SND_HDA_CODEC_CIRRUS bool "Build Cirrus Logic codec support" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 24a2514..c091438 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,4 +1,6 @@ snd-hda-intel-objs := hda_intel.o +# for haswell power well +snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46..48a9d00 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 55108b5..5b6c4e3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -185,20 +185,19 @@ EXPORT_SYMBOL_HDA(snd_hda_get_jack_type); * Compose a 32bit command word to be sent to the HD-audio controller */ static inline unsigned int -make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, +make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { u32 val; - if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + if ((codec->addr & ~0xf) || (nid & ~0x7f) || (verb & ~0xfff) || (parm & ~0xffff)) { - printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", - codec->addr, direct, nid, verb, parm); + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x\n", + codec->addr, nid, verb, parm); return ~0; } val = (u32)codec->addr << 28; - val |= (u32)direct << 27; val |= (u32)nid << 20; val |= verb << 8; val |= parm; @@ -209,7 +208,7 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, * Send and receive a verb */ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, - unsigned int *res) + int flags, unsigned int *res) { struct hda_bus *bus = codec->bus; int err; @@ -222,6 +221,8 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, again: snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); + if (flags & HDA_RW_NO_RESPONSE_FALLBACK) + bus->no_response_fallback = 1; for (;;) { trace_hda_send_cmd(codec, cmd); err = bus->ops.command(bus, cmd); @@ -234,6 +235,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, *res = bus->ops.get_response(bus, codec->addr); trace_hda_get_response(codec, *res); } + bus->no_response_fallback = 0; mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); if (!codec_in_pm(codec) && res && *res == -1 && bus->rirb_error) { @@ -255,7 +257,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -264,12 +266,12 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * Returns the obtained response value, or -1 for an error. */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm) { - unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - if (codec_exec_verb(codec, cmd, &res)) + if (codec_exec_verb(codec, cmd, flags, &res)) return -1; return res; } @@ -279,7 +281,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -287,12 +289,12 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm) +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) { - unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - return codec_exec_verb(codec, cmd, + return codec_exec_verb(codec, cmd, flags, codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -664,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEV_LIST_LEN_MASK; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + dev_len = dev_len < max_devices ? dev_len : max_devices; + + devices = 0; + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices] = (u8)parm; + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS @@ -1214,11 +1274,13 @@ static void hda_jackpoll_work(struct work_struct *work) { struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - if (!codec->jackpoll_interval) - return; snd_hda_jack_set_dirty_all(codec); snd_hda_jack_poll_all(codec); + + if (!codec->jackpoll_interval) + return; + queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, codec->jackpoll_interval); } @@ -2523,7 +2585,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) flush_workqueue(bus->workq); #endif snd_hda_ctls_clear(codec); - /* relase PCMs */ + /* release PCMs */ for (i = 0; i < codec->num_pcms; i++) { if (codec->pcm_info[i].pcm) { snd_device_free(card, codec->pcm_info[i].pcm); @@ -3582,7 +3644,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * snd_hda_codec_write_cache - send a single command with caching * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3591,7 +3653,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { int err; struct hda_cache_head *c; @@ -3600,7 +3662,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, cache_only = codec->cached_write; if (!cache_only) { - err = snd_hda_codec_write(codec, nid, direct, verb, parm); + err = snd_hda_codec_write(codec, nid, flags, verb, parm); if (err < 0) return err; } @@ -3624,7 +3686,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * snd_hda_codec_update_cache - check cache and write the cmd only when needed * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3635,7 +3697,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { struct hda_cache_head *c; u32 key; @@ -3651,7 +3713,7 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, return 0; } mutex_unlock(&codec->bus->cmd_mutex); - return snd_hda_codec_write_cache(codec, nid, direct, verb, parm); + return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); } EXPORT_SYMBOL_HDA(snd_hda_codec_update_cache); @@ -3806,11 +3868,13 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, hda_nid_t fg = codec->afg ? codec->afg : codec->mfg; int count; unsigned int state; + int flags = 0; /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { /* transition time less than 10ms for power down */ msleep(codec->epss ? 10 : 100); + flags = HDA_RW_NO_RESPONSE_FALLBACK; } /* repeat power states setting at most 10 times*/ @@ -3819,7 +3883,7 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, codec->patch_ops.set_power_state(codec, fg, power_state); else { - snd_hda_codec_read(codec, fg, 0, + snd_hda_codec_read(codec, fg, flags, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state); @@ -4461,12 +4525,13 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign - * - * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ -static int get_empty_pcm_device(struct hda_bus *bus, int type) +static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) { /* audio device indices; not linear to keep compatibility */ + /* assigned to static slots up to dev#10; if more needed, assign + * the later slot dynamically (when CONFIG_SND_DYNAMIC_MINORS=y) + */ static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, @@ -4480,18 +4545,28 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) return -EINVAL; } - for (i = 0; audio_idx[type][i] >= 0 ; i++) + for (i = 0; audio_idx[type][i] >= 0; i++) { +#ifndef CONFIG_SND_DYNAMIC_MINORS + if (audio_idx[type][i] >= 8) + break; +#endif if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; + } +#ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ for (i = 10; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } +#endif snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); +#ifndef CONFIG_SND_DYNAMIC_MINORS + snd_printk(KERN_WARNING "Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y\n"); +#endif return -EAGAIN; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..7aa9870 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -251,6 +255,11 @@ enum { #define AC_UNSOL_RES_TAG_SHIFT 26 #define AC_UNSOL_RES_SUBTAG (0x1f<<21) #define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ #define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ #define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ #define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ @@ -352,6 +361,10 @@ enum { #define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ #define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ +/* Display pin's device list length */ +#define AC_DEV_LIST_LEN_MASK 0x3f +#define AC_MAX_DEV_LIST_LEN 64 + /* * Control Parameters */ @@ -460,6 +473,11 @@ enum { #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + /* device device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, @@ -679,6 +697,7 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ + unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ }; @@ -884,6 +903,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -930,6 +950,8 @@ enum { HDA_INPUT, HDA_OUTPUT }; +/* snd_hda_codec_read/write optional flags */ +#define HDA_RW_NO_RESPONSE_FALLBACK (1 << 0) /* * constructors @@ -945,9 +967,9 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec); * low level functions */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm); -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) @@ -969,6 +991,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); @@ -986,11 +1010,11 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, const struct hda_verb *seq); int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_codec_resume_cache(struct hda_codec *codec); /* both for cmd & amp caches */ void snd_hda_codec_flush_cache(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 4b1524a..ac41e9c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -133,12 +133,18 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "line_in_auto_switch"); if (val >= 0) spec->line_in_auto_switch = !!val; + val = snd_hda_get_bool_hint(codec, "auto_mute_via_amp"); + if (val >= 0) + spec->auto_mute_via_amp = !!val; val = snd_hda_get_bool_hint(codec, "need_dac_fix"); if (val >= 0) spec->need_dac_fix = !!val; val = snd_hda_get_bool_hint(codec, "primary_hp"); if (val >= 0) spec->no_primary_hp = !val; + val = snd_hda_get_bool_hint(codec, "multi_io"); + if (val >= 0) + spec->no_multi_io = !val; val = snd_hda_get_bool_hint(codec, "multi_cap_vol"); if (val >= 0) spec->multi_cap_vol = !!val; @@ -519,7 +525,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -621,7 +627,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -645,7 +651,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } @@ -808,6 +814,11 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) * Helper functions for creating mixer ctl elements */ +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + enum { HDA_CTL_WIDGET_VOL, HDA_CTL_WIDGET_MUTE, @@ -815,8 +826,22 @@ enum { }; static const struct snd_kcontrol_new control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), - HDA_BIND_MUTE(NULL, 0, 0, 0), + /* only the put callback is replaced for handling the special mute */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_AMP_FLAG, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = hda_gen_mixer_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_bind_switch_get, + .put = hda_gen_bind_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, }; /* add dynamic controls from template */ @@ -840,7 +865,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[32]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; @@ -922,6 +947,35 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, return add_sw_ctl(codec, pfx, cidx, chs, path); } +/* playback mute control with the software mute bit check */ +static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gen_spec *spec = codec->spec; + + if (spec->auto_mute_via_amp) { + hda_nid_t nid = get_amp_nid(kcontrol); + bool enabled = !((spec->mute_bits >> nid) & 1); + ucontrol->value.integer.value[0] &= enabled; + ucontrol->value.integer.value[1] &= enabled; + } +} + +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); + return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +} + +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); + return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol); +} + /* any ctl assigned to the path with the given index? */ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) { @@ -1510,7 +1564,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, cfg->speaker_pins, spec->multiout.extra_out_nid, spec->speaker_paths); - if (fill_mio_first && cfg->line_outs == 1 && + if (!spec->no_multi_io && + fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = fill_multi_ios(codec, cfg->line_out_pins[0], true); if (!err) @@ -1523,7 +1578,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, spec->private_dac_nids, spec->out_paths, spec->main_out_badness); - if (fill_mio_first && + if (!spec->no_multi_io && fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { /* try to fill multi-io first */ err = fill_multi_ios(codec, cfg->line_out_pins[0], false); @@ -1551,7 +1606,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, return err; badness += err; } - if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + if (!spec->no_multi_io && + cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = fill_multi_ios(codec, cfg->line_out_pins[0], false); if (err < 0) return err; @@ -1569,7 +1625,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, check_aamix_out_path(codec, spec->speaker_paths[0]); } - if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + if (!spec->no_multi_io && + cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2) spec->multi_ios = 1; /* give badness */ @@ -1900,7 +1957,7 @@ static int create_extra_outs(struct hda_codec *codec, int num_pins, for (i = 0; i < num_pins; i++) { const char *name; - char tmp[44]; + char tmp[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int err, idx = 0; if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) @@ -2453,7 +2510,7 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins, } if (get_out_jack_num_items(codec, pin) > 1) { struct snd_kcontrol_new *knew; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; get_jack_mode_name(codec, pin, name, sizeof(name)); knew = snd_hda_gen_add_kctl(spec, name, &out_jack_mode_enum); @@ -2585,7 +2642,7 @@ static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; struct snd_kcontrol_new *knew; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; unsigned int defcfg; if (pin == spec->hp_mic_pin) @@ -3285,7 +3342,7 @@ static int add_single_cap_ctl(struct hda_codec *codec, const char *label, bool inv_dmic) { struct hda_gen_spec *spec = codec->spec; - char tmpname[44]; + char tmpname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = is_switch ? HDA_CTL_WIDGET_MUTE : HDA_CTL_WIDGET_VOL; const char *sfx = is_switch ? "Switch" : "Volume"; unsigned int chs = inv_dmic ? 1 : 3; @@ -3547,7 +3604,7 @@ static int parse_mic_boost(struct hda_codec *codec) struct nid_path *path; unsigned int val; int idx; - char boost_label[44]; + char boost_label[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; idx = imux->items[i].index; if (idx >= imux->num_items) @@ -3693,7 +3750,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, /* check each pin in the given array; returns true if any of them is plugged */ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { - int i, present = 0; + int i; + bool present = false; for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; @@ -3702,14 +3760,15 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) /* don't detect pins retasked as inputs */ if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN) continue; - present |= snd_hda_jack_detect(codec, nid); + if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT) + present = true; } return present; } /* standard HP/line-out auto-mute helper */ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, - bool mute) + int *paths, bool mute) { struct hda_gen_spec *spec = codec->spec; int i; @@ -3719,6 +3778,25 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, unsigned int val, oldval; if (!nid) break; + + if (spec->auto_mute_via_amp) { + struct nid_path *path; + hda_nid_t mute_nid; + + path = snd_hda_get_path_from_idx(codec, paths[i]); + if (!path) + continue; + mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]); + if (!mute_nid) + continue; + if (mute) + spec->mute_bits |= (1ULL << mute_nid); + else + spec->mute_bits &= ~(1ULL << mute_nid); + set_pin_eapd(codec, nid, !mute); + continue; + } + oldval = snd_hda_codec_get_pin_target(codec, nid); if (oldval & PIN_IN) continue; /* no mute for inputs */ @@ -3745,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, void snd_hda_gen_update_outputs(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; + int *paths; int on; /* Control HP pins/amps depending on master_mute state; * in general, HP pins/amps control should be enabled in all cases, * but currently set only for master_mute, just to be safe */ + if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT) + paths = spec->out_paths; + else + paths = spec->hp_paths; do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), - spec->autocfg.hp_pins, spec->master_mute); + spec->autocfg.hp_pins, paths, spec->master_mute); if (!spec->automute_speaker) on = 0; @@ -3760,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec) on = spec->hp_jack_present | spec->line_jack_present; on |= spec->master_mute; spec->speaker_muted = on; + if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) + paths = spec->out_paths; + else + paths = spec->speaker_paths; do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), - spec->autocfg.speaker_pins, on); + spec->autocfg.speaker_pins, paths, on); /* toggle line-out mutes if needed, too */ /* if LO is a copy of either HP or Speaker, don't need to handle it */ @@ -3774,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec) on = spec->hp_jack_present; on |= spec->master_mute; spec->line_out_muted = on; + paths = spec->out_paths; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), - spec->autocfg.line_out_pins, on); + spec->autocfg.line_out_pins, paths, on); } EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs); @@ -3786,6 +3874,10 @@ static void call_update_outputs(struct hda_codec *codec) spec->automute_hook(codec); else snd_hda_gen_update_outputs(codec); + + /* sync the whole vmaster slaves to reflect the new auto-mute status */ + if (spec->auto_mute_via_amp && !codec->bus->shutdown) + snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false); } /* standard HP-automute helper */ @@ -3842,7 +3934,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; - if (snd_hda_jack_detect(codec, pin)) { + if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 7620031..48d4402 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -209,6 +209,7 @@ struct hda_gen_spec { unsigned int master_mute:1; /* master mute over all */ unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */ unsigned int line_in_auto_switch:1; /* allow line-in auto switch */ + unsigned int auto_mute_via_amp:1; /* auto-mute via amp instead of pinctl */ /* parser behavior flags; set before snd_hda_gen_parse_auto_config() */ unsigned int suppress_auto_mute:1; /* suppress input jack auto mute */ @@ -219,6 +220,7 @@ struct hda_gen_spec { unsigned int hp_mic:1; /* Allow HP as a mic-in */ unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */ unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */ + unsigned int no_multi_io:1; /* Don't try multi I/O config */ unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */ unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */ unsigned int own_eapd_ctl:1; /* set EAPD by own function */ @@ -237,6 +239,9 @@ struct hda_gen_spec { unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + /* additional mute flags (only effective with auto_mute_via_amp=1) */ + u64 mute_bits; + /* badness tables for output path evaluations */ const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index ce67608..fe0bda1 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev, \ struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ struct hda_codec *codec = hwdep->private_data; \ unsigned long val; \ - int err = strict_strtoul(buf, 0, &val); \ + int err = kstrtoul(buf, 0, &val); \ if (err < 0) \ return err; \ codec->type = val; \ @@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) p = snd_hda_get_hint(codec, key); if (!p) ret = -ENOENT; - else if (strict_strtoul(p, 0, &val)) + else if (kstrtoul(p, 0, &val)) ret = -EINVAL; else { *valp = val; @@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ struct hda_codec **codecp) \ { \ unsigned long val; \ - if (!strict_strtoul(buf, 0, &val)) \ + if (!kstrtoul(buf, 0, &val)) \ (*codecp)->name = val; \ } diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c new file mode 100644 index 0000000..76c13d5 --- /dev/null +++ b/sound/pci/hda/hda_i915.c @@ -0,0 +1,75 @@ +/* + * hda_i915.c - routines for Haswell HDA controller power well support + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY + * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License + * for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software Foundation, + * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <sound/core.h> +#include <drm/i915_powerwell.h> +#include "hda_i915.h" + +static void (*get_power)(void); +static void (*put_power)(void); + +void hda_display_power(bool enable) +{ + if (!get_power || !put_power) + return; + + snd_printdd("HDA display power %s \n", + enable ? "Enable" : "Disable"); + if (enable) + get_power(); + else + put_power(); +} + +int hda_i915_init(void) +{ + int err = 0; + + get_power = symbol_request(i915_request_power_well); + if (!get_power) { + snd_printk(KERN_WARNING "hda-i915: get_power symbol get fail\n"); + return -ENODEV; + } + + put_power = symbol_request(i915_release_power_well); + if (!put_power) { + symbol_put(i915_request_power_well); + get_power = NULL; + return -ENODEV; + } + + snd_printd("HDA driver get symbol successfully from i915 module\n"); + + return err; +} + +int hda_i915_exit(void) +{ + if (get_power) { + symbol_put(i915_request_power_well); + get_power = NULL; + } + if (put_power) { + symbol_put(i915_release_power_well); + put_power = NULL; + } + + return 0; +} diff --git a/sound/pci/hda/hda_i915.h b/sound/pci/hda/hda_i915.h new file mode 100644 index 0000000..5a63da2 --- /dev/null +++ b/sound/pci/hda/hda_i915.h @@ -0,0 +1,35 @@ +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ +#ifndef __SOUND_HDA_I915_H +#define __SOUND_HDA_I915_H + +#ifdef CONFIG_SND_HDA_I915 +void hda_display_power(bool enable); +int hda_i915_init(void); +int hda_i915_exit(void); +#else +static inline void hda_display_power(bool enable) {} +static inline int hda_i915_init(void) +{ + return -ENODEV; +} +static inline int hda_i915_exit(void) +{ + return 0; +} +#endif + +#endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index de18722..6e61a01 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -62,6 +62,7 @@ #include <linux/vga_switcheroo.h> #include <linux/firmware.h> #include "hda_codec.h" +#include "hda_i915.h" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; @@ -541,6 +542,10 @@ struct azx { /* for pending irqs */ struct work_struct irq_pending_work; +#ifdef CONFIG_SND_HDA_I915 + struct work_struct probe_work; +#endif + /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; @@ -550,6 +555,9 @@ struct azx { #ifdef CONFIG_SND_HDA_DSP_LOADER struct azx_dev saved_azx_dev; #endif + + /* secondary power domain for hdmi audio under vga device */ + struct dev_pm_domain hdmi_pm_domain; }; #define CREATE_TRACE_POINTS @@ -594,6 +602,7 @@ enum { #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ +#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 power well support */ /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ @@ -942,6 +951,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!bus->no_response_fallback) + return -1; + if (!chip->polling_mode && chip->poll_count < 2) { snd_printdd(SFX "%s: azx_get_response timeout, " "polling the codec once: last cmd=0x%08x\n", @@ -1117,37 +1129,52 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct snd_dma_buffer *dmab); #endif -/* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +/* enter link reset */ +static void azx_enter_link_reset(struct azx *chip) { unsigned long timeout; - if (!full_reset) - goto __skip; - - /* clear STATESTS */ - azx_writeb(chip, STATESTS, STATESTS_INT_MASK); - /* reset controller */ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); - while (azx_readb(chip, GCTL) && + while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && time_before(jiffies, timeout)) usleep_range(500, 1000); +} - /* delay for >= 100us for codec PLL to settle per spec - * Rev 0.9 section 5.5.1 - */ - usleep_range(500, 1000); +/* exit link reset */ +static void azx_exit_link_reset(struct azx *chip) +{ + unsigned long timeout; - /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); while (!azx_readb(chip, GCTL) && time_before(jiffies, timeout)) usleep_range(500, 1000); +} + +/* reset codec link */ +static int azx_reset(struct azx *chip, int full_reset) +{ + if (!full_reset) + goto __skip; + + /* clear STATESTS */ + azx_writew(chip, STATESTS, STATESTS_INT_MASK); + + /* reset controller */ + azx_enter_link_reset(chip); + + /* delay for >= 100us for codec PLL to settle per spec + * Rev 0.9 section 5.5.1 + */ + usleep_range(500, 1000); + + /* Bring controller out of reset */ + azx_exit_link_reset(chip); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ usleep_range(1000, 1200); @@ -1218,7 +1245,7 @@ static void azx_int_clear(struct azx *chip) } /* clear STATESTS */ - azx_writeb(chip, STATESTS, STATESTS_INT_MASK); + azx_writew(chip, STATESTS, STATESTS_INT_MASK); /* clear rirb status */ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); @@ -1373,8 +1400,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) int i, ok; #ifdef CONFIG_PM_RUNTIME - if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) - return IRQ_NONE; + if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; #endif spin_lock(&chip->reg_lock); @@ -1385,7 +1413,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) } status = azx_readl(chip, INTSTS); - if (status == 0) { + if (status == 0 || status == 0xffffffff) { spin_unlock(&chip->reg_lock); return IRQ_NONE; } @@ -1427,8 +1455,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) #if 0 /* clear state status int */ - if (azx_readb(chip, STATESTS) & 0x04) - azx_writeb(chip, STATESTS, 0x04); + if (azx_readw(chip, STATESTS) & 0x04) + azx_writew(chip, STATESTS, 0x04); #endif spin_unlock(&chip->reg_lock); @@ -2891,6 +2919,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); + azx_enter_link_reset(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2900,6 +2929,8 @@ static int azx_suspend(struct device *dev) pci_disable_device(pci); pci_save_state(pci); pci_set_power_state(pci, PCI_D3hot); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + hda_display_power(false); return 0; } @@ -2912,6 +2943,8 @@ static int azx_resume(struct device *dev) if (chip->disabled) return 0; + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + hda_display_power(true); pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); if (pci_enable_device(pci) < 0) { @@ -2942,8 +2975,21 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; + if (chip->disabled) + return 0; + + if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + return 0; + + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + azx_stop_chip(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + hda_display_power(false); return 0; } @@ -2951,9 +2997,37 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; + struct hda_bus *bus; + struct hda_codec *codec; + int status; + + if (chip->disabled) + return 0; + + if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + return 0; + + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + hda_display_power(true); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); azx_init_pci(chip); azx_init_chip(chip, 1); + + bus = chip->bus; + if (status && bus) { + list_for_each_entry(codec, &bus->codec_list, list) + if (status & (1 << codec->addr)) + queue_delayed_work(codec->bus->workq, + &codec->jackpoll_work, codec->jackpoll_interval); + } + + /* disable controller Wake Up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + return 0; } @@ -2962,6 +3036,9 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; + if (chip->disabled) + return 0; + if (!power_save_controller || !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return -EBUSY; @@ -3006,7 +3083,6 @@ static void azx_notifier_unregister(struct azx *chip) unregister_reboot_notifier(&chip->reboot_notifier); } -static int azx_first_init(struct azx *chip); static int azx_probe_continue(struct azx *chip); #ifdef SUPPORT_VGA_SWITCHEROO @@ -3033,8 +3109,7 @@ static void azx_vs_set_state(struct pci_dev *pci, snd_printk(KERN_INFO SFX "%s: Start delayed initialization\n", pci_name(chip->pci)); - if (azx_first_init(chip) < 0 || - azx_probe_continue(chip) < 0) { + if (azx_probe_continue(chip) < 0) { snd_printk(KERN_ERR SFX "%s: initialization error\n", pci_name(chip->pci)); @@ -3046,13 +3121,19 @@ static void azx_vs_set_state(struct pci_dev *pci, "%s: %s via VGA-switcheroo\n", pci_name(chip->pci), disabled ? "Disabling" : "Enabling"); if (disabled) { + pm_runtime_put_sync_suspend(&pci->dev); azx_suspend(&pci->dev); + /* when we get suspended by vga switcheroo we end up in D3cold, + * however we have no ACPI handle, so pci/acpi can't put us there, + * put ourselves there */ + pci->current_state = PCI_D3cold; chip->disabled = true; if (snd_hda_lock_devices(chip->bus)) snd_printk(KERN_WARNING SFX "%s: Cannot lock devices!\n", pci_name(chip->pci)); } else { snd_hda_unlock_devices(chip->bus); + pm_runtime_get_noresume(&pci->dev); chip->disabled = false; azx_resume(&pci->dev); } @@ -3107,6 +3188,9 @@ static int register_vga_switcheroo(struct azx *chip) if (err < 0) return err; chip->vga_switcheroo_registered = 1; + + /* register as an optimus hdmi audio power domain */ + vga_switcheroo_init_domain_pm_optimus_hdmi_audio(&chip->pci->dev, &chip->hdmi_pm_domain); return 0; } #else @@ -3120,8 +3204,13 @@ static int register_vga_switcheroo(struct azx *chip) */ static int azx_free(struct azx *chip) { + struct pci_dev *pci = chip->pci; int i; + if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) + && chip->running) + pm_runtime_get_noresume(&pci->dev); + azx_del_card_list(chip); azx_notifier_unregister(chip); @@ -3173,6 +3262,10 @@ static int azx_free(struct azx *chip) if (chip->fw) release_firmware(chip->fw); #endif + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + hda_display_power(false); + hda_i915_exit(); + } kfree(chip); return 0; @@ -3335,6 +3428,7 @@ static struct snd_pci_quirk msi_black_list[] = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ + SND_PCI_QUIRK(0x1179, 0xfb44, "Toshiba Satellite C870", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} @@ -3398,6 +3492,13 @@ static void azx_check_snoop_available(struct azx *chip) } } +#ifdef CONFIG_SND_HDA_I915 +static void azx_probe_work(struct work_struct *work) +{ + azx_probe_continue(container_of(work, struct azx, probe_work)); +} +#endif + /* * constructor */ @@ -3473,7 +3574,13 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, return err; } +#ifdef CONFIG_SND_HDA_I915 + /* continue probing in work context as may trigger request module */ + INIT_WORK(&chip->probe_work, azx_probe_work); +#endif + *rchip = chip; + return 0; } @@ -3730,11 +3837,6 @@ static int azx_probe(struct pci_dev *pci, } probe_now = !chip->disabled; - if (probe_now) { - err = azx_first_init(chip); - if (err < 0) - goto out_free; - } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (patch[dev] && *patch[dev]) { @@ -3749,30 +3851,53 @@ static int azx_probe(struct pci_dev *pci, } #endif /* CONFIG_SND_HDA_PATCH_LOADER */ + /* continue probing in work context, avoid request_module deadlock */ + if (probe_now && (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)) { +#ifdef CONFIG_SND_HDA_I915 + probe_now = false; + schedule_work(&chip->probe_work); +#else + snd_printk(KERN_ERR SFX "Haswell must build in CONFIG_SND_HDA_I915\n"); +#endif + } + if (probe_now) { err = azx_probe_continue(chip); if (err < 0) goto out_free; } - if (pci_dev_run_wake(pci)) - pm_runtime_put_noidle(&pci->dev); - dev++; complete_all(&chip->probe_wait); return 0; out_free: snd_card_free(card); - pci_set_drvdata(pci, NULL); return err; } static int azx_probe_continue(struct azx *chip) { + struct pci_dev *pci = chip->pci; int dev = chip->dev_index; int err; + /* Request power well for Haswell HDA controller and codec */ + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { +#ifdef CONFIG_SND_HDA_I915 + err = hda_i915_init(); + if (err < 0) { + snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); + goto out_free; + } +#endif + hda_display_power(true); + } + + err = azx_first_init(chip); + if (err < 0) + goto out_free; + #ifdef CONFIG_SND_HDA_INPUT_BEEP chip->beep_mode = beep_mode[dev]; #endif @@ -3817,6 +3942,8 @@ static int azx_probe_continue(struct azx *chip) power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); + if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo) + pm_runtime_put_noidle(&pci->dev); return 0; @@ -3829,12 +3956,8 @@ static void azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); - if (pci_dev_run_wake(pci)) - pm_runtime_get_noresume(&pci->dev); - if (card) snd_card_free(card); - pci_set_drvdata(pci, NULL); } /* PCI IDs */ @@ -3864,11 +3987,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | + AZX_DCAPS_I915_POWERWELL }, { PCI_DEVICE(0x8086, 0x0c0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | + AZX_DCAPS_I915_POWERWELL }, { PCI_DEVICE(0x8086, 0x0d0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | + AZX_DCAPS_I915_POWERWELL }, /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, @@ -3878,6 +4004,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* BayTrail */ + { PCI_DEVICE(0x8086, 0x0f04), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 9e0a952..05b3e3e 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_pin_sense); /** - * snd_hda_jack_detect - query pin Presence Detect status + * snd_hda_jack_detect_state - query pin Presence Detect status * @codec: the CODEC to sense * @nid: the pin NID to sense * - * Query and return the pin's Presence Detect status. + * Query and return the pin's Presence Detect status, as either + * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. */ -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return get_jack_plug_state(sense); + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack && jack->phantom_jack) + return HDA_JACK_PHANTOM; + else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + return HDA_JACK_PRESENT; + else + return HDA_JACK_NOT_PRESENT; } -EXPORT_SYMBOL_HDA(snd_hda_jack_detect); +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection @@ -247,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid) { - struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid); - struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid); + struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid); + struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid); if (!gated || !gating) return -EINVAL; @@ -398,7 +404,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, const char *base_name) { unsigned int def_conf, conn; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int idx, err; bool phantom_jack; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index ec12abd..379420c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); + +/* the jack state returned from snd_hda_jack_detect_state() */ +enum { + HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM, +}; + +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); + +static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; +} bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e0bf753..2e7493e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -562,6 +562,14 @@ static inline unsigned int get_wcaps_channels(u32 wcaps) return chans; } +static inline void snd_hda_override_wcaps(struct hda_codec *codec, + hda_nid_t nid, u32 val) +{ + if (nid >= codec->start_nid && + nid < codec->start_nid + codec->num_nodes) + codec->wcaps[nid - codec->start_nid] = val; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); @@ -667,7 +675,7 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, if (state & AC_PWRST_ERROR) return true; state = (state >> 4) & 0x0f; - return (state != target_state); + return (state == target_state); } unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0fee8fa..a8cb22e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -504,6 +504,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, int conn_len) { int c, curr = -1; + const hda_nid_t *list; + int cache_len; if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && @@ -521,6 +523,19 @@ static void print_conn_list(struct snd_info_buffer *buffer, } snd_iprintf(buffer, "\n"); } + + /* Get Cache connections info */ + cache_len = snd_hda_get_conn_list(codec, nid, &list); + if (cache_len != conn_len + || memcmp(list, conn, conn_len)) { + snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); + if (cache_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < cache_len; c++) + snd_iprintf(buffer, " 0x%02x", list[c]); + snd_iprintf(buffer, "\n"); + } + } } static void print_gpio(struct snd_info_buffer *buffer, @@ -567,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + u8 dev_list[AC_MAX_DEV_LIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEV_LIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i, + !!(dev_list[i] & AC_DE_PD), + !!(dev_list[i] & AC_DE_ELDV), + !!(dev_list[i] & AC_DE_IA)); + } +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -736,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 977b0d8..0cbdd87 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -32,7 +32,6 @@ #include "hda_jack.h" #include "hda_generic.h" -#define ENABLE_AD_STATIC_QUIRKS struct ad198x_spec { struct hda_gen_spec gen; @@ -43,114 +42,8 @@ struct ad198x_spec { hda_nid_t eapd_nid; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - -#ifdef ENABLE_AD_STATIC_QUIRKS - const struct snd_kcontrol_new *mixers[6]; - int num_mixers; - const struct hda_verb *init_verbs[6]; /* initialization verbs - * don't forget NULL termination! - */ - unsigned int num_init_verbs; - - /* playback */ - struct hda_multi_out multiout; /* playback set-up - * max_channels, dacs must be set - * dig_out_nid and hp_nid are optional - */ - unsigned int cur_eapd; - unsigned int need_dac_fix; - - /* capture */ - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - hda_nid_t dig_in_nid; /* digital-in NID; optional */ - - /* capture source */ - const struct hda_input_mux *input_mux; - const hda_nid_t *capsrc_nids; - unsigned int cur_mux[3]; - - /* channel model */ - const struct hda_channel_mode *channel_mode; - int num_channel_mode; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ - - unsigned int spdif_route; - - unsigned int jack_present: 1; - unsigned int inv_jack_detect: 1;/* inverted jack-detection */ - unsigned int analog_beep: 1; /* analog beep input present */ - unsigned int avoid_init_slave_vol:1; - -#ifdef CONFIG_PM - struct hda_loopback_check loopback; -#endif - /* for virtual master */ - hda_nid_t vmaster_nid; - const char * const *slave_vols; - const char * const *slave_sws; -#endif /* ENABLE_AD_STATIC_QUIRKS */ -}; - -#ifdef ENABLE_AD_STATIC_QUIRKS -/* - * input MUX handling (common part) - */ -static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} - -static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} - -static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->capsrc_nids[adc_idx], - &spec->cur_mux[adc_idx]); -} - -/* - * initialization (common callbacks) - */ -static int ad198x_init(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - return 0; -} - -static const char * const ad_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Mono", "Speaker", "IEC958", - NULL }; -static const char * const ad1988_6stack_fp_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", "IEC958", - NULL -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ #ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ @@ -160,12 +53,6 @@ static const struct snd_kcontrol_new ad_beep_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new ad_beep2_mixer[] = { - HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT), - HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT), - { } /* end */ -}; - #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ #else @@ -181,8 +68,7 @@ static int create_beep_ctls(struct hda_codec *codec) if (!spec->beep_amp) return 0; - knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer; - for ( ; knew->name; knew++) { + for (knew = ad_beep_mixer ; knew->name; knew++) { int err; struct snd_kcontrol *kctl; kctl = snd_ctl_new1(knew, codec); @@ -199,268 +85,6 @@ static int create_beep_ctls(struct hda_codec *codec) #define create_beep_ctls(codec) 0 #endif -#ifdef ENABLE_AD_STATIC_QUIRKS -static int ad198x_build_controls(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - struct snd_kcontrol *kctl; - unsigned int i; - int err; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid) { - err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; - } - - /* create beep controls if needed */ - err = create_beep_ctls(codec); - if (err < 0) - return err; - - /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - unsigned int vmaster_tlv[4]; - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, vmaster_tlv); - err = __snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, - (spec->slave_vols ? - spec->slave_vols : ad_slave_pfxs), - "Playback Volume", - !spec->avoid_init_slave_vol, NULL); - if (err < 0) - return err; - } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, - (spec->slave_sws ? - spec->slave_sws : ad_slave_pfxs), - "Playback Switch"); - if (err < 0) - return err; - } - - /* assign Capture Source enums to NID */ - kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); - if (!kctl) - kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); - for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]); - if (err < 0) - return err; - } - - /* assign IEC958 enums to NID */ - kctl = snd_hda_find_mixer_ctl(codec, - SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source"); - if (kctl) { - err = snd_hda_add_nid(codec, kctl, 0, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - } - - return 0; -} - -#ifdef CONFIG_PM -static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_check_amp_list_power(codec, &spec->loopback, nid); -} -#endif - -/* - * Analog playback callbacks - */ -static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital out - */ -static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); -} - -/* - * Analog capture - */ -static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); - return 0; -} - -static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); - return 0; -} - -/* - */ -static const struct hda_pcm_stream ad198x_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 6, /* changed later */ - .nid = 0, /* fill later */ - .ops = { - .open = ad198x_playback_pcm_open, - .prepare = ad198x_playback_pcm_prepare, - .cleanup = ad198x_playback_pcm_cleanup, - }, -}; - -static const struct hda_pcm_stream ad198x_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = ad198x_capture_pcm_prepare, - .cleanup = ad198x_capture_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream ad198x_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = ad198x_dig_playback_pcm_open, - .close = ad198x_dig_playback_pcm_close, - .prepare = ad198x_dig_playback_pcm_prepare, - .cleanup = ad198x_dig_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream ad198x_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in alc_build_pcms */ -}; - -static int ad198x_build_pcms(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "AD198x Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - - if (spec->multiout.dig_out_nid) { - info++; - codec->num_pcms++; - codec->spdif_status_reset = 1; - info->name = "AD198x Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } - } - - return 0; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, hda_nid_t hp) @@ -507,18 +131,6 @@ static void ad198x_shutup(struct hda_codec *codec) ad198x_power_eapd(codec); } -static void ad198x_free(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - - if (!spec) - return; - - snd_hda_gen_spec_free(&spec->gen); - kfree(spec); - snd_hda_detach_beep_device(codec); -} - #ifdef CONFIG_PM static int ad198x_suspend(struct hda_codec *codec) { @@ -527,65 +139,6 @@ static int ad198x_suspend(struct hda_codec *codec) } #endif -#ifdef ENABLE_AD_STATIC_QUIRKS -static const struct hda_codec_ops ad198x_patch_ops = { - .build_controls = ad198x_build_controls, - .build_pcms = ad198x_build_pcms, - .init = ad198x_init, - .free = ad198x_free, -#ifdef CONFIG_PM - .check_power_status = ad198x_check_power_status, - .suspend = ad198x_suspend, -#endif - .reboot_notify = ad198x_shutup, -}; - - -/* - * EAPD control - * the private value = nid - */ -#define ad198x_eapd_info snd_ctl_boolean_mono_info - -static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - if (codec->inv_eapd) - ucontrol->value.integer.value[0] = ! spec->cur_eapd; - else - ucontrol->value.integer.value[0] = spec->cur_eapd; - return 0; -} - -static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int eapd; - eapd = !!ucontrol->value.integer.value[0]; - if (codec->inv_eapd) - eapd = !eapd; - if (eapd == spec->cur_eapd) - return 0; - spec->cur_eapd = eapd; - snd_hda_codec_write_cache(codec, nid, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 1; -} - -static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * Automatic parse of I/O pins from the BIOS configuration @@ -646,537 +199,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) * AD1986A specific */ -#ifdef ENABLE_AD_STATIC_QUIRKS -#define AD1986A_SPDIF_OUT 0x02 -#define AD1986A_FRONT_DAC 0x03 -#define AD1986A_SURR_DAC 0x04 -#define AD1986A_CLFE_DAC 0x05 -#define AD1986A_ADC 0x06 - -static const hda_nid_t ad1986a_dac_nids[3] = { - AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC -}; -static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC }; -static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 }; - -static const struct hda_input_mux ad1986a_capture_source = { - .num_items = 7, - .items = { - { "Mic", 0x0 }, - { "CD", 0x1 }, - { "Aux", 0x3 }, - { "Line", 0x4 }, - { "Mix", 0x5 }, - { "Mono", 0x6 }, - { "Phone", 0x7 }, - }, -}; - - -static const struct hda_bind_ctls ad1986a_bind_pcm_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct hda_bind_ctls ad1986a_bind_pcm_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* - * mixers - */ -static const struct snd_kcontrol_new ad1986a_mixers[] = { - /* - * bind volumes/mutes of 3 DACs as a single PCM control for simplicity - */ - HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol), - HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw), - HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* additional mixers for 3stack mode */ -static const struct snd_kcontrol_new ad1986a_3st_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = ad198x_ch_mode_info, - .get = ad198x_ch_mode_get, - .put = ad198x_ch_mode_put, - }, - { } /* end */ -}; - -/* laptop model - 2ch only */ -static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; - -/* master controls both pins 0x1a and 0x1b */ -static const struct hda_bind_ctls ad1986a_laptop_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0, - }, -}; - -static const struct hda_bind_ctls ad1986a_laptop_master_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0, - }, -}; - -static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), - /* - HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -/* laptop-eapd model - 2ch only */ - -static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x4 }, - { "Mix", 0x5 }, - }, -}; - -static const struct hda_input_mux ad1986a_automic_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Mix", 0x5 }, - }, -}; - -static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "External Amplifier", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x1b, /* port-D */ - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT), - { } /* end */ -}; - -/* re-connect the mic boost input according to the jack sensing */ -static void ad1986a_automic(struct hda_codec *codec) -{ - unsigned int present; - present = snd_hda_jack_detect(codec, 0x1f); - /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */ - snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL, - present ? 0 : 2); -} - -#define AD1986A_MIC_EVENT 0x36 - -static void ad1986a_automic_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != AD1986A_MIC_EVENT) - return; - ad1986a_automic(codec); -} - -static int ad1986a_automic_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1986a_automic(codec); - return 0; -} - -/* laptop-automute - 2ch only */ - -static void ad1986a_update_hp(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - unsigned int mute; - - if (spec->jack_present) - mute = HDA_AMP_MUTE; /* mute internal speaker */ - else - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); -} - -static void ad1986a_hp_automute(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - - spec->jack_present = snd_hda_jack_detect(codec, 0x1a); - if (spec->inv_jack_detect) - spec->jack_present = !spec->jack_present; - ad1986a_update_hp(codec); -} - -#define AD1986A_HP_EVENT 0x37 - -static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) != AD1986A_HP_EVENT) - return; - ad1986a_hp_automute(codec); -} - -static int ad1986a_hp_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1986a_hp_automute(codec); - return 0; -} - -/* bind hp and internal speaker mute (with plug check) */ -static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - if (change) - ad1986a_update_hp(codec); - return change; -} - -static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1986a_hp_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - }, - { } /* end */ -}; - - -/* - * initialization verbs - */ -static const struct hda_verb ad1986a_init_verbs[] = { - /* Front, Surround, CLFE DAC; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Downmix - off */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* HP, Line-Out, Surround, CLFE selectors */ - {0x0a, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mono selector */ - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic selector: Mic 1/2 pin */ - {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Line-in selector: Line-in */ - {0x10, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic 1/2 swap */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Record selector: mic */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic, Phone, CD, Aux, Line-In amp; mute as default */ - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* PC beep */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* HP Pin */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* Front, Surround, CLFE Pins */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Mono Pin */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Mic Pin */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* Line, Aux, CD, Beep-In Pin */ - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - { } /* end */ -}; - -static const struct hda_verb ad1986a_ch2_init[] = { - /* Surround out -> Line In */ - { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* Line-in selectors */ - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 }, - /* CLFE -> Mic in */ - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 }, - { } /* end */ -}; - -static const struct hda_verb ad1986a_ch4_init[] = { - /* Surround out -> Surround */ - { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* CLFE -> Mic in */ - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 }, - { } /* end */ -}; - -static const struct hda_verb ad1986a_ch6_init[] = { - /* Surround out -> Surround out */ - { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* CLFE -> CLFE */ - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { } /* end */ -}; - -static const struct hda_channel_mode ad1986a_modes[3] = { - { 2, ad1986a_ch2_init }, - { 4, ad1986a_ch4_init }, - { 6, ad1986a_ch6_init }, -}; - -/* eapd initialization */ -static const struct hda_verb ad1986a_eapd_init_verbs[] = { - {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, - {} -}; - -static const struct hda_verb ad1986a_automic_verbs[] = { - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/ - {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT}, - {} -}; - -/* Ultra initialization */ -static const struct hda_verb ad1986a_ultra_init[] = { - /* eapd initialization */ - { 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, - /* CLFE -> Mic in */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 }, - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, - { } /* end */ -}; - -/* pin sensing on HP jack */ -static const struct hda_verb ad1986a_hp_init_verbs[] = { - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT}, - {} -}; - -static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case AD1986A_HP_EVENT: - ad1986a_hp_automute(codec); - break; - case AD1986A_MIC_EVENT: - ad1986a_automic(codec); - break; - } -} - -static int ad1986a_samsung_p50_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1986a_hp_automute(codec); - ad1986a_automic(codec); - return 0; -} - - -/* models */ -enum { - AD1986A_AUTO, - AD1986A_6STACK, - AD1986A_3STACK, - AD1986A_LAPTOP, - AD1986A_LAPTOP_EAPD, - AD1986A_LAPTOP_AUTOMUTE, - AD1986A_ULTRA, - AD1986A_SAMSUNG, - AD1986A_SAMSUNG_P50, - AD1986A_MODELS -}; - -static const char * const ad1986a_models[AD1986A_MODELS] = { - [AD1986A_AUTO] = "auto", - [AD1986A_6STACK] = "6stack", - [AD1986A_3STACK] = "3stack", - [AD1986A_LAPTOP] = "laptop", - [AD1986A_LAPTOP_EAPD] = "laptop-eapd", - [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", - [AD1986A_ULTRA] = "ultra", - [AD1986A_SAMSUNG] = "samsung", - [AD1986A_SAMSUNG_P50] = "samsung-p50", -}; - -static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), - SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), - SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK), - SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), - SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50), - SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), - SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), - SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), - SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), - {} -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1986a_loopbacks[] = { - { 0x13, HDA_OUTPUT, 0 }, /* Mic */ - { 0x14, HDA_OUTPUT, 0 }, /* Phone */ - { 0x15, HDA_OUTPUT, 0 }, /* CD */ - { 0x16, HDA_OUTPUT, 0 }, /* Aux */ - { 0x17, HDA_OUTPUT, 0 }, /* Line */ - { } /* end */ -}; -#endif - -static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int conf = snd_hda_codec_get_pincfg(codec, nid); - return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ - static int alloc_ad_spec(struct hda_codec *codec) { struct ad198x_spec *spec; @@ -1203,6 +225,11 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec, enum { AD1986A_FIXUP_INV_JACK_DETECT, + AD1986A_FIXUP_ULTRA, + AD1986A_FIXUP_SAMSUNG, + AD1986A_FIXUP_3STACK, + AD1986A_FIXUP_LAPTOP, + AD1986A_FIXUP_LAPTOP_IMIC, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -1210,16 +237,86 @@ static const struct hda_fixup ad1986a_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = ad_fixup_inv_jack_detect, }, + [AD1986A_FIXUP_ULTRA] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1d, 0x90a7013e }, /* int mic */ + {} + }, + }, + [AD1986A_FIXUP_SAMSUNG] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1d, 0x90a7013e }, /* int mic */ + { 0x20, 0x411111f0 }, /* N/A */ + { 0x24, 0x411111f0 }, /* N/A */ + {} + }, + }, + [AD1986A_FIXUP_3STACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02214021 }, /* headphone */ + { 0x1b, 0x01014011 }, /* front */ + { 0x1c, 0x01013012 }, /* surround */ + { 0x1d, 0x01019015 }, /* clfe */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x02a190f0 }, /* mic */ + { 0x20, 0x018130f0 }, /* line-in */ + {} + }, + }, + [AD1986A_FIXUP_LAPTOP] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02214021 }, /* headphone */ + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x02a191f0 }, /* mic */ + { 0x20, 0x411111f0 }, /* N/A */ + {} + }, + }, + [AD1986A_FIXUP_LAPTOP_IMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1d, 0x90a7013e }, /* int mic */ + {} + }, + .chained_before = 1, + .chain_id = AD1986A_FIXUP_LAPTOP, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), + SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), + SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK), + {} +}; + +static const struct hda_model_fixup ad1986a_fixup_models[] = { + { .id = AD1986A_FIXUP_3STACK, .name = "3stack" }, + { .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" }, + { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" }, + { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */ {} }; /* */ -static int ad1986a_parse_auto_config(struct hda_codec *codec) +static int patch_ad1986a(struct hda_codec *codec) { int err; struct ad198x_spec *spec; @@ -1244,7 +341,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) */ spec->gen.multiout.no_share_stream = 1; - snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups); + snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl, + ad1986a_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); err = ad198x_parse_auto_config(codec); @@ -1258,330 +356,11 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) return 0; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1986a(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1986A_MODELS, - ad1986a_models, - ad1986a_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1986A_AUTO; - } - - if (board_config == AD1986A_AUTO) - return ad1986a_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x19); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 6; - spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); - spec->multiout.dac_nids = ad1986a_dac_nids; - spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = ad1986a_adc_nids; - spec->capsrc_nids = ad1986a_capsrc_nids; - spec->input_mux = &ad1986a_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1986a_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1986a_init_verbs; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1986a_loopbacks; -#endif - spec->vmaster_nid = 0x1b; - codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */ - - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - case AD1986A_3STACK: - spec->num_mixers = 2; - spec->mixers[1] = ad1986a_3st_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1986a_ch2_init; - spec->channel_mode = ad1986a_modes; - spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); - spec->need_dac_fix = 1; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - break; - case AD1986A_LAPTOP: - spec->mixers[0] = ad1986a_laptop_mixers; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - break; - case AD1986A_LAPTOP_EAPD: - spec->num_mixers = 3; - spec->mixers[0] = ad1986a_laptop_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->mixers[2] = ad1986a_laptop_intmic_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_laptop_eapd_capture_source; - break; - case AD1986A_SAMSUNG: - spec->num_mixers = 2; - spec->mixers[0] = ad1986a_laptop_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 3; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->init_verbs[2] = ad1986a_automic_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_automic_capture_source; - codec->patch_ops.unsol_event = ad1986a_automic_unsol_event; - codec->patch_ops.init = ad1986a_automic_init; - break; - case AD1986A_SAMSUNG_P50: - spec->num_mixers = 2; - spec->mixers[0] = ad1986a_automute_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 4; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->init_verbs[2] = ad1986a_automic_verbs; - spec->init_verbs[3] = ad1986a_hp_init_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_automic_capture_source; - codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event; - codec->patch_ops.init = ad1986a_samsung_p50_init; - break; - case AD1986A_LAPTOP_AUTOMUTE: - spec->num_mixers = 3; - spec->mixers[0] = ad1986a_automute_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->mixers[2] = ad1986a_laptop_intmic_mixers; - spec->num_init_verbs = 3; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->init_verbs[2] = ad1986a_hp_init_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_laptop_eapd_capture_source; - codec->patch_ops.unsol_event = ad1986a_hp_unsol_event; - codec->patch_ops.init = ad1986a_hp_init; - /* Lenovo N100 seems to report the reversed bit - * for HP jack-sensing - */ - spec->inv_jack_detect = 1; - break; - case AD1986A_ULTRA: - spec->mixers[0] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1986a_ultra_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - spec->multiout.dig_out_nid = 0; - break; - } - - /* AD1986A has a hardware problem that it can't share a stream - * with multiple output pins. The copy of front to surrounds - * causes noisy or silent outputs at a certain timing, e.g. - * changing the volume. - * So, let's disable the shared stream. - */ - spec->multiout.no_share_stream = 1; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1986a ad1986a_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ /* * AD1983 specific */ -#ifdef ENABLE_AD_STATIC_QUIRKS -#define AD1983_SPDIF_OUT 0x02 -#define AD1983_DAC 0x03 -#define AD1983_ADC 0x04 - -static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC }; -static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC }; -static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 }; - -static const struct hda_input_mux ad1983_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x1 }, - { "Mix", 0x2 }, - { "Mix Mono", 0x3 }, - }, -}; - -/* - * SPDIF playback route - */ -static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { "PCM", "ADC" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->spdif_route; - return 0; -} - -static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - if (ucontrol->value.enumerated.item[0] > 1) - return -EINVAL; - if (spec->spdif_route != ucontrol->value.enumerated.item[0]) { - spec->spdif_route = ucontrol->value.enumerated.item[0]; - snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0, - AC_VERB_SET_CONNECT_SEL, - spec->spdif_route); - return 1; - } - return 0; -} - -static const struct snd_kcontrol_new ad1983_mixers[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct hda_verb ad1983_init_verbs[] = { - /* Front, HP, Mono; mute as default */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Beep, PCM, Mic, Line-In: mute */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Front, HP selectors; from Mix */ - {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x06, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* Mono selector; from Mix */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic selector; Mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Line-in selector: Line-in */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic boost: 0dB */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* Record selector: mic */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* SPDIF route: PCM */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Front Pin */ - {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* HP Pin */ - {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* Mono Pin */ - {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Mic Pin */ - {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* Line Pin */ - {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1983_loopbacks[] = { - { 0x12, HDA_OUTPUT, 0 }, /* Mic */ - { 0x13, HDA_OUTPUT, 0 }, /* Line */ - { } /* end */ -}; -#endif - -/* models */ -enum { - AD1983_AUTO, - AD1983_BASIC, - AD1983_MODELS -}; - -static const char * const ad1983_models[AD1983_MODELS] = { - [AD1983_AUTO] = "auto", - [AD1983_BASIC] = "basic", -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - /* * SPDIF mux control for AD1983 auto-parser */ @@ -1656,7 +435,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) return 0; } -static int ad1983_parse_auto_config(struct hda_codec *codec) +static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -1681,437 +460,19 @@ static int ad1983_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1983(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int board_config; - int err; - - board_config = snd_hda_check_board_config(codec, AD1983_MODELS, - ad1983_models, NULL); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1983_AUTO; - } - - if (board_config == AD1983_AUTO) - return ad1983_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids); - spec->multiout.dac_nids = ad1983_dac_nids; - spec->multiout.dig_out_nid = AD1983_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = ad1983_adc_nids; - spec->capsrc_nids = ad1983_capsrc_nids; - spec->input_mux = &ad1983_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1983_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1983_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1983_loopbacks; -#endif - spec->vmaster_nid = 0x05; - - codec->patch_ops = ad198x_patch_ops; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1983 ad1983_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * AD1981 HD specific */ -#ifdef ENABLE_AD_STATIC_QUIRKS -#define AD1981_SPDIF_OUT 0x02 -#define AD1981_DAC 0x03 -#define AD1981_ADC 0x04 - -static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC }; -static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC }; -static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 }; - -/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */ -static const struct hda_input_mux ad1981_capture_source = { - .num_items = 7, - .items = { - { "Front Mic", 0x0 }, - { "Line", 0x1 }, - { "Mix", 0x2 }, - { "Mix Mono", 0x3 }, - { "CD", 0x4 }, - { "Mic", 0x6 }, - { "Aux", 0x7 }, - }, -}; - -static const struct snd_kcontrol_new ad1981_mixers[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* identical with AD1983 */ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct hda_verb ad1981_init_verbs[] = { - /* Front, HP, Mono; mute as default */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Front, HP selectors; from Mix */ - {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x06, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* Mono selector; from Mix */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic Mixer; select Front Mic */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Mic boost: 0dB */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Record selector: Front mic */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* SPDIF route: PCM */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Front Pin */ - {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* HP Pin */ - {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* Mono Pin */ - {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Front & Rear Mic Pins */ - {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* Line Pin */ - {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* Digital Beep */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Line-Out as Input: disabled */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1981_loopbacks[] = { - { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */ - { 0x13, HDA_OUTPUT, 0 }, /* Line */ - { 0x1b, HDA_OUTPUT, 0 }, /* Aux */ - { 0x1c, HDA_OUTPUT, 0 }, /* Mic */ - { 0x1d, HDA_OUTPUT, 0 }, /* CD */ - { } /* end */ -}; -#endif - -/* - * Patch for HP nx6320 - * - * nx6320 uses EAPD in the reverse way - EAPD-on means the internal - * speaker output enabled _and_ mute-LED off. - */ - -#define AD1981_HP_EVENT 0x37 -#define AD1981_MIC_EVENT 0x38 - -static const struct hda_verb ad1981_hp_init_verbs[] = { - {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */ - /* pin sensing on HP and Mic jacks */ - {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, - {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT}, - {} -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - if (! ad198x_eapd_put(kcontrol, ucontrol)) - return 0; - /* change speaker pin appropriately */ - snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0); - /* toggle HP mute appropriately */ - snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - spec->cur_eapd ? 0 : HDA_AMP_MUTE); - return 1; -} - -/* bind volumes of both NID 0x05 and 0x06 */ -static const struct hda_bind_ctls ad1981_hp_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* mute internal speaker if HP is plugged */ -static void ad1981_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x06); - snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -/* toggle input of built-in and mic jack appropriately */ -static void ad1981_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x08); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - -/* unsolicited event for HP jack sensing */ -static void ad1981_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case AD1981_HP_EVENT: - ad1981_hp_automute(codec); - break; - case AD1981_MIC_EVENT: - ad1981_hp_automic(codec); - break; - } -} - -static const struct hda_input_mux ad1981_hp_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Dock Mic", 0x1 }, - { "Mix", 0x2 }, - }, -}; - -static const struct snd_kcontrol_new ad1981_hp_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | 0x05, - .name = "Master Playback Switch", - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad1981_hp_master_sw_put, - .private_value = 0x05, - }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), -#if 0 - /* FIXME: analog mic/line loopback doesn't work with my tests... - * (although recording is OK) - */ - HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), - /* FIXME: does this laptop have analog CD connection? */ - HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), -#endif - HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int ad1981_hp_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1981_hp_automute(codec); - ad1981_hp_automic(codec); - return 0; -} - -/* configuration for Toshiba Laptops */ -static const struct hda_verb ad1981_toshiba_init_verbs[] = { - {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */ - /* pin sensing on HP and Mic jacks */ - {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, - {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT}, - {} -}; - -static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT), - { } -}; - -/* configuration for Lenovo Thinkpad T60 */ -static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* identical with AD1983 */ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct hda_input_mux ad1981_thinkpad_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Mix", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* models */ -enum { - AD1981_AUTO, - AD1981_BASIC, - AD1981_HP, - AD1981_THINKPAD, - AD1981_TOSHIBA, - AD1981_MODELS -}; - -static const char * const ad1981_models[AD1981_MODELS] = { - [AD1981_AUTO] = "auto", - [AD1981_HP] = "hp", - [AD1981_THINKPAD] = "thinkpad", - [AD1981_BASIC] = "basic", - [AD1981_TOSHIBA] = "toshiba" -}; - -static const struct snd_pci_quirk ad1981_cfg_tbl[] = { - SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), - SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), - /* All HP models */ - SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), - /* Lenovo Thinkpad T60/X60/Z6xx */ - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD), - /* HP nx6320 (reversed SSID, H/W bug) */ - SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), - {} -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - /* follow EAPD via vmaster hook */ static void ad_vmaster_eapd_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; struct ad198x_spec *spec = codec->spec; + + if (!spec->eapd_nid) + return; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); @@ -2169,7 +530,7 @@ static const struct snd_pci_quirk ad1981_fixup_tbl[] = { {} }; -static int ad1981_parse_auto_config(struct hda_codec *codec) +static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -2202,110 +563,6 @@ static int ad1981_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1981(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1981_MODELS, - ad1981_models, - ad1981_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1981_AUTO; - } - - if (board_config == AD1981_AUTO) - return ad1981_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return -ENOMEM; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids); - spec->multiout.dac_nids = ad1981_dac_nids; - spec->multiout.dig_out_nid = AD1981_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = ad1981_adc_nids; - spec->capsrc_nids = ad1981_capsrc_nids; - spec->input_mux = &ad1981_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1981_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1981_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1981_loopbacks; -#endif - spec->vmaster_nid = 0x05; - - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - case AD1981_HP: - spec->mixers[0] = ad1981_hp_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1981_hp_init_verbs; - if (!is_jack_available(codec, 0x0a)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1981_hp_capture_source; - - codec->patch_ops.init = ad1981_hp_init; - codec->patch_ops.unsol_event = ad1981_hp_unsol_event; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1981_THINKPAD: - spec->mixers[0] = ad1981_thinkpad_mixers; - spec->input_mux = &ad1981_thinkpad_capture_source; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1981_TOSHIBA: - spec->mixers[0] = ad1981_hp_mixers; - spec->mixers[1] = ad1981_toshiba_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1981_toshiba_init_verbs; - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1981_hp_capture_source; - codec->patch_ops.init = ad1981_hp_init; - codec->patch_ops.unsol_event = ad1981_hp_unsol_event; - break; - } - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1981 ad1981_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * AD1988 @@ -2392,90 +649,7 @@ static int patch_ad1981(struct hda_codec *codec) * E/F quad mic array */ - #ifdef ENABLE_AD_STATIC_QUIRKS -/* models */ -enum { - AD1988_AUTO, - AD1988_6STACK, - AD1988_6STACK_DIG, - AD1988_3STACK, - AD1988_3STACK_DIG, - AD1988_LAPTOP, - AD1988_LAPTOP_DIG, - AD1988_MODEL_LAST, -}; - -/* reivision id to check workarounds */ -#define AD1988A_REV2 0x100200 - -#define is_rev2(codec) \ - ((codec)->vendor_id == 0x11d41988 && \ - (codec)->revision_id == AD1988A_REV2) - -/* - * mixers - */ - -static const hda_nid_t ad1988_6stack_dac_nids[4] = { - 0x04, 0x06, 0x05, 0x0a -}; - -static const hda_nid_t ad1988_3stack_dac_nids[3] = { - 0x04, 0x05, 0x0a -}; - -/* for AD1988A revision-2, DAC2-4 are swapped */ -static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { - 0x04, 0x05, 0x0a, 0x06 -}; - -static const hda_nid_t ad1988_alt_dac_nid[1] = { - 0x03 -}; - -static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { - 0x04, 0x0a, 0x06 -}; - -static const hda_nid_t ad1988_adc_nids[3] = { - 0x08, 0x09, 0x0f -}; - -static const hda_nid_t ad1988_capsrc_nids[3] = { - 0x0c, 0x0d, 0x0e -}; - -#define AD1988_SPDIF_OUT 0x02 -#define AD1988_SPDIF_OUT_HDMI 0x0b -#define AD1988_SPDIF_IN 0x07 - -static const hda_nid_t ad1989b_slave_dig_outs[] = { - AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 -}; - -static const struct hda_input_mux ad1988_6stack_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x1 }, /* port-B */ - { "Line", 0x2 }, /* port-C */ - { "Mic", 0x4 }, /* port-E */ - { "CD", 0x5 }, - { "Mix", 0x9 }, - }, -}; - -static const struct hda_input_mux ad1988_laptop_capture_source = { - .num_items = 3, - .items = { - { "Mic/Line", 0x1 }, /* port-B */ - { "CD", 0x5 }, - { "Mix", 0x9 }, - }, -}; - -/* - */ static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2506,569 +680,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, spec->multiout.num_dacs = spec->multiout.max_channels / 2; return err; } - -/* 6-stack mode */ -static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* 3-stack mode */ -static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = ad198x_ch_mode_info, - .get = ad198x_ch_mode_get, - .put = ad198x_ch_mode_put, - }, - - { } /* end */ -}; - -/* laptop mode */ -static const struct snd_kcontrol_new ad1988_laptop_mixers[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "External Amplifier", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x12, - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x12, /* port-D */ - }, - - { } /* end */ -}; - -/* capture */ -static const struct snd_kcontrol_new ad1988_capture_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 3, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "PCM", "ADC1", "ADC2", "ADC3" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= 4) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int sel; - - sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT); - if (!(sel & 0x80)) - ucontrol->value.enumerated.item[0] = 0; - else { - sel = snd_hda_codec_read(codec, 0x0b, 0, - AC_VERB_GET_CONNECT_SEL, 0); - if (sel < 3) - sel++; - else - sel = 0; - ucontrol->value.enumerated.item[0] = sel; - } - return 0; -} - -static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int val, sel; - int change; - - val = ucontrol->value.enumerated.item[0]; - if (val > 3) - return -EINVAL; - if (!val) { - sel = snd_hda_codec_read(codec, 0x1d, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT); - change = sel & 0x80; - if (change) { - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(0)); - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(1)); - } - } else { - sel = snd_hda_codec_read(codec, 0x1d, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT | 0x01); - change = sel & 0x80; - if (change) { - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(0)); - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(1)); - } - sel = snd_hda_codec_read(codec, 0x0b, 0, - AC_VERB_GET_CONNECT_SEL, 0) + 1; - change |= sel != val; - if (change) - snd_hda_codec_write_cache(codec, 0x0b, 0, - AC_VERB_SET_CONNECT_SEL, - val - 1); - } - return change; -} - -static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Source", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, - .info = ad1988_spdif_playback_source_info, - .get = ad1988_spdif_playback_source_get, - .put = ad1988_spdif_playback_source_put, - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { - HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* - * initialization verbs - */ - -/* - * for 6-stack (+dig) - */ -static const struct hda_verb ad1988_6stack_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-D line-out path */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-F surround path */ - {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-G CLFE path */ - {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-H side path */ - {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B front mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C line-in path */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Port-E mic-in path */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Analog CD Input */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - - { } -}; - -static const struct hda_verb ad1988_6stack_fp_init_verbs[] = { - /* Headphone; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - - { } -}; - -static const struct hda_verb ad1988_capture_init_verbs[] = { - /* mute analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* select ADCs - front-mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - - { } -}; - -static const struct hda_verb ad1988_spdif_init_verbs[] = { - /* SPDIF out sel */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* SPDIF out pin */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - - { } -}; - -static const struct hda_verb ad1988_spdif_in_init_verbs[] = { - /* unmute SPDIF input pin */ - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { } -}; - -/* AD1989 has no ADC -> SPDIF route */ -static const struct hda_verb ad1989_spdif_init_verbs[] = { - /* SPDIF-1 out pin */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - /* SPDIF-2/HDMI out pin */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } -}; - -/* - * verbs for 3stack (+dig) - */ -static const struct hda_verb ad1988_3stack_ch2_init[] = { - /* set port-C to line-in */ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* set port-E to mic-in */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { } /* end */ -}; - -static const struct hda_verb ad1988_3stack_ch6_init[] = { - /* set port-C to surround out */ - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - /* set port-E to CLFE out */ - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -static const struct hda_channel_mode ad1988_3stack_modes[2] = { - { 2, ad1988_3stack_ch2_init }, - { 6, ad1988_3stack_ch6_init }, -}; - -static const struct hda_verb ad1988_3stack_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-D line-out path */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B front mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C line-in/surround path - 6ch mode as default */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */ - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Port-E mic-in/CLFE path - 6ch mode as default */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */ - {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* mute analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* select ADCs - front-mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - { } -}; - -/* - * verbs for laptop mode (+dig) - */ -static const struct hda_verb ad1988_laptop_hp_on[] = { - /* unmute port-A and mute port-D */ - { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; -static const struct hda_verb ad1988_laptop_hp_off[] = { - /* mute port-A and unmute port-D */ - { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -#define AD1988_HP_EVENT 0x01 - -static const struct hda_verb ad1988_laptop_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT }, - /* Port-D line-out path + EAPD */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */ - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C docking station - try to output */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* mute analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* select ADCs - mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - { } -}; - -static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) != AD1988_HP_EVENT) - return; - if (snd_hda_jack_detect(codec, 0x11)) - snd_hda_sequence_write(codec, ad1988_laptop_hp_on); - else - snd_hda_sequence_write(codec, ad1988_laptop_hp_off); -} - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1988_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Line */ - { 0x20, HDA_INPUT, 4 }, /* Mic */ - { 0x20, HDA_INPUT, 6 }, /* CD */ - { } /* end */ -}; -#endif #endif /* ENABLE_AD_STATIC_QUIRKS */ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, @@ -3217,7 +828,34 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec) /* */ -static int ad1988_parse_auto_config(struct hda_codec *codec) +enum { + AD1988_FIXUP_6STACK_DIG, +}; + +static const struct hda_fixup ad1988_fixups[] = { + [AD1988_FIXUP_6STACK_DIG] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x11, 0x02214130 }, /* front-hp */ + { 0x12, 0x01014010 }, /* line-out */ + { 0x14, 0x02a19122 }, /* front-mic */ + { 0x15, 0x01813021 }, /* line-in */ + { 0x16, 0x01011012 }, /* line-out */ + { 0x17, 0x01a19020 }, /* mic */ + { 0x1b, 0x0145f1f0 }, /* SPDIF */ + { 0x24, 0x01016011 }, /* line-out */ + { 0x25, 0x01012013 }, /* line-out */ + { } + } + }, +}; + +static const struct hda_model_fixup ad1988_fixup_models[] = { + { .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" }, + {} +}; + +static int patch_ad1988(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -3231,12 +869,19 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + + snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + err = ad198x_parse_auto_config(codec); if (err < 0) goto error; err = ad1988_add_spdif_mux_ctl(codec); if (err < 0) goto error; + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: @@ -3244,169 +889,6 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) return err; } -/* - */ - -#ifdef ENABLE_AD_STATIC_QUIRKS -static const char * const ad1988_models[AD1988_MODEL_LAST] = { - [AD1988_6STACK] = "6stack", - [AD1988_6STACK_DIG] = "6stack-dig", - [AD1988_3STACK] = "3stack", - [AD1988_3STACK_DIG] = "3stack-dig", - [AD1988_LAPTOP] = "laptop", - [AD1988_LAPTOP_DIG] = "laptop-dig", - [AD1988_AUTO] = "auto", -}; - -static const struct snd_pci_quirk ad1988_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), - {} -}; - -static int patch_ad1988(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST, - ad1988_models, ad1988_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1988_AUTO; - } - - if (board_config == AD1988_AUTO) - return ad1988_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - if (is_rev2(codec)) - snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n"); - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = ad1988_alt_dac_nid[0]; - switch (board_config) { - case AD1988_6STACK: - case AD1988_6STACK_DIG: - spec->multiout.max_channels = 8; - spec->multiout.num_dacs = 4; - if (is_rev2(codec)) - spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2; - else - spec->multiout.dac_nids = ad1988_6stack_dac_nids; - spec->input_mux = &ad1988_6stack_capture_source; - spec->num_mixers = 2; - if (is_rev2(codec)) - spec->mixers[0] = ad1988_6stack_mixers1_rev2; - else - spec->mixers[0] = ad1988_6stack_mixers1; - spec->mixers[1] = ad1988_6stack_mixers2; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_6stack_init_verbs; - if (board_config == AD1988_6STACK_DIG) { - spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; - spec->dig_in_nid = AD1988_SPDIF_IN; - } - break; - case AD1988_3STACK: - case AD1988_3STACK_DIG: - spec->multiout.max_channels = 6; - spec->multiout.num_dacs = 3; - if (is_rev2(codec)) - spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2; - else - spec->multiout.dac_nids = ad1988_3stack_dac_nids; - spec->input_mux = &ad1988_6stack_capture_source; - spec->channel_mode = ad1988_3stack_modes; - spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes); - spec->num_mixers = 2; - if (is_rev2(codec)) - spec->mixers[0] = ad1988_3stack_mixers1_rev2; - else - spec->mixers[0] = ad1988_3stack_mixers1; - spec->mixers[1] = ad1988_3stack_mixers2; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_3stack_init_verbs; - if (board_config == AD1988_3STACK_DIG) - spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; - break; - case AD1988_LAPTOP: - case AD1988_LAPTOP_DIG: - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1988_3stack_dac_nids; - spec->input_mux = &ad1988_laptop_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1988_laptop_mixers; - codec->inv_eapd = 1; /* inverted EAPD */ - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_laptop_init_verbs; - if (board_config == AD1988_LAPTOP_DIG) - spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; - break; - } - - spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); - spec->adc_nids = ad1988_adc_nids; - spec->capsrc_nids = ad1988_capsrc_nids; - spec->mixers[spec->num_mixers++] = ad1988_capture_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs; - if (spec->multiout.dig_out_nid) { - if (codec->vendor_id >= 0x11d4989a) { - spec->mixers[spec->num_mixers++] = - ad1989_spdif_out_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1989_spdif_init_verbs; - codec->slave_dig_outs = ad1989b_slave_dig_outs; - } else { - spec->mixers[spec->num_mixers++] = - ad1988_spdif_out_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1988_spdif_init_verbs; - } - } - if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) { - spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1988_spdif_in_init_verbs; - } - - codec->patch_ops = ad198x_patch_ops; - switch (board_config) { - case AD1988_LAPTOP: - case AD1988_LAPTOP_DIG: - codec->patch_ops.unsol_event = ad1988_laptop_unsol_event; - break; - } -#ifdef CONFIG_PM - spec->loopback.amplist = ad1988_loopbacks; -#endif - spec->vmaster_nid = 0x04; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1988 ad1988_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * AD1884 / AD1984 @@ -3420,167 +902,19 @@ static int patch_ad1988(struct hda_codec *codec) * * AD1984 = AD1884 + two digital mic-ins * - * FIXME: - * For simplicity, we share the single DAC for both HP and line-outs - * right now. The inidividual playbacks could be easily implemented, - * but no build-up framework is given, so far. - */ - -#ifdef ENABLE_AD_STATIC_QUIRKS -static const hda_nid_t ad1884_dac_nids[1] = { - 0x04, -}; - -static const hda_nid_t ad1884_adc_nids[2] = { - 0x08, 0x09, -}; - -static const hda_nid_t ad1884_capsrc_nids[2] = { - 0x0c, 0x0d, -}; - -#define AD1884_SPDIF_OUT 0x02 - -static const struct hda_input_mux ad1884_capture_source = { - .num_items = 4, - .items = { - { "Front Mic", 0x0 }, - { "Mic", 0x1 }, - { "CD", 0x2 }, - { "Mix", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new ad1884_base_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984_dmic_mixers[] = { - HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, - HDA_INPUT), - HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, - HDA_INPUT), - { } /* end */ -}; - -/* - * initialization verbs + * AD1883 / AD1884A / AD1984A / AD1984B + * + * port-B (0x14) - front mic-in + * port-E (0x1c) - rear mic-in + * port-F (0x16) - CD / ext out + * port-C (0x15) - rear line-in + * port-D (0x12) - rear line-out + * port-A (0x11) - front hp-out + * + * AD1984A = AD1884A + digital-mic + * AD1883 = equivalent with AD1984A + * AD1984B = AD1984A + extra SPDIF-out */ -static const struct hda_verb ad1884_init_verbs[] = { - /* DACs; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-A (HP) mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* HP selector - select DAC2 */ - {0x22, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Port-D (Line-out) mixer */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-D pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono-out mixer */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Mono-out pin */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono selector */ - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Port-B (front mic) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Port-C (rear mic) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - /* SPDIF output selector */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1884_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Mic */ - { 0x20, HDA_INPUT, 2 }, /* CD */ - { 0x20, HDA_INPUT, 4 }, /* Docking */ - { } /* end */ -}; -#endif - -static const char * const ad1884_slave_vols[] = { - "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958", - NULL -}; - -enum { - AD1884_AUTO, - AD1884_BASIC, - AD1884_MODELS -}; - -static const char * const ad1884_models[AD1884_MODELS] = { - [AD1884_AUTO] = "auto", - [AD1884_BASIC] = "basic", -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* set the upper-limit for mixer amp to 0dB for avoiding the possible * damage by overloading @@ -3596,24 +930,56 @@ static void ad1884_fixup_amp_override(struct hda_codec *codec, (1 << AC_AMPCAP_MUTE_SHIFT)); } +/* toggle GPIO1 according to the mute state */ +static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct ad198x_spec *spec = codec->spec; + + if (spec->eapd_nid) + ad_vmaster_eapd_hook(private_data, enabled); + snd_hda_codec_update_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, + enabled ? 0x00 : 0x02); +} + static void ad1884_fixup_hp_eapd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct ad198x_spec *spec = codec->spec; + static const struct hda_verb gpio_init_verbs[] = { + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, + {}, + }; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; + snd_hda_sequence_write_cache(codec, gpio_init_verbs); + break; + case HDA_FIXUP_ACT_PROBE: if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) spec->eapd_nid = spec->gen.autocfg.line_out_pins[0]; else spec->eapd_nid = spec->gen.autocfg.speaker_pins[0]; - if (spec->eapd_nid) - spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + break; } } +/* set magic COEFs for dmic */ +static const struct hda_verb ad1884_dmic_init_verbs[] = { + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + {} +}; + enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, + AD1884_FIXUP_DMIC_COEF, + AD1884_FIXUP_HP_TOUCHSMART, }; static const struct hda_fixup ad1884_fixups[] = { @@ -3627,15 +993,27 @@ static const struct hda_fixup ad1884_fixups[] = { .chained = true, .chain_id = AD1884_FIXUP_AMP_OVERRIDE, }, + [AD1884_FIXUP_DMIC_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = ad1884_dmic_init_verbs, + }, + [AD1884_FIXUP_HP_TOUCHSMART] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = ad1884_dmic_init_verbs, + .chained = true, + .chain_id = AD1884_FIXUP_HP_EAPD, + }, }; static const struct snd_pci_quirk ad1884_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART), SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), {} }; -static int ad1884_parse_auto_config(struct hda_codec *codec) +static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -3668,1170 +1046,6 @@ static int ad1884_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1884_basic(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err; - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); - spec->multiout.dac_nids = ad1884_dac_nids; - spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; - spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids); - spec->adc_nids = ad1884_adc_nids; - spec->capsrc_nids = ad1884_capsrc_nids; - spec->input_mux = &ad1884_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1884_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1884_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1884_loopbacks; -#endif - spec->vmaster_nid = 0x04; - /* we need to cover all playback volumes */ - spec->slave_vols = ad1884_slave_vols; - /* slaves may contain input volumes, so we can't raise to 0dB blindly */ - spec->avoid_init_slave_vol = 1; - - codec->patch_ops = ad198x_patch_ops; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} - -static int patch_ad1884(struct hda_codec *codec) -{ - int board_config; - - board_config = snd_hda_check_board_config(codec, AD1884_MODELS, - ad1884_models, NULL); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1884_AUTO; - } - - if (board_config == AD1884_AUTO) - return ad1884_parse_auto_config(codec); - else - return patch_ad1884_basic(codec); -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1884 ad1884_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - -#ifdef ENABLE_AD_STATIC_QUIRKS -/* - * Lenovo Thinkpad T61/X61 - */ -static const struct hda_input_mux ad1984_thinkpad_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "Mix", 0x3 }, - { "Dock Mic", 0x4 }, - }, -}; - - -/* - * Dell Precision T3400 - */ -static const struct hda_input_mux ad1984_dell_desktop_capture_source = { - .num_items = 3, - .items = { - { "Front Mic", 0x0 }, - { "Line-In", 0x1 }, - { "Mix", 0x3 }, - }, -}; - - -static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -/* additional verbs */ -static const struct hda_verb ad1984_thinkpad_init_verbs[] = { - /* Port-E (docking station mic) pin */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* docking mic boost */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Analog PC Beeper - allow firmware/ACPI beeps */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a}, - /* Analog mixer - docking mic; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* enable EAPD bit */ - {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - { } /* end */ -}; - -/* - * Dell Precision T3400 - */ -static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -/* Digial MIC ADC NID 0x05 + 0x06 */ -static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - snd_hda_codec_setup_stream(codec, 0x05 + substream->number, - stream_tag, 0, format); - return 0; -} - -static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number); - return 0; -} - -static const struct hda_pcm_stream ad1984_pcm_dmic_capture = { - .substreams = 2, - .channels_min = 2, - .channels_max = 2, - .nid = 0x05, - .ops = { - .prepare = ad1984_pcm_dmic_prepare, - .cleanup = ad1984_pcm_dmic_cleanup - }, -}; - -static int ad1984_build_pcms(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - struct hda_pcm *info; - int err; - - err = ad198x_build_pcms(codec); - if (err < 0) - return err; - - info = spec->pcm_rec + codec->num_pcms; - codec->num_pcms++; - info->name = "AD1984 Digital Mic"; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture; - return 0; -} - -/* models */ -enum { - AD1984_AUTO, - AD1984_BASIC, - AD1984_THINKPAD, - AD1984_DELL_DESKTOP, - AD1984_MODELS -}; - -static const char * const ad1984_models[AD1984_MODELS] = { - [AD1984_AUTO] = "auto", - [AD1984_BASIC] = "basic", - [AD1984_THINKPAD] = "thinkpad", - [AD1984_DELL_DESKTOP] = "dell_desktop", -}; - -static const struct snd_pci_quirk ad1984_cfg_tbl[] = { - /* Lenovo Thinkpad T61/X61 */ - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), - SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), - SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP), - {} -}; - -static int patch_ad1984(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int board_config, err; - - board_config = snd_hda_check_board_config(codec, AD1984_MODELS, - ad1984_models, ad1984_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1984_AUTO; - } - - if (board_config == AD1984_AUTO) - return ad1884_parse_auto_config(codec); - - err = patch_ad1884_basic(codec); - if (err < 0) - return err; - spec = codec->spec; - - switch (board_config) { - case AD1984_BASIC: - /* additional digital mics */ - spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers; - codec->patch_ops.build_pcms = ad1984_build_pcms; - break; - case AD1984_THINKPAD: - if (codec->subsystem_id == 0x17aa20fb) { - /* Thinpad X300 does not have the ability to do SPDIF, - or attach to docking station to use SPDIF */ - spec->multiout.dig_out_nid = 0; - } else - spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; - spec->input_mux = &ad1984_thinkpad_capture_source; - spec->mixers[0] = ad1984_thinkpad_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; - spec->analog_beep = 1; - break; - case AD1984_DELL_DESKTOP: - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1984_dell_desktop_capture_source; - spec->mixers[0] = ad1984_dell_desktop_mixers; - break; - } - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1984 ad1884_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - -/* - * AD1883 / AD1884A / AD1984A / AD1984B - * - * port-B (0x14) - front mic-in - * port-E (0x1c) - rear mic-in - * port-F (0x16) - CD / ext out - * port-C (0x15) - rear line-in - * port-D (0x12) - rear line-out - * port-A (0x11) - front hp-out - * - * AD1984A = AD1884A + digital-mic - * AD1883 = equivalent with AD1984A - * AD1984B = AD1984A + extra SPDIF-out - * - * FIXME: - * We share the single DAC for both HP and line-outs (see AD1884/1984). - */ - -#ifdef ENABLE_AD_STATIC_QUIRKS -static const hda_nid_t ad1884a_dac_nids[1] = { - 0x03, -}; - -#define ad1884a_adc_nids ad1884_adc_nids -#define ad1884a_capsrc_nids ad1884_capsrc_nids - -#define AD1884A_SPDIF_OUT 0x02 - -static const struct hda_input_mux ad1884a_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x0 }, - { "Mic", 0x4 }, - { "Line", 0x1 }, - { "CD", 0x2 }, - { "Mix", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new ad1884a_base_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -/* - * initialization verbs - */ -static const struct hda_verb ad1884a_init_verbs[] = { - /* DACs; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - /* Port-A (HP) mixer - route only from analog mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-D (Line-out) mixer - route only from analog mixer */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-D pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono-out mixer - route only from analog mixer */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Mono-out pin */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-B (front mic) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Port-C (rear line-in) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Port-E (rear mic) pin */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */ - /* Port-F (CD) pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* capture sources */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* SPDIF output amp */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1884a_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Mic */ - { 0x20, HDA_INPUT, 2 }, /* CD */ - { 0x20, HDA_INPUT, 4 }, /* Docking */ - { } /* end */ -}; -#endif - -/* - * Laptop model - * - * Port A: Headphone jack - * Port B: MIC jack - * Port C: Internal MIC - * Port D: Dock Line Out (if enabled) - * Port E: Dock Line In (if enabled) - * Port F: Internal speakers - */ - -static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - int mute = (!ucontrol->value.integer.value[0] && - !ucontrol->value.integer.value[1]); - /* toggle GPIO1 according to the mute state */ - snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - mute ? 0x02 : 0x0); - return ret; -} - -static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1884a_mobile_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1884a_mobile_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* mute internal speaker if HP is plugged */ -static void ad1884a_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x11); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, - present ? 0x00 : 0x02); -} - -/* switch to external mic if plugged */ -static void ad1884a_hp_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x14); - snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, - present ? 0 : 1); -} - -#define AD1884A_HP_EVENT 0x37 -#define AD1884A_MIC_EVENT 0x36 - -/* unsolicited event for HP jack sensing */ -static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case AD1884A_HP_EVENT: - ad1884a_hp_automute(codec); - break; - case AD1884A_MIC_EVENT: - ad1884a_hp_automic(codec); - break; - } -} - -/* initialize jack-sensing, too */ -static int ad1884a_hp_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1884a_hp_automute(codec); - ad1884a_hp_automic(codec); - return 0; -} - -/* mute internal speaker if HP or docking HP is plugged */ -static void ad1884a_laptop_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x11); - if (!present) - present = snd_hda_jack_detect(codec, 0x12); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, - present ? 0x00 : 0x02); -} - -/* switch to external mic if plugged */ -static void ad1884a_laptop_automic(struct hda_codec *codec) -{ - unsigned int idx; - - if (snd_hda_jack_detect(codec, 0x14)) - idx = 0; - else if (snd_hda_jack_detect(codec, 0x1c)) - idx = 4; - else - idx = 1; - snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx); -} - -/* unsolicited event for HP jack sensing */ -static void ad1884a_laptop_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case AD1884A_HP_EVENT: - ad1884a_laptop_automute(codec); - break; - case AD1884A_MIC_EVENT: - ad1884a_laptop_automic(codec); - break; - } -} - -/* initialize jack-sensing, too */ -static int ad1884a_laptop_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1884a_laptop_automute(codec); - ad1884a_laptop_automic(codec); - return 0; -} - -/* additional verbs for laptop model */ -static const struct hda_verb ad1884a_laptop_verbs[] = { - /* Port-A (HP) pin - always unmuted */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-F (int speaker) mixer - route only from analog mixer */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F (int speaker) pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* required for compaq 6530s/6531s speaker output */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-C pin - internal mic-in */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* Port-D (docking line-out) pin - default unmuted */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - /* allow to touch GPIO1 (for mute control) */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ - { } /* end */ -}; - -static const struct hda_verb ad1884a_mobile_verbs[] = { - /* DACs; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - /* Port-A (HP) mixer - route only from analog mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-A (HP) pin - always unmuted */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-B (mic jack) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* Port-C (int mic) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* Port-F (int speaker) mixer - route only from analog mixer */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* capture sources */ - /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - /* allow to touch GPIO1 (for mute control) */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ - { } /* end */ -}; - -/* - * Thinkpad X300 - * 0x11 - HP - * 0x12 - speaker - * 0x14 - mic-in - * 0x17 - built-in mic - */ - -static const struct hda_verb ad1984a_thinkpad_verbs[] = { - /* HP unmute */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* turn on EAPD */ - {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - /* internal mic - dmic */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* set magic COEFs for dmic */ - {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, - {0x01, AC_VERB_SET_PROC_COEF, 0x08}, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -static const struct hda_input_mux ad1984a_thinkpad_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x5 }, - { "Mix", 0x3 }, - }, -}; - -/* mute internal speaker if HP is plugged */ -static void ad1984a_thinkpad_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x11); - snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -/* unsolicited event for HP jack sensing */ -static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != AD1884A_HP_EVENT) - return; - ad1984a_thinkpad_automute(codec); -} - -/* initialize jack-sensing, too */ -static int ad1984a_thinkpad_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1984a_thinkpad_automute(codec); - return 0; -} - -/* - * Precision R5500 - * 0x12 - HP/line-out - * 0x13 - speaker (mono) - * 0x15 - mic-in - */ - -static const struct hda_verb ad1984a_precision_verbs[] = { - /* Unmute main output path */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */ - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Select mic as input */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */ - /* Configure as mic */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* HP unmute */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* turn on EAPD */ - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - /* unsolicited event for pin-sense */ - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984a_precision_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - - -/* mute internal speaker if HP is plugged */ -static void ad1984a_precision_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x12); - snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - - -/* unsolicited event for HP jack sensing */ -static void ad1984a_precision_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != AD1884A_HP_EVENT) - return; - ad1984a_precision_automute(codec); -} - -/* initialize jack-sensing, too */ -static int ad1984a_precision_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1984a_precision_automute(codec); - return 0; -} - - -/* - * HP Touchsmart - * port-A (0x11) - front hp-out - * port-B (0x14) - unused - * port-C (0x15) - unused - * port-D (0x12) - rear line out - * port-E (0x1c) - front mic-in - * port-F (0x16) - Internal speakers - * digital-mic (0x17) - Internal mic - */ - -static const struct hda_verb ad1984a_touchsmart_verbs[] = { - /* DACs; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - /* Port-A (HP) mixer - route only from analog mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-A (HP) pin - always unmuted */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-E (int speaker) mixer - route only from analog mixer */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, - /* Port-E pin */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /* Port-F (int speaker) mixer - route only from analog mixer */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* capture sources */ - /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - /* allow to touch GPIO1 (for mute control) */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ - /* internal mic - dmic */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* set magic COEFs for dmic */ - {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, - {0x01, AC_VERB_SET_PROC_COEF, 0x08}, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), -/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_AMP_FLAG, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1884a_mobile_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT), - { } /* end */ -}; - -/* switch to external mic if plugged */ -static void ad1984a_touchsmart_automic(struct hda_codec *codec) -{ - if (snd_hda_jack_detect(codec, 0x1c)) - snd_hda_codec_write(codec, 0x0c, 0, - AC_VERB_SET_CONNECT_SEL, 0x4); - else - snd_hda_codec_write(codec, 0x0c, 0, - AC_VERB_SET_CONNECT_SEL, 0x5); -} - - -/* unsolicited event for HP jack sensing */ -static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case AD1884A_HP_EVENT: - ad1884a_hp_automute(codec); - break; - case AD1884A_MIC_EVENT: - ad1984a_touchsmart_automic(codec); - break; - } -} - -/* initialize jack-sensing, too */ -static int ad1984a_touchsmart_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1884a_hp_automute(codec); - ad1984a_touchsmart_automic(codec); - return 0; -} - - -/* - */ - -enum { - AD1884A_AUTO, - AD1884A_DESKTOP, - AD1884A_LAPTOP, - AD1884A_MOBILE, - AD1884A_THINKPAD, - AD1984A_TOUCHSMART, - AD1984A_PRECISION, - AD1884A_MODELS -}; - -static const char * const ad1884a_models[AD1884A_MODELS] = { - [AD1884A_AUTO] = "auto", - [AD1884A_DESKTOP] = "desktop", - [AD1884A_LAPTOP] = "laptop", - [AD1884A_MOBILE] = "mobile", - [AD1884A_THINKPAD] = "thinkpad", - [AD1984A_TOUCHSMART] = "touchsmart", - [AD1984A_PRECISION] = "precision", -}; - -static const struct snd_pci_quirk ad1884a_cfg_tbl[] = { - SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION), - SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), - SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), - SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), - {} -}; - -static int patch_ad1884a(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1884A_MODELS, - ad1884a_models, - ad1884a_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1884A_AUTO; - } - - if (board_config == AD1884A_AUTO) - return ad1884_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); - spec->multiout.dac_nids = ad1884a_dac_nids; - spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT; - spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids); - spec->adc_nids = ad1884a_adc_nids; - spec->capsrc_nids = ad1884a_capsrc_nids; - spec->input_mux = &ad1884a_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1884a_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1884a_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1884a_loopbacks; -#endif - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - case AD1884A_LAPTOP: - spec->mixers[0] = ad1884a_laptop_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event; - codec->patch_ops.init = ad1884a_laptop_init; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1884A_MOBILE: - spec->mixers[0] = ad1884a_mobile_mixers; - spec->init_verbs[0] = ad1884a_mobile_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; - codec->patch_ops.init = ad1884a_hp_init; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1884A_THINKPAD: - spec->mixers[0] = ad1984a_thinkpad_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1984a_thinkpad_verbs; - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1984a_thinkpad_capture_source; - codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; - codec->patch_ops.init = ad1984a_thinkpad_init; - break; - case AD1984A_PRECISION: - spec->mixers[0] = ad1984a_precision_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1984a_precision_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1984a_precision_unsol_event; - codec->patch_ops.init = ad1984a_precision_init; - break; - case AD1984A_TOUCHSMART: - spec->mixers[0] = ad1984a_touchsmart_mixers; - spec->init_verbs[0] = ad1984a_touchsmart_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; - codec->patch_ops.init = ad1984a_touchsmart_init; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1884a ad1884_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - /* * AD1882 / AD1882A * @@ -4844,299 +1058,7 @@ static int patch_ad1884a(struct hda_codec *codec) * port-G - rear clfe-out (6stack) */ -#ifdef ENABLE_AD_STATIC_QUIRKS -static const hda_nid_t ad1882_dac_nids[3] = { - 0x04, 0x03, 0x05 -}; - -static const hda_nid_t ad1882_adc_nids[2] = { - 0x08, 0x09, -}; - -static const hda_nid_t ad1882_capsrc_nids[2] = { - 0x0c, 0x0d, -}; - -#define AD1882_SPDIF_OUT 0x02 - -/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */ -static const struct hda_input_mux ad1882_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x1 }, - { "Mic", 0x4 }, - { "Line", 0x2 }, - { "CD", 0x3 }, - { "Mix", 0x7 }, - }, -}; - -/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */ -static const struct hda_input_mux ad1882a_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x1 }, - { "Mic", 0x4}, - { "Line", 0x2 }, - { "Digital Mic", 0x06 }, - { "Mix", 0x7 }, - }, -}; - -static const struct snd_kcontrol_new ad1882_base_mixers[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1882_loopback_mixers[] = { - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = { - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1882_3stack_mixers[] = { - HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = ad198x_ch_mode_info, - .get = ad198x_ch_mode_get, - .put = ad198x_ch_mode_put, - }, - { } /* end */ -}; - -/* simple auto-mute control for AD1882 3-stack board */ -#define AD1882_HP_EVENT 0x01 - -static void ad1882_3stack_automute(struct hda_codec *codec) -{ - bool mute = snd_hda_jack_detect(codec, 0x11); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - mute ? 0 : PIN_OUT); -} - -static int ad1882_3stack_automute_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1882_3stack_automute(codec); - return 0; -} - -static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case AD1882_HP_EVENT: - ad1882_3stack_automute(codec); - break; - } -} - -static const struct snd_kcontrol_new ad1882_6stack_mixers[] = { - HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb ad1882_ch2_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } /* end */ -}; - -static const struct hda_verb ad1882_ch4_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } /* end */ -}; - -static const struct hda_verb ad1882_ch6_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } /* end */ -}; - -static const struct hda_channel_mode ad1882_modes[3] = { - { 2, ad1882_ch2_init }, - { 4, ad1882_ch4_init }, - { 6, ad1882_ch6_init }, -}; - -/* - * initialization verbs - */ -static const struct hda_verb ad1882_init_verbs[] = { - /* DACs; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-A (HP) mixer */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* HP selector - select DAC2 */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Port-D (Line-out) mixer */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-D pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono-out mixer */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Mono-out pin */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-B (front mic) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ - /* Port-C (line-in) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ - /* Port-C mixer - mute as input */ - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Port-E (mic-in) pin */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ - /* Port-E mixer - mute as input */ - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Port-F (surround) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-G (CLFE) */ - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - /* SPDIF output selector */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } /* end */ -}; - -static const struct hda_verb ad1882_3stack_automute_verbs[] = { - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT}, - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1882_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Mic */ - { 0x20, HDA_INPUT, 4 }, /* Line */ - { 0x20, HDA_INPUT, 6 }, /* CD */ - { } /* end */ -}; -#endif - -/* models */ -enum { - AD1882_AUTO, - AD1882_3STACK, - AD1882_6STACK, - AD1882_3STACK_AUTOMUTE, - AD1882_MODELS -}; - -static const char * const ad1882_models[AD1986A_MODELS] = { - [AD1882_AUTO] = "auto", - [AD1882_3STACK] = "3stack", - [AD1882_6STACK] = "6stack", - [AD1882_3STACK_AUTOMUTE] = "3stack-automute", -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - -static int ad1882_parse_auto_config(struct hda_codec *codec) +static int patch_ad1882(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -5163,110 +1085,20 @@ static int ad1882_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1882(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1882_MODELS, - ad1882_models, NULL); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1882_AUTO; - } - - if (board_config == AD1882_AUTO) - return ad1882_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 6; - spec->multiout.num_dacs = 3; - spec->multiout.dac_nids = ad1882_dac_nids; - spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; - spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); - spec->adc_nids = ad1882_adc_nids; - spec->capsrc_nids = ad1882_capsrc_nids; - if (codec->vendor_id == 0x11d41882) - spec->input_mux = &ad1882_capture_source; - else - spec->input_mux = &ad1882a_capture_source; - spec->num_mixers = 2; - spec->mixers[0] = ad1882_base_mixers; - if (codec->vendor_id == 0x11d41882) - spec->mixers[1] = ad1882_loopback_mixers; - else - spec->mixers[1] = ad1882a_loopback_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1882_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1882_loopbacks; -#endif - spec->vmaster_nid = 0x04; - - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - default: - case AD1882_3STACK: - case AD1882_3STACK_AUTOMUTE: - spec->num_mixers = 3; - spec->mixers[2] = ad1882_3stack_mixers; - spec->channel_mode = ad1882_modes; - spec->num_channel_mode = ARRAY_SIZE(ad1882_modes); - spec->need_dac_fix = 1; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - if (board_config != AD1882_3STACK) { - spec->init_verbs[spec->num_init_verbs++] = - ad1882_3stack_automute_verbs; - codec->patch_ops.unsol_event = ad1882_3stack_unsol_event; - codec->patch_ops.init = ad1882_3stack_automute_init; - } - break; - case AD1882_6STACK: - spec->num_mixers = 3; - spec->mixers[2] = ad1882_6stack_mixers; - break; - } - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1882 ad1882_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * patch entries */ static const struct hda_codec_preset snd_hda_preset_analog[] = { - { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, + { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, - { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, + { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, - { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a }, - { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a }, + { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 }, + { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, - { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, + { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 90ff7a3..6e9876f 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -139,7 +139,7 @@ enum { #define DSP_SPEAKER_OUT_LATENCY 7 struct ct_effect { - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; hda_nid_t nid; int mid; /*effect module ID*/ int reqs[EFFECT_VALS_MAX_COUNT]; /*effect module request*/ @@ -270,7 +270,7 @@ enum { }; struct ct_tuning_ctl { - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; hda_nid_t parent_nid; hda_nid_t nid; int mid; /*effect module ID*/ @@ -3103,7 +3103,7 @@ static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, const char *name, int dir) { - char namestr[44]; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type); @@ -3935,7 +3935,7 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int dir) { - char namestr[44]; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index cccaf9c..b524f89 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -169,7 +169,7 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); - if (spec->gpio_eapd_hp) { + if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) { spec->gpio_data = spec->gen.hp_jack_present ? spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, @@ -291,10 +291,11 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; - /* init_verb sequence for C0/C1/C2 errata*/ - snd_hda_sequence_write(codec, cs_errata_init_verbs); - - snd_hda_sequence_write(codec, cs_coef_init_verbs); + if (spec->vendor_nid == CS420X_VENDOR_NID) { + /* init_verb sequence for C0/C1/C2 errata*/ + snd_hda_sequence_write(codec, cs_errata_init_verbs); + snd_hda_sequence_write(codec, cs_coef_init_verbs); + } snd_hda_gen_init(codec); @@ -307,8 +308,10 @@ static int cs_init(struct hda_codec *codec) spec->gpio_data); } - init_input_coef(codec); - init_digital_coef(codec); + if (spec->vendor_nid == CS420X_VENDOR_NID) { + init_input_coef(codec); + init_digital_coef(codec); + } return 0; } @@ -552,6 +555,76 @@ static int patch_cs420x(struct hda_codec *codec) } /* + * CS4208 support: + * Its layout is no longer compatible with CS4206/CS4207, and the generic + * parser seems working fairly well, except for trivial fixups. + */ +enum { + CS4208_GPIO0, +}; + +static const struct hda_model_fixup cs4208_models[] = { + { .id = CS4208_GPIO0, .name = "gpio0" }, + {} +}; + +static const struct snd_pci_quirk cs4208_fixup_tbl[] = { + /* codec SSID */ + SND_PCI_QUIRK(0x106b, 0x7100, "MacBookPro 6,1", CS4208_GPIO0), + SND_PCI_QUIRK(0x106b, 0x7200, "MacBookPro 6,2", CS4208_GPIO0), + {} /* terminator */ +}; + +static void cs4208_fixup_gpio0(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct cs_spec *spec = codec->spec; + spec->gpio_eapd_hp = 0; + spec->gpio_eapd_speaker = 1; + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; + } +} + +static const struct hda_fixup cs4208_fixups[] = { + [CS4208_GPIO0] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs4208_fixup_gpio0, + }, +}; + +static int patch_cs4208(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = cs_alloc_spec(codec, 0); /* no specific w/a */ + if (!spec) + return -ENOMEM; + + spec->gen.automute_hook = cs_automute; + + snd_hda_pick_fixup(codec, cs4208_models, cs4208_fixup_tbl, + cs4208_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + err = cs_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs_patch_ops; + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + + return 0; + + error: + cs_free(codec); + return err; +} + +/* * Cirrus Logic CS4210 * * 1 DAC => HP(sense) / Speakers, @@ -991,6 +1064,7 @@ static int patch_cs4213(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134208, .name = "CS4208", .patch = patch_cs4208 }, { .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 }, { .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 }, {} /* terminator */ @@ -998,6 +1072,7 @@ static const struct hda_codec_preset snd_hda_preset_cirrus[] = { MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134208"); MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_ALIAS("snd-hda-codec-id:10134213"); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b314d3e..4edd2d0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -66,6 +66,8 @@ struct conexant_spec { hda_nid_t eapds[4]; bool dynamic_eapd; + unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + #ifdef ENABLE_CXT_STATIC_QUIRKS const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -2947,7 +2949,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), {} @@ -3201,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec) snd_hda_gen_init(codec); if (!spec->dynamic_eapd) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + return 0; } @@ -3225,6 +3229,8 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_HEADPHONE_MIC_PIN, + CXT_FIXUP_HEADPHONE_MIC, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3247,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec, (0 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_update_headset_mode(struct hda_codec *codec) +{ + /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */ + int i; + bool mic_mode = false; + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == mux_pin) { + mic_mode = !!cfg->inputs[i].is_headphone_mic; + break; + } + + if (mic_mode) { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */ + spec->gen.hp_jack_present = false; + } else { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */ + spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]); + } + + snd_hda_gen_update_outputs(codec); +} + +static void cxt_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + cxt_update_headset_mode(codec); +} + +static void cxt_fixup_headphone_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; + spec->gen.automute_hook = cxt_update_headset_mode; + break; + case HDA_FIXUP_ACT_INIT: + cxt_update_headset_mode(codec); + break; + } +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -3303,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_HEADPHONE_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADPHONE_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { } + } + }, + [CXT_FIXUP_HEADPHONE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headphone_mic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3312,12 +3384,14 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), @@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, + spec->parse_flags); if (err < 0) goto error; @@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->bus->allow_bus_reset = 1; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..3d8cd044 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -67,6 +69,8 @@ struct hdmi_spec_per_pin { struct delayed_work work; struct snd_kcontrol *eld_ctl; int repoll_count; + bool setup; /* the stream has been set up by prepare callback */ + int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ @@ -551,6 +555,17 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } } + if (!ca) { + /* if there was no match, select the regular ALSA channel + * allocation with the matching number of channels */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels) { + ca = channel_allocations[i].ca_index; + break; + } + } + } + snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); @@ -868,18 +883,24 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - bool non_pcm, - struct snd_pcm_substream *substream) +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool non_pcm) { - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; - int channels = substream->runtime->channels; + int channels = per_pin->channels; struct hdmi_eld *eld; int ca; union audio_infoframe ai; + if (!channels) + return; + + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -959,6 +980,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pin_nid; int pin_idx; struct hda_jack_tbl *jack; + int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) @@ -967,8 +989,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, - "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, + "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); @@ -1018,13 +1040,18 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid) +static void haswell_verify_D0(struct hda_codec *codec, + hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; + + /* For Haswell, the converter 1/2 may keep in D3 state after bootup, + * thus pins could only choose converter 0 for use. Make sure the + * converters are in correct power state */ + if (!snd_hda_check_power_state(codec, cvt_nid, AC_PWRST_D0)) + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); - pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; - if (pwr != AC_PWRST_D0) { + if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D0)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); msleep(40); @@ -1032,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid) pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1067,8 +1075,8 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, pin_nid); + if (is_haswell(codec)) + haswell_verify_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1101,26 +1109,15 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return 0; } -/* - * HDA PCM callbacks - */ -static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int hdmi_choose_cvt(struct hda_codec *codec, + int pin_idx, int *cvt_id, int *mux_id) { struct hdmi_spec *spec = codec->spec; - struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, mux_idx = 0; struct hdmi_spec_per_pin *per_pin; - struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; + int cvt_idx, mux_idx = 0; - /* Validate hinfo */ - pin_idx = hinfo_to_pin_index(spec, hinfo); - if (snd_BUG_ON(pin_idx < 0)) - return -EINVAL; per_pin = get_pin(spec, pin_idx); - eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { @@ -1138,17 +1135,89 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, continue; break; } + /* No free converters */ if (cvt_idx == spec->num_cvts) return -ENODEV; + if (cvt_id) + *cvt_id = cvt_idx; + if (mux_id) + *mux_id = mux_idx; + + return 0; +} + +static void haswell_config_cvts(struct hda_codec *codec, + int pin_id, int mux_id) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; + int pin_idx, mux_idx; + int curr; + int err; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + per_pin = get_pin(spec, pin_idx); + + if (pin_idx == pin_id) + continue; + + curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + /* Choose another unused converter */ + if (curr == mux_id) { + err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); + if (err < 0) + return; + snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); + } + } +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int pin_idx, cvt_idx, mux_idx = 0; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + struct hdmi_spec_per_cvt *per_cvt = NULL; + int err; + + /* Validate hinfo */ + pin_idx = hinfo_to_pin_index(spec, hinfo); + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; + + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + if (err < 0) + return err; + + per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); + + /* configure unused pins to choose other converters */ + if (is_haswell(codec)) + haswell_config_cvts(codec, pin_idx, mux_idx); + snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1263,6 +1332,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld_changed = true; } if (update_eld) { + bool old_eld_valid = pin_eld->eld_valid; pin_eld->eld_valid = eld->eld_valid; eld_changed = pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, @@ -1272,6 +1342,14 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_size); pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + + /* Haswell-specific workaround: re-setup when the transcoder is + * changed during the stream playback + */ + if (is_haswell(codec) && + eld->eld_valid && !old_eld_valid && per_pin->setup) + hdmi_setup_audio_infoframe(codec, per_pin, + per_pin->non_pcm); } mutex_unlock(&pin_eld->lock); @@ -1311,7 +1389,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -1444,14 +1522,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hda_nid_t cvt_nid = hinfo->nid; struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); - hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); + per_pin->channels = substream->runtime->channels; + per_pin->setup = true; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); + hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1491,6 +1572,9 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); + + per_pin->setup = false; + per_pin->channels = 0; } return 0; @@ -1626,8 +1710,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) - hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm, - substream); + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); return 0; } @@ -1715,6 +1798,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; + + if (!codec->pcm_info[pin_idx].pcm) + break; err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); @@ -1798,12 +1884,33 @@ static void generic_hdmi_free(struct hda_codec *codec) kfree(spec); } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + generic_hdmi_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hdmi_present_sense(per_pin, 1); + } + return 0; +} +#endif + static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, .free = generic_hdmi_free, .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; @@ -1821,7 +1928,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } @@ -1892,7 +1998,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -1903,8 +2009,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); @@ -2536,6 +2644,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2588,6 +2697,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 403010c..bc07d36 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -37,6 +37,9 @@ #include "hda_jack.h" #include "hda_generic.h" +/* keep halting ALC5505 DSP, for power saving */ +#define HALT_REALTEK_ALC5505 + /* unsol event tags */ #define ALC_DCVOL_EVENT 0x08 @@ -115,6 +118,7 @@ struct alc_spec { int init_amp; int codec_variant; /* flag for other variants */ + bool has_alc5505_dsp; /* for PLL fix */ hda_nid_t pll_nid; @@ -278,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); msleep(200); + snd_hda_shutup_pins(codec); } /* generic EAPD initialization */ @@ -822,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); - snd_hda_shutup_pins(codec); + else + snd_hda_shutup_pins(codec); } #define alc_free snd_hda_gen_free @@ -1027,6 +1033,7 @@ enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, + ALC880_FIXUP_LG_LW25, ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, ALC880_FIXUP_TCL_S700, @@ -1085,6 +1092,14 @@ static const struct hda_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_LG_LW25] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0181344f }, /* line-in */ + { 0x1b, 0x0321403f }, /* headphone */ + { } + } + }, [ALC880_FIXUP_W810] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1337,6 +1352,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), /* Below is the copied entries from alc880_quirks.c. @@ -1839,8 +1855,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.no_primary_hp = 1; + spec->gen.no_multi_io = 1; + } } static const struct hda_fixup alc882_fixups[] = { @@ -2519,6 +2537,7 @@ enum { ALC269_TYPE_ALC269VD, ALC269_TYPE_ALC280, ALC269_TYPE_ALC282, + ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, }; @@ -2544,6 +2563,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: case ALC269_TYPE_ALC282: + case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: ssids = alc269_ssids; break; @@ -2569,18 +2589,185 @@ static void alc269_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return; - if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } + snd_hda_shutup_pins(codec); +} + +static void alc283_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) + return; + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ + /* Headphone capless set to high power mode */ + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + if (hp_pin_sense) + msleep(85); + /* Index 0x46 Combo jack auto switch control 2 */ + /* 3k pull low control for Headset jack. */ + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val & ~(3 << 12)); + /* Headphone capless set to normal mode */ + alc_write_coef_idx(codec, 0x43, 0x9614); +} + +static void alc283_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) { + alc269_shutup(codec); + return; + } + + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val | (3 << 12)); + + if (hp_pin_sense) + msleep(85); + snd_hda_shutup_pins(codec); + alc_write_coef_idx(codec, 0x43, 0x9614); +} + +static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, + unsigned int val) +{ + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_COEF_INDEX, index_reg >> 1); + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_PROC_COEF, val & 0xffff); /* LSB */ + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_PROC_COEF, val >> 16); /* MSB */ +} + +static int alc5505_coef_get(struct hda_codec *codec, unsigned int index_reg) +{ + unsigned int val; + + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_COEF_INDEX, index_reg >> 1); + val = snd_hda_codec_read(codec, 0x51, 0, AC_VERB_GET_PROC_COEF, 0) + & 0xffff; + val |= snd_hda_codec_read(codec, 0x51, 0, AC_VERB_GET_PROC_COEF, 0) + << 16; + return val; +} + +static void alc5505_dsp_halt(struct hda_codec *codec) +{ + unsigned int val; + + alc5505_coef_set(codec, 0x3000, 0x000c); /* DSP CPU stop */ + alc5505_coef_set(codec, 0x880c, 0x0008); /* DDR enter self refresh */ + alc5505_coef_set(codec, 0x61c0, 0x11110080); /* Clock control for PLL and CPU */ + alc5505_coef_set(codec, 0x6230, 0xfc0d4011); /* Disable Input OP */ + alc5505_coef_set(codec, 0x61b4, 0x040a2b03); /* Stop PLL2 */ + alc5505_coef_set(codec, 0x61b0, 0x00005b17); /* Stop PLL1 */ + alc5505_coef_set(codec, 0x61b8, 0x04133303); /* Stop PLL3 */ + val = alc5505_coef_get(codec, 0x6220); + alc5505_coef_set(codec, 0x6220, (val | 0x3000)); /* switch Ringbuffer clock to DBUS clock */ +} + +static void alc5505_dsp_back_from_halt(struct hda_codec *codec) +{ + alc5505_coef_set(codec, 0x61b8, 0x04133302); + alc5505_coef_set(codec, 0x61b0, 0x00005b16); + alc5505_coef_set(codec, 0x61b4, 0x040a2b02); + alc5505_coef_set(codec, 0x6230, 0xf80d4011); + alc5505_coef_set(codec, 0x6220, 0x2002010f); + alc5505_coef_set(codec, 0x880c, 0x00000004); +} + +static void alc5505_dsp_init(struct hda_codec *codec) +{ + unsigned int val; + + alc5505_dsp_halt(codec); + alc5505_dsp_back_from_halt(codec); + alc5505_coef_set(codec, 0x61b0, 0x5b14); /* PLL1 control */ + alc5505_coef_set(codec, 0x61b0, 0x5b16); + alc5505_coef_set(codec, 0x61b4, 0x04132b00); /* PLL2 control */ + alc5505_coef_set(codec, 0x61b4, 0x04132b02); + alc5505_coef_set(codec, 0x61b8, 0x041f3300); /* PLL3 control*/ + alc5505_coef_set(codec, 0x61b8, 0x041f3302); + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_CODEC_RESET, 0); /* Function reset */ + alc5505_coef_set(codec, 0x61b8, 0x041b3302); + alc5505_coef_set(codec, 0x61b8, 0x04173302); + alc5505_coef_set(codec, 0x61b8, 0x04163302); + alc5505_coef_set(codec, 0x8800, 0x348b328b); /* DRAM control */ + alc5505_coef_set(codec, 0x8808, 0x00020022); /* DRAM control */ + alc5505_coef_set(codec, 0x8818, 0x00000400); /* DRAM control */ + + val = alc5505_coef_get(codec, 0x6200) >> 16; /* Read revision ID */ + if (val <= 3) + alc5505_coef_set(codec, 0x6220, 0x2002010f); /* I/O PAD Configuration */ + else + alc5505_coef_set(codec, 0x6220, 0x6002018f); + + alc5505_coef_set(codec, 0x61ac, 0x055525f0); /**/ + alc5505_coef_set(codec, 0x61c0, 0x12230080); /* Clock control */ + alc5505_coef_set(codec, 0x61b4, 0x040e2b02); /* PLL2 control */ + alc5505_coef_set(codec, 0x61bc, 0x010234f8); /* OSC Control */ + alc5505_coef_set(codec, 0x880c, 0x00000004); /* DRAM Function control */ + alc5505_coef_set(codec, 0x880c, 0x00000003); + alc5505_coef_set(codec, 0x880c, 0x00000010); + +#ifdef HALT_REALTEK_ALC5505 + alc5505_dsp_halt(codec); +#endif } +#ifdef HALT_REALTEK_ALC5505 +#define alc5505_dsp_suspend(codec) /* NOP */ +#define alc5505_dsp_resume(codec) /* NOP */ +#else +#define alc5505_dsp_suspend(codec) alc5505_dsp_halt(codec) +#define alc5505_dsp_resume(codec) alc5505_dsp_back_from_halt(codec) +#endif + #ifdef CONFIG_PM +static int alc269_suspend(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec->has_alc5505_dsp) + alc5505_dsp_suspend(codec); + return alc_suspend(codec); +} + static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2603,7 +2790,11 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + if (spec->has_alc5505_dsp) + alc5505_dsp_resume(codec); + return 0; } #endif /* CONFIG_PM */ @@ -3143,6 +3334,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */ +static int find_ext_mic_pin(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + hda_nid_t nid; + unsigned int defcfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + nid = cfg->inputs[i].pin; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT) + continue; + return nid; + } + + return 0; +} + static void alc271_hp_gate_mic_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -3150,11 +3363,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PROBE) { - if (snd_BUG_ON(!spec->gen.am_entry[1].pin || - !spec->gen.autocfg.hp_pins[0])) + int mic_pin = find_ext_mic_pin(codec); + int hp_pin = spec->gen.autocfg.hp_pins[0]; + + if (snd_BUG_ON(!mic_pin || !hp_pin)) return; - snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin, - spec->gen.autocfg.hp_pins[0]); + snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin); } } @@ -3190,6 +3404,95 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, } } +static void alc283_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + + msleep(200); + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0; + + msleep(600); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc283_chromebook_caps(struct hda_codec *codec) +{ + snd_hda_override_wcaps(codec, 0x03, 0); +} + +static void alc283_fixup_chromebook(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + int val; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + alc283_chromebook_caps(codec); + spec->gen.hp_automute_hook = alc283_hp_automute_hook; + /* MIC2-VREF control */ + /* Set to manual mode */ + val = alc_read_coef_idx(codec, 0x06); + alc_write_coef_idx(codec, 0x06, val & ~0x000c); + break; + } +} + +/* mute tablet speaker pin (0x14) via dock plugging in addition */ +static void asus_tx300_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_gen_update_outputs(codec); + if (snd_hda_jack_detect(codec, 0x1b)) + spec->gen.mute_bits |= (1ULL << 0x14); +} + +static void alc282_fixup_asus_tx300(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + /* TX300 needs to set up GPIO2 for the speaker amp */ + static const struct hda_verb gpio2_verbs[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, + {} + }; + static const struct hda_pintbl dock_pins[] = { + { 0x1b, 0x21114000 }, /* dock speaker pin */ + {} + }; + struct snd_kcontrol *kctl; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, gpio2_verbs); + snd_hda_apply_pincfgs(codec, dock_pins); + spec->gen.auto_mute_via_amp = 1; + spec->gen.automute_hook = asus_tx300_automute; + snd_hda_jack_detect_enable_callback(codec, 0x1b, + HDA_GEN_HP_EVENT, + snd_hda_gen_hp_automute); + break; + case HDA_FIXUP_ACT_BUILD: + /* this is a bit tricky; give more sane names for the main + * (tablet) speaker and the dock speaker, respectively + */ + kctl = snd_hda_find_mixer_ctl(codec, "Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Dock Speaker Playback Switch"); + kctl = snd_hda_find_mixer_ctl(codec, "Bass Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Speaker Playback Switch"); + break; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3225,6 +3528,9 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_ORDISSIMO_EVE2, + ALC283_FIXUP_CHROME_BOOK, + ALC282_FIXUP_ASUS_TX300, }; static const struct hda_fixup alc269_fixups[] = { @@ -3467,11 +3773,33 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, }, + [ALC269VB_FIXUP_ORDISSIMO_EVE2] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x18, 0x03a11d20 }, /* mic */ + { 0x19, 0x411111f0 }, /* Unused bogus pin */ + { } + }, + }, + [ALC283_FIXUP_CHROME_BOOK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc283_fixup_chromebook, + }, + [ALC282_FIXUP_ASUS_TX300] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc282_fixup_asus_tx300, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), + SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3482,6 +3810,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3495,14 +3825,21 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), @@ -3520,11 +3857,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), - SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), - SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), - SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), - SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), @@ -3535,10 +3867,19 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 /* Below is a quirk table taken from the old code. @@ -3704,11 +4045,15 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0290: spec->codec_variant = ALC269_TYPE_ALC280; break; - case 0x10ec0233: case 0x10ec0282: - case 0x10ec0283: spec->codec_variant = ALC269_TYPE_ALC282; break; + case 0x10ec0233: + case 0x10ec0283: + spec->codec_variant = ALC269_TYPE_ALC283; + spec->shutup = alc283_shutup; + spec->init_hook = alc283_init; + break; case 0x10ec0284: case 0x10ec0292: spec->codec_variant = ALC269_TYPE_ALC284; @@ -3718,6 +4063,11 @@ static int patch_alc269(struct hda_codec *codec) break; } + if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { + spec->has_alc5505_dsp = true; + spec->init_hook = alc5505_dsp_init; + } + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) @@ -3728,9 +4078,11 @@ static int patch_alc269(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM + codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; #endif - spec->shutup = alc269_shutup; + if (!spec->shutup) + spec->shutup = alc269_shutup; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -4194,9 +4546,11 @@ static const struct hda_fixup alc662_fixups[] = { static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..fba0cef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_VERBS, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_DELL_BIOS_AMIC, STAC_DELL_BIOS_SPDIF, STAC_927X_DELL_DMIC, STAC_927X_VOLKNOB, @@ -417,9 +418,11 @@ static void stac_update_outputs(struct hda_codec *codec) val &= ~spec->eapd_mask; else val |= spec->eapd_mask; - if (spec->gpio_data != val) + if (spec->gpio_data != val) { + spec->gpio_data = val; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, val); + } } } @@ -2233,6 +2236,10 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Folio", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x1900, "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2000, + "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2100, + "HP", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389, @@ -2813,6 +2820,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), @@ -3224,10 +3232,8 @@ static const struct hda_fixup stac927x_fixups[] = { [STAC_DELL_BIOS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - /* configure the analog microphone on some laptops */ - { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ - { 0x0f, 0x0227011f }, + { 0x0f, 0x0221101f }, /* correct the front input jack as a mic */ { 0x0e, 0x02a79130 }, {} @@ -3235,6 +3241,16 @@ static const struct hda_fixup stac927x_fixups[] = { .chained = true, .chain_id = STAC_927X_DELL_DMIC, }, + [STAC_DELL_BIOS_AMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* configure the analog microphone on some laptops */ + { 0x0c, 0x90a79130 }, + {} + }, + .chained = true, + .chain_id = STAC_DELL_BIOS, + }, [STAC_DELL_BIOS_SPDIF] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3263,6 +3279,7 @@ static const struct hda_model_fixup stac927x_models[] = { { .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" }, { .id = STAC_DELL_3ST, .name = "dell-3stack" }, { .id = STAC_DELL_BIOS, .name = "dell-bios" }, + { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" }, { .id = STAC_927X_VOLKNOB, .name = "volknob" }, {} }; @@ -3608,20 +3625,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio; int i; /* override some hints */ stac_store_hints(codec); /* set up GPIO */ - gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. * otherwise, unsol event will turn it on/off dynamically */ if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); + spec->gpio_data |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); snd_hda_gen_init(codec); @@ -3707,14 +3722,6 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #endif #ifdef CONFIG_PM -static int stac_resume(struct hda_codec *codec) -{ - codec->patch_ops.init(codec); - snd_hda_codec_resume_amp(codec); - snd_hda_codec_resume_cache(codec); - return 0; -} - static int stac_suspend(struct hda_codec *codec) { stac_shutup(codec); @@ -3743,7 +3750,6 @@ static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg, } #else #define stac_suspend NULL -#define stac_resume NULL #define stac_set_power_state NULL #endif /* CONFIG_PM */ @@ -3755,7 +3761,6 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, - .resume = stac_resume, #endif .reboot_notify = stac_shutup, }; @@ -3921,6 +3926,7 @@ static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + spec->gpio_mask |= spec->eapd_mask; if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e524554..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } @@ -480,14 +480,9 @@ static int via_suspend(struct hda_codec *codec) struct via_spec *spec = codec->spec; vt1708_stop_hp_work(codec); - if (spec->codec_type == VT1802) { - /* Fix pop noise on headphones */ - int i; - for (i = 0; i < spec->gen.autocfg.hp_outs; i++) - snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i], - 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - 0x00); - } + /* Fix pop noise on headphones */ + if (spec->codec_type == VT1802) + snd_hda_shutup_pins(codec); return 0; } @@ -746,6 +741,8 @@ static int patch_vt1708(struct hda_codec *codec) /* don't support the input jack switching due to lack of unsol event */ /* (it may work with polling, though, but it needs testing) */ spec->gen.suppress_auto_mic = 1; + /* Some machines show the broken speaker mute */ + spec->gen.auto_mute_via_amp = 1; /* Add HP and CD pin config connect bit re-config action */ vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID); @@ -910,6 +907,8 @@ static const struct hda_verb vt1708S_init_verbs[] = { static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, int offset, int num_steps, int step_size) { + snd_hda_override_wcaps(codec, pin, + get_wcaps(codec, pin) | AC_WCAP_IN_AMP); snd_hda_override_amp_caps(codec, pin, HDA_INPUT, (offset << AC_AMPCAP_OFFSET_SHIFT) | (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 806407a..28ec872 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2807,7 +2807,6 @@ static void snd_ice1712_remove(struct pci_dev *pci) if (ice->card_info && ice->card_info->chip_exit) ice->card_info->chip_exit(ice); snd_card_free(card); - pci_set_drvdata(pci, NULL); } static struct pci_driver ice1712_driver = { diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ce70e7f..5004717 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2800,7 +2800,6 @@ static void snd_vt1724_remove(struct pci_dev *pci) if (ice->card_info && ice->card_info->chip_exit) ice->card_info->chip_exit(ice); snd_card_free(card); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b8fe405..59c8aae 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3364,7 +3364,6 @@ static int snd_intel8x0_probe(struct pci_dev *pci, static void snd_intel8x0_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver intel8x0_driver = { diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index fea09e8..3573c11 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1328,7 +1328,6 @@ static int snd_intel8x0m_probe(struct pci_dev *pci, static void snd_intel8x0m_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver intel8x0m_driver = { diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 43b4228..9cf9829 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2473,7 +2473,6 @@ snd_korg1212_probe(struct pci_dev *pci, static void snd_korg1212_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver korg1212_driver = { diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 322b638..7307d97 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -759,7 +759,6 @@ out_free: static void lola_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* PCI IDs */ diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 298bc9b..3230e57 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1139,7 +1139,6 @@ out_free: static void snd_lx6464es_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index c76ac14..d541736 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2775,7 +2775,6 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_m3_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver m3_driver = { diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 934dec9..1e0f6ee 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1377,7 +1377,6 @@ static int snd_mixart_probe(struct pci_dev *pci, static void snd_mixart_remove(struct pci_dev *pci) { snd_mixart_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver mixart_driver = { diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 6febedb..fe79fff 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1746,7 +1746,6 @@ static int snd_nm256_probe(struct pci_dev *pci, static void snd_nm256_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9562dc6..b0cb48a 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -722,7 +722,6 @@ EXPORT_SYMBOL(oxygen_pci_probe); void oxygen_pci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } EXPORT_SYMBOL(oxygen_pci_remove); diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b97384a..d379b28 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1691,7 +1691,6 @@ static int pcxhr_probe(struct pci_dev *pci, static void pcxhr_remove(struct pci_dev *pci) { pcxhr_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver pcxhr_driver = { diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 63c1c80..56cc891 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2066,7 +2066,6 @@ static void snd_riptide_joystick_remove(struct pci_dev *pci) if (gameport) { release_region(gameport->io, 8); gameport_unregister_port(gameport); - pci_set_drvdata(pci, NULL); } } #endif @@ -2179,7 +2178,6 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_card_riptide_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver driver = { diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 0ecd410..cc26346 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1981,7 +1981,6 @@ snd_rme32_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_rme32_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver rme32_driver = { diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 5fb88ac..bb9ebc5 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -28,6 +28,7 @@ #include <linux/interrupt.h> #include <linux/pci.h> #include <linux/module.h> +#include <linux/vmalloc.h> #include <sound/core.h> #include <sound/info.h> @@ -198,6 +199,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard."); #define RME96_AD1852_VOL_BITS 14 #define RME96_AD1855_VOL_BITS 10 +/* Defines for snd_rme96_trigger */ +#define RME96_TB_START_PLAYBACK 1 +#define RME96_TB_START_CAPTURE 2 +#define RME96_TB_STOP_PLAYBACK 4 +#define RME96_TB_STOP_CAPTURE 8 +#define RME96_TB_RESET_PLAYPOS 16 +#define RME96_TB_RESET_CAPTUREPOS 32 +#define RME96_TB_CLEAR_PLAYBACK_IRQ 64 +#define RME96_TB_CLEAR_CAPTURE_IRQ 128 +#define RME96_RESUME_PLAYBACK (RME96_TB_START_PLAYBACK) +#define RME96_RESUME_CAPTURE (RME96_TB_START_CAPTURE) +#define RME96_RESUME_BOTH (RME96_RESUME_PLAYBACK \ + | RME96_RESUME_CAPTURE) +#define RME96_START_PLAYBACK (RME96_TB_START_PLAYBACK \ + | RME96_TB_RESET_PLAYPOS) +#define RME96_START_CAPTURE (RME96_TB_START_CAPTURE \ + | RME96_TB_RESET_CAPTUREPOS) +#define RME96_START_BOTH (RME96_START_PLAYBACK \ + | RME96_START_CAPTURE) +#define RME96_STOP_PLAYBACK (RME96_TB_STOP_PLAYBACK \ + | RME96_TB_CLEAR_PLAYBACK_IRQ) +#define RME96_STOP_CAPTURE (RME96_TB_STOP_CAPTURE \ + | RME96_TB_CLEAR_CAPTURE_IRQ) +#define RME96_STOP_BOTH (RME96_STOP_PLAYBACK \ + | RME96_STOP_CAPTURE) struct rme96 { spinlock_t lock; @@ -214,6 +240,13 @@ struct rme96 { u8 rev; /* card revision number */ +#ifdef CONFIG_PM + u32 playback_pointer; + u32 capture_pointer; + void *playback_suspend_buffer; + void *capture_suspend_buffer; +#endif + struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -344,6 +377,8 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -373,6 +408,8 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -402,6 +439,8 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -427,6 +466,8 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -1045,54 +1086,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream, } static void -snd_rme96_playback_start(struct rme96 *rme96, - int from_pause) +snd_rme96_trigger(struct rme96 *rme96, + int op) { - if (!from_pause) { + if (op & RME96_TB_RESET_PLAYPOS) writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS); - } - - rme96->wcreg |= RME96_WCR_START; - writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); -} - -static void -snd_rme96_capture_start(struct rme96 *rme96, - int from_pause) -{ - if (!from_pause) { + if (op & RME96_TB_RESET_CAPTUREPOS) writel(0, rme96->iobase + RME96_IO_RESET_REC_POS); - } - - rme96->wcreg |= RME96_WCR_START_2; + if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) { + rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); + if (rme96->rcreg & RME96_RCR_IRQ) + writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ); + } + if (op & RME96_TB_CLEAR_CAPTURE_IRQ) { + rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); + if (rme96->rcreg & RME96_RCR_IRQ_2) + writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ); + } + if (op & RME96_TB_START_PLAYBACK) + rme96->wcreg |= RME96_WCR_START; + if (op & RME96_TB_STOP_PLAYBACK) + rme96->wcreg &= ~RME96_WCR_START; + if (op & RME96_TB_START_CAPTURE) + rme96->wcreg |= RME96_WCR_START_2; + if (op & RME96_TB_STOP_CAPTURE) + rme96->wcreg &= ~RME96_WCR_START_2; writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); } -static void -snd_rme96_playback_stop(struct rme96 *rme96) -{ - /* - * Check if there is an unconfirmed IRQ, if so confirm it, or else - * the hardware will not stop generating interrupts - */ - rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); - if (rme96->rcreg & RME96_RCR_IRQ) { - writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ); - } - rme96->wcreg &= ~RME96_WCR_START; - writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); -} -static void -snd_rme96_capture_stop(struct rme96 *rme96) -{ - rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); - if (rme96->rcreg & RME96_RCR_IRQ_2) { - writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ); - } - rme96->wcreg &= ~RME96_WCR_START_2; - writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); -} static irqreturn_t snd_rme96_interrupt(int irq, @@ -1155,6 +1177,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); spin_lock_irq(&rme96->lock); if (rme96->playback_substream != NULL) { spin_unlock_irq(&rme96->lock); @@ -1191,6 +1214,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); runtime->hw = snd_rme96_capture_spdif_info; if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG && (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0) @@ -1222,6 +1246,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); spin_lock_irq(&rme96->lock); if (rme96->playback_substream != NULL) { spin_unlock_irq(&rme96->lock); @@ -1253,6 +1278,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); runtime->hw = snd_rme96_capture_adat_info; if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) { /* makes no sense to use analog input. Note that analog @@ -1288,7 +1314,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISPLAYING(rme96)) { - snd_rme96_playback_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK); } rme96->playback_substream = NULL; rme96->playback_periodsize = 0; @@ -1309,7 +1335,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISRECORDING(rme96)) { - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_CAPTURE); } rme96->capture_substream = NULL; rme96->capture_periodsize = 0; @@ -1324,7 +1350,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISPLAYING(rme96)) { - snd_rme96_playback_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK); } writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS); spin_unlock_irq(&rme96->lock); @@ -1338,7 +1364,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISRECORDING(rme96)) { - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_CAPTURE); } writel(0, rme96->iobase + RME96_IO_RESET_REC_POS); spin_unlock_irq(&rme96->lock); @@ -1350,41 +1376,55 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); + struct snd_pcm_substream *s; + bool sync; + + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) == rme96) + snd_pcm_trigger_done(s, substream); + } + + sync = (rme96->playback_substream && rme96->capture_substream) && + (rme96->playback_substream->group == + rme96->capture_substream->group); switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (!RME96_ISPLAYING(rme96)) { - if (substream != rme96->playback_substream) { + if (substream != rme96->playback_substream) return -EBUSY; - } - snd_rme96_playback_start(rme96, 0); + snd_rme96_trigger(rme96, sync ? RME96_START_BOTH + : RME96_START_PLAYBACK); } break; + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: if (RME96_ISPLAYING(rme96)) { - if (substream != rme96->playback_substream) { + if (substream != rme96->playback_substream) return -EBUSY; - } - snd_rme96_playback_stop(rme96); + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_PLAYBACK); } break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (RME96_ISPLAYING(rme96)) { - snd_rme96_playback_stop(rme96); - } + if (RME96_ISPLAYING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_PLAYBACK); break; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!RME96_ISPLAYING(rme96)) { - snd_rme96_playback_start(rme96, 1); - } + if (!RME96_ISPLAYING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH + : RME96_RESUME_PLAYBACK); break; - + default: return -EINVAL; } + return 0; } @@ -1393,38 +1433,51 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); + struct snd_pcm_substream *s; + bool sync; + + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) == rme96) + snd_pcm_trigger_done(s, substream); + } + + sync = (rme96->playback_substream && rme96->capture_substream) && + (rme96->playback_substream->group == + rme96->capture_substream->group); switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (!RME96_ISRECORDING(rme96)) { - if (substream != rme96->capture_substream) { + if (substream != rme96->capture_substream) return -EBUSY; - } - snd_rme96_capture_start(rme96, 0); + snd_rme96_trigger(rme96, sync ? RME96_START_BOTH + : RME96_START_CAPTURE); } break; + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: if (RME96_ISRECORDING(rme96)) { - if (substream != rme96->capture_substream) { + if (substream != rme96->capture_substream) return -EBUSY; - } - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_CAPTURE); } break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (RME96_ISRECORDING(rme96)) { - snd_rme96_capture_stop(rme96); - } + if (RME96_ISRECORDING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_CAPTURE); break; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!RME96_ISRECORDING(rme96)) { - snd_rme96_capture_start(rme96, 1); - } + if (!RME96_ISRECORDING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH + : RME96_RESUME_CAPTURE); break; - + default: return -EINVAL; } @@ -1505,8 +1558,7 @@ snd_rme96_free(void *private_data) return; } if (rme96->irq >= 0) { - snd_rme96_playback_stop(rme96); - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_BOTH); rme96->areg &= ~RME96_AR_DAC_EN; writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); free_irq(rme96->irq, (void *)rme96); @@ -1520,6 +1572,10 @@ snd_rme96_free(void *private_data) pci_release_regions(rme96->pci); rme96->port = 0; } +#ifdef CONFIG_PM + vfree(rme96->playback_suspend_buffer); + vfree(rme96->capture_suspend_buffer); +#endif pci_disable_device(rme96->pci); } @@ -1606,8 +1662,7 @@ snd_rme96_create(struct rme96 *rme96) rme96->capture_periodsize = 0; /* make sure playback/capture is stopped, if by some reason active */ - snd_rme96_playback_stop(rme96); - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_BOTH); /* set default values in registers */ rme96->wcreg = @@ -2319,6 +2374,87 @@ snd_rme96_create_switches(struct snd_card *card, * Card initialisation */ +#ifdef CONFIG_PM + +static int +snd_rme96_suspend(struct pci_dev *pci, + pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct rme96 *rme96 = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend(rme96->playback_substream); + snd_pcm_suspend(rme96->capture_substream); + + /* save capture & playback pointers */ + rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS) + & RME96_RCR_AUDIO_ADDR_MASK; + rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS) + & RME96_RCR_AUDIO_ADDR_MASK; + + /* save playback and capture buffers */ + memcpy_fromio(rme96->playback_suspend_buffer, + rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE); + memcpy_fromio(rme96->capture_suspend_buffer, + rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE); + + /* disable the DAC */ + rme96->areg &= ~RME96_AR_DAC_EN; + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + + pci_disable_device(pci); + pci_save_state(pci); + + return 0; +} + +static int +snd_rme96_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct rme96 *rme96 = card->private_data; + + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n"); + snd_card_disconnect(card); + return -EIO; + } + + /* reset playback and record buffer pointers */ + writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS + + rme96->playback_pointer); + writel(0, rme96->iobase + RME96_IO_SET_REC_POS + + rme96->capture_pointer); + + /* restore playback and capture buffers */ + memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER, + rme96->playback_suspend_buffer, RME96_BUFFER_SIZE); + memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER, + rme96->capture_suspend_buffer, RME96_BUFFER_SIZE); + + /* reset the ADC */ + writel(rme96->areg | RME96_AR_PD2, + rme96->iobase + RME96_IO_ADDITIONAL_REG); + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + + /* reset and enable DAC, restore analog volume */ + snd_rme96_reset_dac(rme96); + rme96->areg |= RME96_AR_DAC_EN; + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + if (RME96_HAS_ANALOG_OUT(rme96)) { + usleep_range(3000, 10000); + snd_rme96_apply_dac_volume(rme96); + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} + +#endif + static void snd_rme96_card_free(struct snd_card *card) { snd_rme96_free(card->private_data); @@ -2355,6 +2491,23 @@ snd_rme96_probe(struct pci_dev *pci, return err; } +#ifdef CONFIG_PM + rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); + if (!rme96->playback_suspend_buffer) { + snd_printk(KERN_ERR + "Failed to allocate playback suspend buffer!\n"); + snd_card_free(card); + return -ENOMEM; + } + rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); + if (!rme96->capture_suspend_buffer) { + snd_printk(KERN_ERR + "Failed to allocate capture suspend buffer!\n"); + snd_card_free(card); + return -ENOMEM; + } +#endif + strcpy(card->driver, "Digi96"); switch (rme96->pci->device) { case PCI_DEVICE_ID_RME_DIGI96: @@ -2390,7 +2543,6 @@ snd_rme96_probe(struct pci_dev *pci, static void snd_rme96_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver rme96_driver = { @@ -2398,6 +2550,10 @@ static struct pci_driver rme96_driver = { .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = snd_rme96_remove, +#ifdef CONFIG_PM + .suspend = snd_rme96_suspend, + .resume = snd_rme96_resume, +#endif }; module_pci_driver(rme96_driver); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 94084cd..4f255df 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5412,7 +5412,6 @@ static int snd_hdsp_probe(struct pci_dev *pci, static void snd_hdsp_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver hdsp_driver = { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9ea05e9..3cde55b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -38,6 +38,97 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ + +/* ************* Register Documentation ******************************************************* + * + * Work in progress! Documentation is based on the code in this file. + * + * --------- HDSPM_controlRegister --------- + * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number + * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit + * : . : . : . : x . : HDSPM_AudioInterruptEnable \_ setting both bits + * : . : . : . : . x: HDSPM_Start / enables audio IO + * : . : . : . : x. : HDSPM_ClockModeMaster - 1: Master, 0: Slave + * : . : . : . : .210 : HDSPM_LatencyMask - 3 Bit value for latency + * : . : . : . : . : 0:64, 1:128, 2:256, 3:512, + * : . : . : . : . : 4:1024, 5:2048, 6:4096, 7:8192 + * :x . : . : . x:xx . : HDSPM_FrequencyMask + * : . : . : . :10 . : HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=?? + * : . : . : . x: . : <MADI> HDSPM_DoubleSpeed + * :x . : . : . : . : <MADI> HDSPM_QuadSpeed + * : . 3 : . 10: 2 . : . : HDSPM_SyncRefMask : + * : . : . x: . : . : HDSPM_SyncRef0 + * : . : . x : . : . : HDSPM_SyncRef1 + * : . : . : x . : . : <AES32> HDSPM_SyncRef2 + * : . x : . : . : . : <AES32> HDSPM_SyncRef3 + * : . : . 10: . : . : <MADI> sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn + * : . 3 : . 10: 2 . : . : <AES32> 0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn? + * : . x : . : . : . : <MADIe> HDSPe_FLOAT_FORMAT + * : . : . : x . : . : <MADI> HDSPM_InputSelect0 : 0=optical,1=coax + * : . : . :x . : . : <MADI> HDSPM_InputSelect1 + * : . : .x : . : . : <MADI> HDSPM_clr_tms + * : . : . : . x : . : <MADI> HDSPM_TX_64ch + * : . : . : . x : . : <AES32> HDSPM_Emphasis + * : . : . : .x : . : <MADI> HDSPM_AutoInp + * : . : . x : . : . : <MADI> HDSPM_SMUX + * : . : .x : . : . : <MADI> HDSPM_clr_tms + * : . : x. : . : . : <MADI> HDSPM_taxi_reset + * : . x: . : . : . : <MADI> HDSPM_LineOut + * : . x: . : . : . : <AES32> ?????????????????? + * : . : x. : . : . : <AES32> HDSPM_WCK48 + * : . : . : .x : . : <AES32> HDSPM_Dolby + * : . : x . : . : . : HDSPM_Midi0InterruptEnable + * : . :x . : . : . : HDSPM_Midi1InterruptEnable + * : . : x . : . : . : HDSPM_Midi2InterruptEnable + * : . x : . : . : . : <MADI> HDSPM_Midi3InterruptEnable + * : . x : . : . : . : <AES32> HDSPM_DS_DoubleWire + * : .x : . : . : . : <AES32> HDSPM_QS_DoubleWire + * : x. : . : . : . : <AES32> HDSPM_QS_QuadWire + * : . : . : . x : . : <AES32> HDSPM_Professional + * : x . : . : . : . : HDSPM_wclk_sel + * : . : . : . : . : + * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number + * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit + * + * + * + * AIO / RayDAT only + * + * ------------ HDSPM_WR_SETTINGS ---------- + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte + * :1098.7654:3210.9876:5432.1098:7654.3210: + * :||||.||||:||||.||||:||||.||||:||||.||||: bit number + * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit + * : . : . : . : . x: HDSPM_c0Master 1: Master, 0: Slave + * : . : . : . : . x : HDSPM_c0_SyncRef0 + * : . : . : . : . x : HDSPM_c0_SyncRef1 + * : . : . : . : .x : HDSPM_c0_SyncRef2 + * : . : . : . : x. : HDSPM_c0_SyncRef3 + * : . : . : . : 3.210 : HDSPM_c0_SyncRefMask: + * : . : . : . : . : RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4, + * : . : . : . : . : 9:TCO, 10:SyncIn + * : . : . : . : . : AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT, + * : . : . : . : . : 9:TCO, 10:SyncIn + * : . : . : . : . : + * : . : . : . : . : + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte + * :1098.7654:3210.9876:5432.1098:7654.3210: + * :||||.||||:||||.||||:||||.||||:||||.||||: bit number + * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit + * + */ #include <linux/init.h> #include <linux/delay.h> #include <linux/interrupt.h> @@ -95,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_controlRegister 64 #define HDSPM_interruptConfirmation 96 #define HDSPM_control2Reg 256 /* not in specs ???????? */ -#define HDSPM_freqReg 256 /* for AES32 */ +#define HDSPM_freqReg 256 /* for setting arbitrary clock values (DDS feature) */ #define HDSPM_midiDataOut0 352 /* just believe in old code */ #define HDSPM_midiDataOut1 356 #define HDSPM_eeprom_wr 384 /* for AES32 */ @@ -258,6 +349,25 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wclk_sel (1<<30) +/* additional control register bits for AIO*/ +#define HDSPM_c0_Wck48 0x20 /* also RayDAT */ +#define HDSPM_c0_Input0 0x1000 +#define HDSPM_c0_Input1 0x2000 +#define HDSPM_c0_Spdif_Opt 0x4000 +#define HDSPM_c0_Pro 0x8000 +#define HDSPM_c0_clr_tms 0x10000 +#define HDSPM_c0_AEB1 0x20000 +#define HDSPM_c0_AEB2 0x40000 +#define HDSPM_c0_LineOut 0x80000 +#define HDSPM_c0_AD_GAIN0 0x100000 +#define HDSPM_c0_AD_GAIN1 0x200000 +#define HDSPM_c0_DA_GAIN0 0x400000 +#define HDSPM_c0_DA_GAIN1 0x800000 +#define HDSPM_c0_PH_GAIN0 0x1000000 +#define HDSPM_c0_PH_GAIN1 0x2000000 +#define HDSPM_c0_Sym6db 0x4000000 + + /* --- bit helper defines */ #define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2) #define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|\ @@ -341,11 +451,11 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */ #define HDSPM_madiSync (1<<18) /* MADI is in sync */ -#define HDSPM_tcoLock 0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */ -#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status */ +#define HDSPM_tcoLockMadi 0x00000020 /* Optional TCO locked status for HDSPe MADI*/ +#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/ -#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */ -#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */ +#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */ +#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */ #define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */ /* since 64byte accurate, last 6 bits are not used */ @@ -363,7 +473,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); * Interrupt */ #define HDSPM_tco_detect 0x08000000 -#define HDSPM_tco_lock 0x20000000 +#define HDSPM_tcoLockAes 0x20000000 /* Optional TCO locked status for HDSPe AES */ #define HDSPM_s2_tco_detect 0x00000040 #define HDSPM_s2_AEBO_D 0x00000080 @@ -400,8 +510,8 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */ #define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */ -#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */ -/* missing Bit for 111=128, 1000=176.4, 1001=192 */ +#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, 111=128 */ +#define HDSPM_wc_freq3 0x800 /* 1000=176.4, 1001=192 */ #define HDSPM_SyncRef0 0x10000 /* Sync Reference */ #define HDSPM_SyncRef1 0x20000 @@ -412,13 +522,17 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync) -#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2|\ + HDSPM_wc_freq3) #define HDSPM_wcFreq32 (HDSPM_wc_freq0) #define HDSPM_wcFreq44_1 (HDSPM_wc_freq1) #define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1) #define HDSPM_wcFreq64 (HDSPM_wc_freq2) #define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2) #define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq128 (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq176_4 (HDSPM_wc_freq3) +#define HDSPM_wcFreq192 (HDSPM_wc_freq0|HDSPM_wc_freq3) #define HDSPM_status1_F_0 0x0400000 #define HDSPM_status1_F_1 0x0800000 @@ -457,7 +571,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_AES32_AUTOSYNC_FROM_AES6 6 #define HDSPM_AES32_AUTOSYNC_FROM_AES7 7 #define HDSPM_AES32_AUTOSYNC_FROM_AES8 8 -#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9 +#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9 +#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10 +#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11 /* status2 */ /* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */ @@ -533,36 +649,39 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); /* names for speed modes */ static char *hdspm_speed_names[] = { "single", "double", "quad" }; -static char *texts_autosync_aes_tco[] = { "Word Clock", +static const char *const texts_autosync_aes_tco[] = { "Word Clock", "AES1", "AES2", "AES3", "AES4", "AES5", "AES6", "AES7", "AES8", - "TCO" }; -static char *texts_autosync_aes[] = { "Word Clock", + "TCO", "Sync In" +}; +static const char *const texts_autosync_aes[] = { "Word Clock", "AES1", "AES2", "AES3", "AES4", - "AES5", "AES6", "AES7", "AES8" }; -static char *texts_autosync_madi_tco[] = { "Word Clock", + "AES5", "AES6", "AES7", "AES8", + "Sync In" +}; +static const char *const texts_autosync_madi_tco[] = { "Word Clock", "MADI", "TCO", "Sync In" }; -static char *texts_autosync_madi[] = { "Word Clock", +static const char *const texts_autosync_madi[] = { "Word Clock", "MADI", "Sync In" }; -static char *texts_autosync_raydat_tco[] = { +static const char *const texts_autosync_raydat_tco[] = { "Word Clock", "ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4", "AES", "SPDIF", "TCO", "Sync In" }; -static char *texts_autosync_raydat[] = { +static const char *const texts_autosync_raydat[] = { "Word Clock", "ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4", "AES", "SPDIF", "Sync In" }; -static char *texts_autosync_aio_tco[] = { +static const char *const texts_autosync_aio_tco[] = { "Word Clock", "ADAT", "AES", "SPDIF", "TCO", "Sync In" }; -static char *texts_autosync_aio[] = { "Word Clock", +static const char *const texts_autosync_aio[] = { "Word Clock", "ADAT", "AES", "SPDIF", "Sync In" }; -static char *texts_freq[] = { +static const char *const texts_freq[] = { "No Lock", "32 kHz", "44.1 kHz", @@ -625,7 +744,8 @@ static char *texts_ports_aio_in_ss[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", - "ADAT.7", "ADAT.8" + "ADAT.7", "ADAT.8", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ss[] = { @@ -634,14 +754,16 @@ static char *texts_ports_aio_out_ss[] = { "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", "ADAT.7", "ADAT.8", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_ds[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ds[] = { @@ -649,14 +771,16 @@ static char *texts_ports_aio_out_ds[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_qs[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_qs[] = { @@ -664,7 +788,8 @@ static char *texts_ports_aio_out_qs[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aes32[] = { @@ -741,8 +866,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in, */ 10, 11, /* spdif in */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ - -1, -1, - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -756,7 +881,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -769,7 +895,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 14, 16, 18, /* adat in */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -784,7 +911,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 14, 16, 18, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -798,7 +925,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 16, /* adat in */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -813,7 +941,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 16, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -852,11 +981,11 @@ struct hdspm_midi { }; struct hdspm_tco { - int input; - int framerate; - int wordclock; - int samplerate; - int pull; + int input; /* 0: LTC, 1:Video, 2: WC*/ + int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */ + int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */ + int samplerate; /* 0=44.1, 1=48, 2= freq from app */ + int pull; /* 0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/ int term; /* 0 = off, 1 = on */ }; @@ -875,7 +1004,7 @@ struct hdspm { u32 control_register; /* cached value */ u32 control2_register; /* cached value */ - u32 settings_register; + u32 settings_register; /* cached value for AIO / RayDat (sync reference, master/slave) */ struct hdspm_midi midi[4]; struct tasklet_struct midi_tasklet; @@ -937,7 +1066,7 @@ struct hdspm { struct hdspm_tco *tco; /* NULL if no TCO detected */ - char **texts_autosync; + const char *const *texts_autosync; int texts_autosync_items; cycles_t last_interrupt; @@ -972,12 +1101,24 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm); static inline int hdspm_get_pll_freq(struct hdspm *hdspm); static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm); static int hdspm_autosync_ref(struct hdspm *hdspm); +static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out); static int snd_hdspm_set_defaults(struct hdspm *hdspm); static int hdspm_system_clock_mode(struct hdspm *hdspm); static void hdspm_set_sgbuf(struct hdspm *hdspm, struct snd_pcm_substream *substream, unsigned int reg, int channels); +static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx); +static int hdspm_wc_sync_check(struct hdspm *hdspm); +static int hdspm_tco_sync_check(struct hdspm *hdspm); +static int hdspm_sync_in_sync_check(struct hdspm *hdspm); + +static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index); +static int hdspm_get_tco_sample_rate(struct hdspm *hdspm); +static int hdspm_get_wc_sample_rate(struct hdspm *hdspm); + + + static inline int HDSPM_bit2freq(int n) { static const int bit2freq_tab[] = { @@ -988,6 +1129,12 @@ static inline int HDSPM_bit2freq(int n) return bit2freq_tab[n]; } +static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm) +{ + return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type)); +} + + /* Write/read to/from HDSPM with Adresses in Bytes not words but only 32Bit writes are allowed */ @@ -1087,10 +1234,27 @@ static int hdspm_round_frequency(int rate) return 48000; } -static int hdspm_tco_sync_check(struct hdspm *hdspm); -static int hdspm_sync_in_sync_check(struct hdspm *hdspm); +/* QS and DS rates normally can not be detected + * automatically by the card. Only exception is MADI + * in 96k frame mode. + * + * So if we read SS values (32 .. 48k), check for + * user-provided DS/QS bits in the control register + * and multiply the base frequency accordingly. + */ +static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) +{ + if (rate <= 48000) { + if (hdspm->control_register & HDSPM_QuadSpeed) + return rate * 4; + else if (hdspm->control_register & + HDSPM_DoubleSpeed) + return rate * 2; + } + return rate; +} -/* check for external sample rate */ +/* check for external sample rate, returns the sample rate in Hz*/ static int hdspm_external_sample_rate(struct hdspm *hdspm) { unsigned int status, status2, timecode; @@ -1103,17 +1267,36 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); syncref = hdspm_autosync_ref(hdspm); + switch (syncref) { + case HDSPM_AES32_AUTOSYNC_FROM_WORD: + /* Check WC sync and get sample rate */ + if (hdspm_wc_sync_check(hdspm)) + return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm)); + break; - if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD && - status & HDSPM_AES32_wcLock) - return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF); + case HDSPM_AES32_AUTOSYNC_FROM_AES1: + case HDSPM_AES32_AUTOSYNC_FROM_AES2: + case HDSPM_AES32_AUTOSYNC_FROM_AES3: + case HDSPM_AES32_AUTOSYNC_FROM_AES4: + case HDSPM_AES32_AUTOSYNC_FROM_AES5: + case HDSPM_AES32_AUTOSYNC_FROM_AES6: + case HDSPM_AES32_AUTOSYNC_FROM_AES7: + case HDSPM_AES32_AUTOSYNC_FROM_AES8: + /* Check AES sync and get sample rate */ + if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)) + return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm, + syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)); + break; - if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 && - syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 && - status2 & (HDSPM_LockAES >> - (syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))) - return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF); - return 0; + + case HDSPM_AES32_AUTOSYNC_FROM_TCO: + /* Check TCO sync and get sample rate */ + if (hdspm_tco_sync_check(hdspm)) + return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm)); + break; + default: + return 0; + } /* end switch(syncref) */ break; case MADIface: @@ -1181,6 +1364,15 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) case HDSPM_wcFreq96: rate = 96000; break; + case HDSPM_wcFreq128: + rate = 128000; + break; + case HDSPM_wcFreq176_4: + rate = 176400; + break; + case HDSPM_wcFreq192: + rate = 192000; + break; default: rate = 0; break; @@ -1192,7 +1384,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) */ if (rate != 0 && (status2 & HDSPM_SelSyncRefMask) == HDSPM_SelSyncRef_WORD) - return rate; + return hdspm_rate_multiplier(hdspm, rate); /* maybe a madi input (which is taken if sel sync is madi) */ if (status & HDSPM_madiLock) { @@ -1255,21 +1447,8 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) } } - /* QS and DS rates normally can not be detected - * automatically by the card. Only exception is MADI - * in 96k frame mode. - * - * So if we read SS values (32 .. 48k), check for - * user-provided DS/QS bits in the control register - * and multiply the base frequency accordingly. - */ - if (rate <= 48000) { - if (hdspm->control_register & HDSPM_QuadSpeed) - rate *= 4; - else if (hdspm->control_register & - HDSPM_DoubleSpeed) - rate *= 2; - } + rate = hdspm_rate_multiplier(hdspm, rate); + break; } @@ -2109,6 +2288,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm) status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 16) & 0xF; break; + case AES32: + status = hdspm_read(hdspm, HDSPM_statusRegister); + return (status >> HDSPM_AES32_wcFreq_bit) & 0xF; default: break; } @@ -2132,6 +2314,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm) status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 20) & 0xF; break; + case AES32: + status = hdspm_read(hdspm, HDSPM_statusRegister); + return (status >> 1) & 0xF; default: break; } @@ -2163,6 +2348,23 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm) return 0; } +/** + * Returns the AES sample rate class for the given card. + **/ +static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index) +{ + int timecode; + + switch (hdspm->io_type) { + case AES32: + timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); + return (timecode >> (4*index)) & 0xF; + break; + default: + break; + } + return 0; +} /** * Returns the sample rate class for input source <idx> for @@ -2176,16 +2378,24 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx) } #define ENUMERATED_CTL_INFO(info, texts) \ -{ \ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \ - uinfo->count = 1; \ - uinfo->value.enumerated.items = ARRAY_SIZE(texts); \ - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \ - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; \ - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \ -} + snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts) +/* Helper function to query the external sample rate and return the + * corresponding enum to be returned to userspace. + */ +static int hdspm_external_rate_to_enum(struct hdspm *hdspm) +{ + int rate = hdspm_external_sample_rate(hdspm); + int i, selected_rate = 0; + for (i = 1; i < 10; i++) + if (HDSPM_bit2freq(i) == rate) { + selected_rate = i; + break; + } + return selected_rate; +} + #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -2250,7 +2460,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, default: ucontrol->value.enumerated.item[0] = hdspm_get_s1_sample_rate(hdspm, - ucontrol->id.index-1); + kcontrol->private_value-1); } break; @@ -2269,28 +2479,24 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[0] = hdspm_get_sync_in_sample_rate(hdspm); break; + case 11: /* External Rate */ + ucontrol->value.enumerated.item[0] = + hdspm_external_rate_to_enum(hdspm); + break; default: /* AES1 to AES8 */ ucontrol->value.enumerated.item[0] = - hdspm_get_s1_sample_rate(hdspm, - kcontrol->private_value-1); + hdspm_get_aes_sample_rate(hdspm, + kcontrol->private_value - + HDSPM_AES32_AUTOSYNC_FROM_AES1); break; } break; case MADI: case MADIface: - { - int rate = hdspm_external_sample_rate(hdspm); - int i, selected_rate = 0; - for (i = 1; i < 10; i++) - if (HDSPM_bit2freq(i) == rate) { - selected_rate = i; - break; - } - ucontrol->value.enumerated.item[0] = selected_rate; - } + ucontrol->value.enumerated.item[0] = + hdspm_external_rate_to_enum(hdspm); break; - default: break; } @@ -2339,33 +2545,17 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm) **/ static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode) { - switch (hdspm->io_type) { - case AIO: - case RayDAT: - if (0 == mode) - hdspm->settings_register |= HDSPM_c0Master; - else - hdspm->settings_register &= ~HDSPM_c0Master; - - hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register); - break; - - default: - if (0 == mode) - hdspm->control_register |= HDSPM_ClockModeMaster; - else - hdspm->control_register &= ~HDSPM_ClockModeMaster; - - hdspm_write(hdspm, HDSPM_controlRegister, - hdspm->control_register); - } + hdspm_set_toggle_setting(hdspm, + (hdspm_is_raydat_or_aio(hdspm)) ? + HDSPM_c0Master : HDSPM_ClockModeMaster, + (0 == mode)); } static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Master", "AutoSync" }; + static const char *const texts[] = { "Master", "AutoSync" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -2789,16 +2979,7 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = hdspm->texts_autosync_items; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - hdspm->texts_autosync[uinfo->value.enumerated.item]); + snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync); return 0; } @@ -2853,19 +3034,20 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol, static int hdspm_autosync_ref(struct hdspm *hdspm) { + /* This looks at the autosync selected sync reference */ if (AES32 == hdspm->io_type) { + unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister); - unsigned int syncref = - (status >> HDSPM_AES32_syncref_bit) & 0xF; - if (syncref == 0) - return HDSPM_AES32_AUTOSYNC_FROM_WORD; - if (syncref <= 8) + unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF; + if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) && + (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) { return syncref; + } return HDSPM_AES32_AUTOSYNC_FROM_NONE; + } else if (MADI == hdspm->io_type) { - /* This looks at the autosync selected sync reference */ - unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2); + unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2); switch (status2 & HDSPM_SelSyncRefMask) { case HDSPM_SelSyncRef_WORD: return HDSPM_AUTOSYNC_FROM_WORD; @@ -2878,7 +3060,7 @@ static int hdspm_autosync_ref(struct hdspm *hdspm) case HDSPM_SelSyncRef_NVALID: return HDSPM_AUTOSYNC_FROM_NONE; default: - return 0; + return HDSPM_AUTOSYNC_FROM_NONE; } } @@ -2892,31 +3074,15 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol, struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); if (AES32 == hdspm->io_type) { - static char *texts[] = { "WordClock", "AES1", "AES2", "AES3", - "AES4", "AES5", "AES6", "AES7", "AES8", "None"}; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 10; - if (uinfo->value.enumerated.item >= - uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3", + "AES4", "AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"}; + + ENUMERATED_CTL_INFO(uinfo, texts); } else if (MADI == hdspm->io_type) { - static char *texts[] = {"Word Clock", "MADI", "TCO", + static const char *const texts[] = {"Word Clock", "MADI", "TCO", "Sync In", "None" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item >= - uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + ENUMERATED_CTL_INFO(uinfo, texts); } return 0; } @@ -2944,7 +3110,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No video", "NTSC", "PAL"}; + static const char *const texts[] = {"No video", "NTSC", "PAL"}; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -2990,7 +3156,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps", + static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps", "30 fps"}; ENUMERATED_CTL_INFO(uinfo, texts); return 0; @@ -3007,19 +3173,19 @@ static int hdspm_tco_ltc_frames(struct hdspm *hdspm) HDSPM_TCO1_LTC_Format_MSB)) { case 0: /* 24 fps */ - ret = 1; + ret = fps_24; break; case HDSPM_TCO1_LTC_Format_LSB: /* 25 fps */ - ret = 2; + ret = fps_25; break; case HDSPM_TCO1_LTC_Format_MSB: - /* 25 fps */ - ret = 3; + /* 29.97 fps */ + ret = fps_2997; break; default: /* 30 fps */ - ret = 4; + ret = fps_30; break; } } @@ -3047,16 +3213,35 @@ static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol, static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask) { - return (hdspm->control_register & regmask) ? 1 : 0; + u32 reg; + + if (hdspm_is_raydat_or_aio(hdspm)) + reg = hdspm->settings_register; + else + reg = hdspm->control_register; + + return (reg & regmask) ? 1 : 0; } static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out) { + u32 *reg; + u32 target_reg; + + if (hdspm_is_raydat_or_aio(hdspm)) { + reg = &(hdspm->settings_register); + target_reg = HDSPM_WR_SETTINGS; + } else { + reg = &(hdspm->control_register); + target_reg = HDSPM_controlRegister; + } + if (out) - hdspm->control_register |= regmask; + *reg |= regmask; else - hdspm->control_register &= ~regmask; - hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register); + *reg &= ~regmask; + + hdspm_write(hdspm, target_reg, *reg); return 0; } @@ -3121,7 +3306,7 @@ static int hdspm_set_input_select(struct hdspm * hdspm, int out) static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "optical", "coaxial" }; + static const char *const texts[] = { "optical", "coaxial" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3183,7 +3368,7 @@ static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds) static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Single", "Double" }; + static const char *const texts[] = { "Single", "Double" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3256,7 +3441,7 @@ static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode) static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Single", "Double", "Quad" }; + static const char *const texts[] = { "Single", "Double", "Quad" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3293,6 +3478,84 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol, return change; } +#define HDSPM_CONTROL_TRISTATE(xname, xindex) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .private_value = xindex, \ + .info = snd_hdspm_info_tristate, \ + .get = snd_hdspm_get_tristate, \ + .put = snd_hdspm_put_tristate \ +} + +static int hdspm_tristate(struct hdspm *hdspm, u32 regmask) +{ + u32 reg = hdspm->settings_register & (regmask * 3); + return reg / regmask; +} + +static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask) +{ + hdspm->settings_register &= ~(regmask * 3); + hdspm->settings_register |= (regmask * mode); + hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register); + + return 0; +} + +static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + u32 regmask = kcontrol->private_value; + + static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" }; + static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" }; + + switch (regmask) { + case HDSPM_c0_Input0: + ENUMERATED_CTL_INFO(uinfo, texts_spdif); + break; + default: + ENUMERATED_CTL_INFO(uinfo, texts_levels); + break; + } + return 0; +} + +static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + u32 regmask = kcontrol->private_value; + + spin_lock_irq(&hdspm->lock); + ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask); + spin_unlock_irq(&hdspm->lock); + return 0; +} + +static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + u32 regmask = kcontrol->private_value; + int change; + int val; + + if (!snd_hdspm_use_is_exclusive(hdspm)) + return -EBUSY; + val = ucontrol->value.integer.value[0]; + if (val < 0) + val = 0; + if (val > 2) + val = 2; + + spin_lock_irq(&hdspm->lock); + change = val != hdspm_tristate(hdspm, regmask); + hdspm_set_tristate(hdspm, val, regmask); + spin_unlock_irq(&hdspm->lock); + return change; +} + #define HDSPM_MADI_SPEEDMODE(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ @@ -3332,7 +3595,7 @@ static int hdspm_set_madi_speedmode(struct hdspm *hdspm, int mode) static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Single", "Double", "Quad" }; + static const char *const texts[] = { "Single", "Double", "Quad" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3567,7 +3830,7 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" }; + static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3575,7 +3838,7 @@ static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol, static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "No Lock", "Lock" }; + static const char *const texts[] = { "No Lock", "Lock" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3725,9 +3988,18 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) if (hdspm->tco) { switch (hdspm->io_type) { case MADI: + status = hdspm_read(hdspm, HDSPM_statusRegister); + if (status & HDSPM_tcoLockMadi) { + if (status & HDSPM_tcoSync) + return 2; + else + return 1; + } + return 0; + break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_tcoLock) { + if (status & HDSPM_tcoLockAes) { if (status & HDSPM_tcoSync) return 2; else @@ -3787,7 +4059,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 5: /* SYNC IN */ val = hdspm_sync_in_sync_check(hdspm); break; default: - val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); + val = hdspm_s1_sync_check(hdspm, + kcontrol->private_value-1); } break; @@ -3955,7 +4228,8 @@ static void hdspm_tco_write(struct hdspm *hdspm) static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "44.1 kHz", "48 kHz" }; + /* TODO freq from app could be supported here, see tco->samplerate */ + static const char *const texts[] = { "44.1 kHz", "48 kHz" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4001,7 +4275,8 @@ static int snd_hdspm_put_tco_sample_rate(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" }; + static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %", + "+ 4 %", "- 4 %" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4046,7 +4321,7 @@ static int snd_hdspm_put_tco_pull(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" }; + static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4092,7 +4367,7 @@ static int snd_hdspm_put_tco_wck_conversion(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "24 fps", "25 fps", "29.97fps", + static const char *const texts[] = { "24 fps", "25 fps", "29.97fps", "29.97 dfps", "30 fps", "30 dfps" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; @@ -4139,7 +4414,7 @@ static int snd_hdspm_put_tco_frame_rate(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "LTC", "Video", "WCK" }; + static const char *const texts[] = { "LTC", "Video", "WCK" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4264,7 +4539,6 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = { HDSPM_INTERNAL_CLOCK("Internal Clock", 0), HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0), HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0), - HDSPM_AUTOSYNC_REF("AutoSync Reference", 0), HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0), HDSPM_SYNC_CHECK("WC SyncCheck", 0), @@ -4278,7 +4552,16 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = { HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2), HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3), HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4), - HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5) + HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5), + HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0), + HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt), + HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro), + HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1), + HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db), + HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48), + HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0), + HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0), + HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0) /* HDSPM_INPUT_SELECT("Input Select", 0), @@ -4315,7 +4598,9 @@ static struct snd_kcontrol_new snd_hdspm_controls_raydat[] = { HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5), HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6), HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7), - HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8) + HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8), + HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro), + HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48) }; static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = { @@ -4325,7 +4610,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = { HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0), HDSPM_AUTOSYNC_REF("AutoSync Reference", 0), HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), - HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0), + HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11), HDSPM_SYNC_CHECK("WC Sync Check", 0), HDSPM_SYNC_CHECK("AES1 Sync Check", 1), HDSPM_SYNC_CHECK("AES2 Sync Check", 2), @@ -4481,77 +4766,22 @@ static int snd_hdspm_create_controls(struct snd_card *card, ------------------------------------------------------------*/ static void -snd_hdspm_proc_read_madi(struct snd_info_entry * entry, - struct snd_info_buffer *buffer) +snd_hdspm_proc_read_tco(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status, status2, control, freq; - - char *pref_sync_ref; - char *autosync_ref; - char *system_clock_mode; - char *insel; - int x, x2; - - /* TCO stuff */ + unsigned int status, control; int a, ltc, frames, seconds, minutes, hours; unsigned int period; u64 freq_const = 0; u32 rate; + snd_iprintf(buffer, "--- TCO ---\n"); + status = hdspm_read(hdspm, HDSPM_statusRegister); - status2 = hdspm_read(hdspm, HDSPM_statusRegister2); control = hdspm->control_register; - freq = hdspm_read(hdspm, HDSPM_timecodeRegister); - snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", - hdspm->card_name, hdspm->card->number + 1, - hdspm->firmware_rev, - (status2 & HDSPM_version0) | - (status2 & HDSPM_version1) | (status2 & - HDSPM_version2)); - - snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n", - (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF, - hdspm->serial); - - snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", - hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase); - - snd_iprintf(buffer, "--- System ---\n"); - snd_iprintf(buffer, - "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n", - status & HDSPM_audioIRQPending, - (status & HDSPM_midi0IRQPending) ? 1 : 0, - (status & HDSPM_midi1IRQPending) ? 1 : 0, - hdspm->irq_count); - snd_iprintf(buffer, - "HW pointer: id = %d, rawptr = %d (%d->%d) " - "estimated= %ld (bytes)\n", - ((status & HDSPM_BufferID) ? 1 : 0), - (status & HDSPM_BufferPositionMask), - (status & HDSPM_BufferPositionMask) % - (2 * (int)hdspm->period_bytes), - ((status & HDSPM_BufferPositionMask) - 64) % - (2 * (int)hdspm->period_bytes), - (long) hdspm_hw_pointer(hdspm) * 4); - - snd_iprintf(buffer, - "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n", - hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF); - snd_iprintf(buffer, - "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n", - hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF); - snd_iprintf(buffer, - "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, " - "status2=0x%x\n", - hdspm->control_register, hdspm->control2_register, - status, status2); if (status & HDSPM_tco_detect) { snd_iprintf(buffer, "TCO module detected.\n"); a = hdspm_read(hdspm, HDSPM_RD_TCO+4); @@ -4645,6 +4875,75 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, } else { snd_iprintf(buffer, "No TCO module detected.\n"); } +} + +static void +snd_hdspm_proc_read_madi(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdspm *hdspm = entry->private_data; + unsigned int status, status2, control, freq; + + char *pref_sync_ref; + char *autosync_ref; + char *system_clock_mode; + char *insel; + int x, x2; + + status = hdspm_read(hdspm, HDSPM_statusRegister); + status2 = hdspm_read(hdspm, HDSPM_statusRegister2); + control = hdspm->control_register; + freq = hdspm_read(hdspm, HDSPM_timecodeRegister); + + snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", + hdspm->card_name, hdspm->card->number + 1, + hdspm->firmware_rev, + (status2 & HDSPM_version0) | + (status2 & HDSPM_version1) | (status2 & + HDSPM_version2)); + + snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n", + (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF, + hdspm->serial); + + snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", + hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase); + + snd_iprintf(buffer, "--- System ---\n"); + + snd_iprintf(buffer, + "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n", + status & HDSPM_audioIRQPending, + (status & HDSPM_midi0IRQPending) ? 1 : 0, + (status & HDSPM_midi1IRQPending) ? 1 : 0, + hdspm->irq_count); + snd_iprintf(buffer, + "HW pointer: id = %d, rawptr = %d (%d->%d) " + "estimated= %ld (bytes)\n", + ((status & HDSPM_BufferID) ? 1 : 0), + (status & HDSPM_BufferPositionMask), + (status & HDSPM_BufferPositionMask) % + (2 * (int)hdspm->period_bytes), + ((status & HDSPM_BufferPositionMask) - 64) % + (2 * (int)hdspm->period_bytes), + (long) hdspm_hw_pointer(hdspm) * 4); + + snd_iprintf(buffer, + "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n", + hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF); + snd_iprintf(buffer, + "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n", + hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF); + snd_iprintf(buffer, + "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, " + "status2=0x%x\n", + hdspm->control_register, hdspm->control2_register, + status, status2); + snd_iprintf(buffer, "--- Settings ---\n"); @@ -4748,6 +5047,9 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, (status & HDSPM_RX_64ch) ? "64 channels" : "56 channels"); + /* call readout function for TCO specific status */ + snd_hdspm_proc_read_tco(entry, buffer); + snd_iprintf(buffer, "\n"); } @@ -4889,11 +5191,18 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, autosync_ref = "AES7"; break; case HDSPM_AES32_AUTOSYNC_FROM_AES8: autosync_ref = "AES8"; break; + case HDSPM_AES32_AUTOSYNC_FROM_TCO: + autosync_ref = "TCO"; break; + case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN: + autosync_ref = "Sync In"; break; default: autosync_ref = "---"; break; } snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref); + /* call readout function for TCO specific status */ + snd_hdspm_proc_read_tco(entry, buffer); + snd_iprintf(buffer, "\n"); } @@ -5077,7 +5386,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) case AES32: hdspm->control_register = - HDSPM_ClockModeMaster | /* Master Cloack Mode on */ + HDSPM_ClockModeMaster | /* Master Clock Mode on */ hdspm_encode_latency(7) | /* latency max=8192samples */ HDSPM_SyncRef0 | /* AES1 is syncclock */ HDSPM_LineOut | /* Analog output in */ @@ -5103,9 +5412,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) all_in_all_mixer(hdspm, 0 * UNITY_GAIN); - if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) { + if (hdspm_is_raydat_or_aio(hdspm)) hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register); - } /* set a default rate so that the channel map is set up. */ hdspm_set_rate(hdspm, 48000, 1); @@ -5351,6 +5659,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream, */ + /* For AES cards, the float format bit is the same as the + * preferred sync reference. Since we don't want to break + * sync settings, we have to skip the remaining part of this + * function. + */ + if (hdspm->io_type == AES32) { + return 0; + } + + /* Switch to native float format if requested */ if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) { if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT)) @@ -5993,7 +6311,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, ltc.format = fps_2997; break; default: - ltc.format = 30; + ltc.format = fps_30; break; } if (i & HDSPM_TCO1_set_drop_frame_flag) { @@ -6459,10 +6777,6 @@ static int snd_hdspm_create(struct snd_card *card, break; case AIO: - if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { - snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n"); - } - hdspm->ss_in_channels = AIO_IN_SS_CHANNELS; hdspm->ds_in_channels = AIO_IN_DS_CHANNELS; hdspm->qs_in_channels = AIO_IN_QS_CHANNELS; @@ -6470,6 +6784,20 @@ static int snd_hdspm_create(struct snd_card *card, hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS; hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS; + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { + snd_printk(KERN_INFO "HDSPM: AEB input board found\n"); + hdspm->ss_in_channels += 4; + hdspm->ds_in_channels += 4; + hdspm->qs_in_channels += 4; + } + + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) { + snd_printk(KERN_INFO "HDSPM: AEB output board found\n"); + hdspm->ss_out_channels += 4; + hdspm->ds_out_channels += 4; + hdspm->qs_out_channels += 4; + } + hdspm->channel_map_out_ss = channel_map_aio_out_ss; hdspm->channel_map_out_ds = channel_map_aio_out_ds; hdspm->channel_map_out_qs = channel_map_aio_out_qs; @@ -6538,6 +6866,7 @@ static int snd_hdspm_create(struct snd_card *card, break; case MADI: + case AES32: if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) { hdspm->midiPorts++; hdspm->tco = kzalloc(sizeof(struct hdspm_tco), @@ -6545,7 +6874,7 @@ static int snd_hdspm_create(struct snd_card *card, if (NULL != hdspm->tco) { hdspm_tco_write(hdspm); } - snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n"); + snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n"); } else { hdspm->tco = NULL; } @@ -6560,10 +6889,12 @@ static int snd_hdspm_create(struct snd_card *card, case AES32: if (hdspm->tco) { hdspm->texts_autosync = texts_autosync_aes_tco; - hdspm->texts_autosync_items = 10; + hdspm->texts_autosync_items = + ARRAY_SIZE(texts_autosync_aes_tco); } else { hdspm->texts_autosync = texts_autosync_aes; - hdspm->texts_autosync_items = 9; + hdspm->texts_autosync_items = + ARRAY_SIZE(texts_autosync_aes); } break; @@ -6737,7 +7068,6 @@ static int snd_hdspm_probe(struct pci_dev *pci, static void snd_hdspm_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver hdspm_driver = { diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 773a67f..b96d9e1 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2628,7 +2628,6 @@ static int snd_rme9652_probe(struct pci_dev *pci, static void snd_rme9652_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver rme9652_driver = { diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 748e82d..e413b4e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1482,7 +1482,6 @@ error_out: static void snd_sis7019_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver sis7019_driver = { diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index a2e7686..2a46bf9 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1528,7 +1528,6 @@ static int snd_sonic_probe(struct pci_dev *pci, static void snd_sonic_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver sonicvibes_driver = { diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 1aefd62..b3b588b 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -169,7 +169,6 @@ static int snd_trident_probe(struct pci_dev *pci, static void snd_trident_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver trident_driver = { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index d756a35..5ae6f04 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1940,7 +1940,7 @@ static int snd_via686_create_gameport(struct via82xx *chip, unsigned char *legac r = request_region(JOYSTICK_ADDR, 8, "VIA686 gameport"); if (!r) { - printk(KERN_WARNING "via82xx: cannot reserve joystick port 0x%#x\n", + printk(KERN_WARNING "via82xx: cannot reserve joystick port %#x\n", JOYSTICK_ADDR); return -EBUSY; } @@ -2646,7 +2646,6 @@ static int snd_via82xx_probe(struct pci_dev *pci, static void snd_via82xx_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver via82xx_driver = { diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 4f5fd80..ca19028 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1227,7 +1227,6 @@ static int snd_via82xx_probe(struct pci_dev *pci, static void snd_via82xx_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver via82xx_modem_driver = { diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index e2f1ab3..ab8a9b1 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -254,7 +254,6 @@ static int snd_vx222_probe(struct pci_dev *pci, static void snd_vx222_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 01c4965..e8932b2 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -347,7 +347,6 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, static void snd_card_ymfpci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver ymfpci_driver = { diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 22056c5..d591c15 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2258,7 +2258,7 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) /* FIXME: temporarily disabled, otherwise we cannot fire up * the chip again unless reboot. ACPI bug? */ - pci_set_power_state(chip->pci, 3); + pci_set_power_state(chip->pci, PCI_D3hot); #endif #ifdef CONFIG_PM_SLEEP diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 09fc848..8abb521 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -139,7 +139,6 @@ __error: static int snd_pmac_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/sh/aica.c b/sound/sh/aica.c index e59a73a..78a3697 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -598,7 +598,6 @@ static int snd_aica_remove(struct platform_device *devptr) return -ENODEV; snd_card_free(dreamcastcard->card); kfree(dreamcastcard); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index e68c4fc..7c9422c 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -290,8 +290,6 @@ static int snd_sh_dac_pcm(struct snd_sh_dac *chip, int device) static int snd_sh_dac_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; } diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 9e675c7..5138b84 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,12 +26,9 @@ if SND_SOC config SND_SOC_AC97_BUS bool -config SND_SOC_DMAENGINE_PCM - bool - config SND_SOC_GENERIC_DMAENGINE_PCM bool - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" @@ -51,6 +48,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 197b6ae..61a64d2 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,10 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o -ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) -snd-soc-core-objs += soc-dmaengine-pcm.o -endif - ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o endif @@ -29,6 +25,7 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 3fdd87f..e48d38a 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -13,6 +13,7 @@ config SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_DMA tristate depends on SND_ATMEL_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC tristate @@ -32,6 +33,26 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + +config SND_AT91_SOC_SAM9X5_WM8731 + tristate "SoC Audio support for WM8731-based at91sam9x5 board" + depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5 + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8731 + help + Say Y if you want to add support for audio SoC on an + at91sam9x5 based board that is using WM8731 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967cc..5baabc8 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,10 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o +snd-atmel-soc-wm8904-objs := atmel_wm8904.o +snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o +obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 1d38fd0..06082e5 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -81,7 +81,9 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); @@ -89,138 +91,52 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, } } -/*--------------------------------------------------------------------------*\ - * DMAENGINE operations -\*--------------------------------------------------------------------------*/ -static bool filter(struct dma_chan *chan, void *slave) -{ - struct at_dma_slave *sl = slave; - - if (sl->dma_dev == chan->device->dev) { - chan->private = sl; - return true; - } else { - return false; - } -} - static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd) + struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pcm_dma_params *prtd; struct ssc_device *ssc; - struct dma_chan *dma_chan; - struct dma_slave_config slave_config; int ret; + prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ssc = prtd->ssc; - ret = snd_hwparams_to_dma_slave_config(substream, params, - &slave_config); + ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) { pr_err("atmel-pcm: hwparams to dma slave configure failed\n"); return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = (dma_addr_t)ssc->phybase + SSC_THR; - slave_config.dst_maxburst = 1; + slave_config->dst_addr = ssc->phybase + SSC_THR; + slave_config->dst_maxburst = 1; } else { - slave_config.src_addr = (dma_addr_t)ssc->phybase + SSC_RHR; - slave_config.src_maxburst = 1; - } - - dma_chan = snd_dmaengine_pcm_get_chan(substream); - if (dmaengine_slave_config(dma_chan, &slave_config)) { - pr_err("atmel-pcm: failed to configure dma channel\n"); - ret = -EBUSY; - return ret; - } - - return 0; -} - -static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_pcm_dma_params *prtd; - struct ssc_device *ssc; - struct at_dma_slave *sdata = NULL; - int ret; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ssc = prtd->ssc; - if (ssc->pdev) - sdata = ssc->pdev->dev.platform_data; - - ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata); - if (ret) { - pr_err("atmel-pcm: dmaengine pcm open failed\n"); - return -EINVAL; - } - - ret = atmel_pcm_configure_dma(substream, params, prtd); - if (ret) { - pr_err("atmel-pcm: failed to configure dmai\n"); - goto err; + slave_config->src_addr = ssc->phybase + SSC_RHR; + slave_config->src_maxburst = 1; } prtd->dma_intr_handler = atmel_pcm_dma_irq; return 0; -err: - snd_dmaengine_pcm_close_release_chan(substream); - return ret; } -static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_pcm_dma_params *prtd; - - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error); - ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable); - - return 0; -} - -static int atmel_pcm_open(struct snd_pcm_substream *substream) -{ - snd_soc_set_runtime_hwparams(substream, &atmel_pcm_dma_hardware); - - return 0; -} - -static struct snd_pcm_ops atmel_pcm_ops = { - .open = atmel_pcm_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = atmel_pcm_hw_params, - .prepare = atmel_pcm_dma_prepare, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = atmel_pcm_mmap, -}; - -static struct snd_soc_platform_driver atmel_soc_platform = { - .ops = &atmel_pcm_ops, - .pcm_new = atmel_pcm_new, - .pcm_free = atmel_pcm_free, +static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { + .prepare_slave_config = atmel_pcm_configure_dma, + .pcm_hardware = &atmel_pcm_dma_hardware, + .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE, }; int atmel_pcm_dma_platform_register(struct device *dev) { - return snd_soc_register_platform(dev, &atmel_soc_platform); + return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } EXPORT_SYMBOL(atmel_pcm_dma_platform_register); void atmel_pcm_dma_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(dev); + snd_dmaengine_pcm_unregister(dev); } EXPORT_SYMBOL(atmel_pcm_dma_platform_unregister); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f3fdfa0..bb53dea 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -73,6 +73,7 @@ static struct atmel_ssc_mask ssc_tx_mask = { .ssc_disable = SSC_BIT(CR_TXDIS), .ssc_endx = SSC_BIT(SR_ENDTX), .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .ssc_error = SSC_BIT(SR_OVRUN), .pdc_enable = ATMEL_PDC_TXTEN, .pdc_disable = ATMEL_PDC_TXTDIS, }; @@ -82,6 +83,7 @@ static struct atmel_ssc_mask ssc_rx_mask = { .ssc_disable = SSC_BIT(CR_RXDIS), .ssc_endx = SSC_BIT(SR_ENDRX), .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .ssc_error = SSC_BIT(SR_OVRUN), .pdc_enable = ATMEL_PDC_RXTEN, .pdc_disable = ATMEL_PDC_RXTDIS, }; @@ -196,15 +198,27 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; - int dir_mask; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dir = 0; dir_mask = SSC_DIR_MASK_PLAYBACK; - else + } else { + dir = 1; dir_mask = SSC_DIR_MASK_CAPTURE; + } + + dma_params = &ssc_dma_params[dai->id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + snd_soc_dai_set_dma_data(dai, substream, dma_params); spin_lock_irq(&ssc_p->lock); if (ssc_p->dir_mask & dir_mask) { @@ -325,7 +339,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int id = dai->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; struct atmel_pcm_dma_params *dma_params; @@ -344,19 +357,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, else dir = 1; - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The snd_soc_pcm_stream->dma_data field is only used to communicate - * the appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_params); + dma_params = ssc_p->dma_params[dir]; channels = params_channels(params); @@ -648,6 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c new file mode 100644 index 0000000..7222380 --- /dev/null +++ b/sound/soc/atmel/atmel_wm8904.c @@ -0,0 +1,254 @@ +/* + * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec. + * + * Copyright (C) 2012 Atmel + * + * Author: Bo Shen <voice.shen@atmel.com> + * + * GPLv2 or later + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/pinctrl/consumer.h> + +#include <sound/soc.h> + +#include "../codecs/wm8904.h" +#include "atmel_ssc_dai.h" + +#define MCLK_RATE 32768 + +static struct clk *mclk; + +static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int atmel_asoc_wm8904_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->codec_dai; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, + 32768, params_rate(params) * 256); + if (ret < 0) { + pr_err("%s - failed to set wm8904 codec PLL.", __func__); + return ret; + } + + /* + * As here wm8904 use FLL output as its system clock + * so calling set_sysclk won't care freq parameter + * then we pass 0 + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("%s -failed to set wm8904 SYSCLK\n", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops atmel_asoc_wm8904_ops = { + .hw_params = atmel_asoc_wm8904_hw_params, +}; + +static int atmel_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + switch (level) { + case SND_SOC_BIAS_PREPARE: + clk_prepare_enable(mclk); + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(mclk); + break; + default: + break; + } + } + + return 0; +}; + +static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { + .name = "WM8904", + .stream_name = "WM8904 PCM", + .codec_dai_name = "wm8904-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &atmel_asoc_wm8904_ops, +}; + +static struct snd_soc_card atmel_asoc_wm8904_card = { + .name = "atmel_asoc_wm8904", + .owner = THIS_MODULE, + .set_bias_level = atmel_set_bias_level, + .dai_link = &atmel_asoc_wm8904_dailink, + .num_links = 1, + .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets), + .fully_routed = true, +}; + +static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int ret; + + if (!np) { + dev_err(&pdev->dev, "only device tree supported\n"); + return -EINVAL; + } + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "failed to parse card name\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio routing\n"); + return ret; + } + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "failed to get dai and pcm info\n"); + ret = -EINVAL; + return ret; + } + dailink->cpu_of_node = cpu_np; + dailink->platform_of_node = cpu_np; + of_node_put(cpu_np); + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "failed to get codec info\n"); + ret = -EINVAL; + return ret; + } + dailink->codec_of_node = codec_np; + of_node_put(codec_np); + + return 0; +} + +static int atmel_asoc_wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + struct clk *clk_src; + struct pinctrl *pinctrl; + int id, ret; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } + + card->dev = &pdev->dev; + ret = atmel_asoc_wm8904_dt_init(pdev); + if (ret) { + dev_err(&pdev->dev, "failed to init dt info\n"); + return ret; + } + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + ret = atmel_ssc_set_audio(id); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id); + return ret; + } + + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "failed to get pck0\n"); + ret = PTR_ERR(mclk); + goto err_set_audio; + } + + clk_src = clk_get(NULL, "clk32k"); + if (IS_ERR(clk_src)) { + dev_err(&pdev->dev, "failed to get clk32k\n"); + ret = PTR_ERR(clk_src); + goto err_set_audio; + } + + ret = clk_set_parent(mclk, clk_src); + clk_put(clk_src); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set MCLK parent\n"); + goto err_set_audio; + } + + dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); + clk_set_rate(mclk, MCLK_RATE); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + goto err_set_audio; + } + + return 0; + +err_set_audio: + atmel_ssc_put_audio(id); + return ret; +} + +static int atmel_asoc_wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int id; + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(id); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { + { .compatible = "atmel,asoc-wm8904", }, + { } +}; +#endif + +static struct platform_driver atmel_asoc_wm8904_driver = { + .driver = { + .name = "atmel-wm8904-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + }, + .probe = atmel_asoc_wm8904_probe, + .remove = atmel_asoc_wm8904_remove, +}; + +module_platform_driver(atmel_asoc_wm8904_driver); + +/* Module information */ +MODULE_AUTHOR("Bo Shen <voice.shen@atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 2d6fbd0..802717e 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -38,8 +38,6 @@ #include <linux/platform_device.h> #include <linux/i2c.h> -#include <linux/pinctrl/consumer.h> - #include <linux/atmel-ssc.h> #include <sound/core.h> @@ -203,15 +201,8 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) struct device_node *codec_np, *cpu_np; struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; - struct pinctrl *pinctrl; int ret; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "Failed to request pinctrl for mck\n"); - return PTR_ERR(pinctrl); - } - if (!np) { if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c new file mode 100644 index 0000000..992ae38 --- /dev/null +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -0,0 +1,208 @@ +/* + * sam9x5_wm8731 -- SoC audio for AT91SAM9X5-based boards + * that are using WM8731 as codec. + * + * Copyright (C) 2011 Atmel, + * Nicolas Ferre <nicolas.ferre@atmel.com> + * + * Copyright (C) 2013 Paratronic, + * Richard Genoud <richard.genoud@gmail.com> + * + * Based on sam9g20_wm8731.c by: + * Sedji Gaouaou <sedji.gaouaou@atmel.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/of.h> +#include <linux/export.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> +#include <linux/device.h> + +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8731.h" +#include "atmel_ssc_dai.h" + + +#define MCLK_RATE 12288000 + +#define DRV_NAME "sam9x5-snd-wm8731" + +struct sam9x5_drvdata { + int ssc_id; +}; + +/* + * Logic for a wm8731 as connected on a at91sam9x5ek based board. + */ +static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = rtd->dev; + int ret; + + dev_dbg(dev, "ASoC: %s called\n", __func__); + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + MCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to set WM8731 SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Audio paths on at91sam9x5ek board: + * + * |A| ------------> | | ---R----> Headphone Jack + * |T| <----\ | WM | ---L--/ + * |9| ---> CLK <--> | 8731 | <--R----- Line In Jack + * |1| <------------ | | <--L--/ + */ +static const struct snd_soc_dapm_widget sam9x5_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card; + struct snd_soc_dai_link *dai; + struct sam9x5_drvdata *priv; + int ret; + + if (!np) { + dev_err(&pdev->dev, "No device node supplied\n"); + return -EINVAL; + } + + card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL); + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL); + if (!dai || !card || !priv) { + ret = -ENOMEM; + goto out; + } + + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dai_link = dai; + card->num_links = 1; + card->dapm_widgets = sam9x5_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sam9x5_dapm_widgets); + dai->name = "WM8731"; + dai->stream_name = "WM8731 PCM"; + dai->codec_dai_name = "wm8731-hifi"; + dai->init = sam9x5_wm8731_init; + dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "atmel,model node missing\n"); + goto out; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "atmel,audio-routing node missing\n"); + goto out; + } + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "atmel,audio-codec node missing\n"); + ret = -EINVAL; + goto out; + } + + dai->codec_of_node = codec_np; + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "atmel,ssc-controller node missing\n"); + ret = -EINVAL; + goto out; + } + dai->cpu_of_node = cpu_np; + dai->platform_of_node = cpu_np; + + priv->ssc_id = of_alias_get_id(cpu_np, "ssc"); + + ret = atmel_ssc_set_audio(priv->ssc_id); + if (ret != 0) { + dev_err(&pdev->dev, + "ASoC: Failed to set SSC %d for audio: %d\n", + ret, priv->ssc_id); + goto out; + } + + of_node_put(codec_np); + of_node_put(cpu_np); + + platform_set_drvdata(pdev, card); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, + "ASoC: Platform device allocation failed\n"); + goto out_put_audio; + } + + dev_dbg(&pdev->dev, "ASoC: %s ok\n", __func__); + + return ret; + +out_put_audio: + atmel_ssc_put_audio(priv->ssc_id); +out: + return ret; +} + +static int sam9x5_wm8731_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sam9x5_drvdata *priv = card->drvdata; + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(priv->ssc_id); + + return 0; +} + +static const struct of_device_id sam9x5_wm8731_of_match[] = { + { .compatible = "atmel,sam9x5-wm8731-audio", }, + {}, +}; +MODULE_DEVICE_TABLE(of, sam9x5_wm8731_of_match); + +static struct platform_driver sam9x5_wm8731_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), + }, + .probe = sam9x5_wm8731_driver_probe, + .remove = sam9x5_wm8731_driver_remove, +}; +module_platform_driver(sam9x5_wm8731_driver); + +/* Module information */ +MODULE_AUTHOR("Nicolas Ferre <nicolas.ferre@atmel.com>"); +MODULE_AUTHOR("Richard Genoud <richard.genoud@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC machine driver for AT91SAM9x5 - WM8731"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 44b8dce..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -179,13 +179,12 @@ static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ac97c_bus_ops = { .read = au1xac97c_ac97_read, .write = au1xac97c_ac97_write, .reset = au1xac97c_ac97_cold_reset, .warm_reset = au1xac97c_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */ static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -272,6 +271,10 @@ static int au1xac97c_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); + ret = snd_soc_set_ac97_ops(&ac97c_bus_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, &au1xac97c_dai_driver, 1); if (ret) @@ -338,19 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -static int __init au1xac97c_load(void) -{ - ac97c_workdata = NULL; - return platform_driver_register(&au1xac97c_driver); -} - -static void __exit au1xac97c_unload(void) -{ - platform_driver_unregister(&au1xac97c_driver); -} - -module_init(au1xac97c_load); -module_exit(au1xac97c_unload); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index a497a0c..decba87 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -73,12 +73,14 @@ static struct snd_soc_dai_link db1300_ac97_dai = { static struct snd_soc_card db1300_ac97_machine = { .name = "DB1300_AC97", + .owner = THIS_MODULE, .dai_link = &db1300_ac97_dai, .num_links = 1, }; static struct snd_soc_card db1550_ac97_machine = { .name = "DB1550_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -145,6 +147,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = { static struct snd_soc_card db1300_i2s_machine = { .name = "DB1300_I2S", + .owner = THIS_MODULE, .dai_link = &db1300_i2s_dai, .num_links = 1, }; @@ -161,6 +164,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = { static struct snd_soc_card db1550_i2s_machine = { .name = "DB1550_I2S", + .owner = THIS_MODULE, .dai_link = &db1550_i2s_dai, .num_links = 1, }; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 8f1862a..986dcec 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -201,13 +201,12 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = au1xpsc_ac97_read, .write = au1xpsc_ac97_write, .reset = au1xpsc_ac97_cold_reset, .warm_reset = au1xpsc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -380,18 +379,9 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) mutex_init(&wd->lock); iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) - return -ENODEV; - - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) @@ -423,6 +413,10 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); + ret = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, &wd->dai_drv, 1); if (ret) @@ -503,19 +497,7 @@ static struct platform_driver au1xpsc_ac97_driver = { .remove = au1xpsc_ac97_drvremove, }; -static int __init au1xpsc_ac97_load(void) -{ - au1xpsc_ac97_workdata = NULL; - return platform_driver_register(&au1xpsc_ac97_driver); -} - -static void __exit au1xpsc_ac97_unload(void) -{ - platform_driver_unregister(&au1xpsc_ac97_driver); -} - -module_init(au1xpsc_ac97_load); -module_exit(au1xpsc_ac97_unload); +module_platform_driver(au1xpsc_ac97_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 16b88f5..54f74f8 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -56,6 +56,23 @@ config SND_SOC_BFIN_EVAL_ADAV80X Note: This driver assumes that the ADAV80X digital record and playback interfaces are connected to the first SPORT port on the BF5XX board. +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + +config SND_BF5XX_SOC_AD193X + tristate "SoC AD193X Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD193X + help + Say Y if you want to add support for AD193X codec on Blackfin. + This driver supports AD1936, AD1937, AD1938 and AD1939. + config SND_BF5XX_SOC_AD73311 tristate "SoC AD73311 Audio support for Blackfin" depends on SND_BF5XX_I2S @@ -72,33 +89,6 @@ config SND_BFIN_AD73311_SE Enter the GPIO used to control AD73311's SE pin. Acceptable values are 0 to 7 -config SND_BF5XX_TDM - tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" - depends on (BLACKFIN && SND_SOC) - select SND_BF5XX_SOC_SPORT - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in TDM - mode. - You will also need to select the audio interfaces to support below. - -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD193X - tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD193X - help - Say Y if you want to add support for AD193X codec on Blackfin. - This driver supports AD1936, AD1937, AD1938 and AD1939. - config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN @@ -174,9 +164,6 @@ config SND_BF5XX_SOC_I2S config SND_BF6XX_SOC_I2S tristate -config SND_BF5XX_SOC_TDM - tristate - config SND_BF5XX_SOC_AC97 tristate diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 6fea1f4..ad0a6e9 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,23 +1,19 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o -snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o -snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o -obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o -obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support snd-ad1836-objs := bf5xx-ad1836.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 7e2f360..53f8408 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h deleted file mode 100644 index d324d58..0000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-ac97-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * 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 _BF5XX_AC97_PCM_H -#define _BF5XX_AC97_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 4902173..e82eb37 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -198,13 +198,12 @@ static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) #endif } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops bf5xx_ac97_ops = { .read = bf5xx_ac97_read, .write = bf5xx_ac97_write, .warm_reset = bf5xx_ac97_warm_reset, .reset = bf5xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) @@ -231,9 +230,9 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) return 0; #if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport, 16, 0x3FF, 1); + ret = sport_set_multichannel(sport, 16, 0x3FF, 0x3FF, 1); #else - ret = sport_set_multichannel(sport, 16, 0x1F, 1); + ret = sport_set_multichannel(sport, 16, 0x1F, 0x1F, 1); #endif if (ret) { pr_err("SPORT is busy!\n"); @@ -293,13 +292,15 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ - if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) { - pr_err("Failed to request GPIO_%d for reset\n", - CONFIG_SND_BF5XX_RESET_GPIO_NUM); - ret = -1; - goto gpio_err; + ret = devm_gpio_request_one(&pdev->dev, + CONFIG_SND_BF5XX_RESET_GPIO_NUM, + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); + if (ret) { + dev_err(&pdev->dev, + "Failed to request GPIO_%d for reset: %d\n", + CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); + return ret; } - gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), @@ -311,9 +312,9 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /*SPORT works in TDM mode to simulate AC97 transfers*/ #if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 0x3FF, 1); #else - ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); + ret = sport_set_multichannel(sport_handle, 16, 0x1F, 0x1F, 1); #endif if (ret) { pr_err("SPORT is busy!\n"); @@ -335,6 +336,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) goto sport_config_err; } + ret = snd_soc_set_ac97_ops(&bf5xx_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto sport_config_err; + } + ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, &bfin_ac97_dai, 1); if (ret) { @@ -349,10 +356,7 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -gpio_err: -#endif + snd_soc_set_ac97_ops(NULL); return ret; } @@ -363,9 +367,7 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e..a680fdc 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,8 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97_bus_ops bf5xx_ac97_ops; -extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { u16 ac97_tag; /* slot 0 */ diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d23f4b0..8fcfc4e 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -30,15 +30,10 @@ #include "../codecs/ad1836.h" -#include "bf5xx-tdm-pcm.h" -#include "bf5xx-tdm.h" - static struct snd_soc_card bf5xx_ad1836; -static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ad1836_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; @@ -49,13 +44,13 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + return 0; } -static struct snd_soc_ops bf5xx_ad1836_ops = { - .hw_params = bf5xx_ad1836_hw_params, -}; - #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) @@ -63,9 +58,9 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai = { .name = "ad1836", .stream_name = "AD1836", .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .ops = &bf5xx_ad1836_ops, + .platform_name = "bfin-i2s-pcm-audio", .dai_fmt = BF5XX_AD1836_DAIFMT, + .init = bf5xx_ad1836_init, }; static struct snd_soc_card bf5xx_ad1836 = { diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 0e55e9f..603ad1f 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -39,30 +39,16 @@ #include "../codecs/ad193x.h" -#include "bf5xx-tdm-pcm.h" -#include "bf5xx-tdm.h" - static struct snd_soc_card bf5xx_ad193x; -static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ad193x_link_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int clk = 0; - unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; - int ret = 0; - - switch (params_rate(params)) { - case 48000: - clk = 24576000; - break; - } + int ret; /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -71,9 +57,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set cpu DAI channel mapping */ - ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), - channel_map, ARRAY_SIZE(channel_map), channel_map); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); if (ret < 0) return ret; @@ -83,30 +67,26 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, #define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_ops bf5xx_ad193x_ops = { - .hw_params = bf5xx_ad193x_hw_params, -}; - static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { { .name = "ad193x", .stream_name = "AD193X", - .cpu_dai_name = "bfin-tdm.0", + .cpu_dai_name = "bfin-i2s.0", .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-tdm-pcm-audio", + .platform_name = "bfin-i2s-pcm-audio", .codec_name = "spi0.5", - .ops = &bf5xx_ad193x_ops, .dai_fmt = BF5XX_AD193X_DAIFMT, + .init = bf5xx_ad193x_link_init, }, { .name = "ad193x", .stream_name = "AD193X", - .cpu_dai_name = "bfin-tdm.1", + .cpu_dai_name = "bfin-i2s.1", .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-tdm-pcm-audio", + .platform_name = "bfin-i2s-pcm-audio", .codec_name = "spi0.5", - .ops = &bf5xx_ad193x_ops, .dai_fmt = BF5XX_AD193X_DAIFMT, + .init = bf5xx_ad193x_link_init, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index b30f88b..3450e8f 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -48,7 +48,6 @@ #include "../codecs/ad1980.h" -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 61cc91d..786bbdd 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -45,7 +45,6 @@ #include "../codecs/ad73311.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" #if CONFIG_SND_BF5XX_SPORT_NUM == 0 #define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 262c1de..9cb4a80 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -39,8 +39,8 @@ #include <asm/dma.h> -#include "bf5xx-i2s-pcm.h" #include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" static void bf5xx_dma_irq(void *data) { @@ -50,7 +50,6 @@ static void bf5xx_dma_irq(void *data) static const struct snd_pcm_hardware bf5xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER, .formats = SNDRV_PCM_FMTBIT_S16_LE | @@ -67,10 +66,16 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - snd_pcm_lib_malloc_pages(substream, size); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int buffer_size = params_buffer_bytes(params); + struct bf5xx_i2s_pcm_data *dma_data; - return 0; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) + buffer_size = buffer_size / params_channels(params) * 8; + + return snd_pcm_lib_malloc_pages(substream, buffer_size); } static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) @@ -82,9 +87,16 @@ static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct sport_device *sport = runtime->private_data; int period_bytes = frames_to_bytes(runtime, runtime->period_size); + struct bf5xx_i2s_pcm_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) + period_bytes = period_bytes / runtime->channels * 8; pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -131,10 +143,15 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct sport_device *sport = runtime->private_data; unsigned int diff; snd_pcm_uframes_t frames; + struct bf5xx_i2s_pcm_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff = sport_curr_offset_tx(sport); @@ -151,6 +168,8 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) diff = 0; frames = bytes_to_frames(substream->runtime, diff); + if (dma_data->tdm_mode) + frames = frames * runtime->channels / 8; return frames; } @@ -162,11 +181,18 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dma_buffer *buf = &substream->dma_buffer; + struct bf5xx_i2s_pcm_data *dma_data; int ret; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + pr_debug("%s enter\n", __func__); snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); + if (dma_data->tdm_mode) + runtime->hw.buffer_bytes_max /= 4; + else + runtime->hw.info |= SNDRV_PCM_INFO_MMAP; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -202,6 +228,88 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, return 0 ; } +static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int sample_size = runtime->sample_bits / 8; + struct bf5xx_i2s_pcm_data *dma_data; + unsigned int i; + void *src, *dst; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = runtime->dma_area; + dst += pos * sample_size * 8; + + while (count--) { + for (i = 0; i < runtime->channels; i++) { + memcpy(dst + dma_data->map[i] * + sample_size, src, sample_size); + src += sample_size; + } + dst += 8 * sample_size; + } + } else { + src = runtime->dma_area; + src += pos * sample_size * 8; + dst = buf; + + while (count--) { + for (i = 0; i < runtime->channels; i++) { + memcpy(dst, src + dma_data->map[i] * + sample_size, sample_size); + dst += sample_size; + } + src += 8 * sample_size; + } + } + } else { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = runtime->dma_area; + dst += frames_to_bytes(runtime, pos); + } else { + src = runtime->dma_area; + src += frames_to_bytes(runtime, pos); + dst = buf; + } + + memcpy(dst, src, frames_to_bytes(runtime, count)); + } + + return 0; +} + +static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int sample_size = runtime->sample_bits / 8; + void *buf = runtime->dma_area; + struct bf5xx_i2s_pcm_data *dma_data; + unsigned int offset, size; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) { + offset = pos * 8 * sample_size; + size = count * 8 * sample_size; + } else { + offset = frames_to_bytes(runtime, pos); + size = frames_to_bytes(runtime, count); + } + + snd_pcm_format_set_silence(runtime->format, buf + offset, size); + + return 0; +} + static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, @@ -211,57 +319,16 @@ static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .trigger = bf5xx_pcm_trigger, .pointer = bf5xx_pcm_pointer, .mmap = bf5xx_pcm_mmap, + .copy = bf5xx_pcm_copy, + .silence = bf5xx_pcm_silence, }; -static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) { - pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } - buf->bytes = size; - - pr_debug("%s, area:%p, size:0x%08lx\n", __func__, - buf->area, buf->bytes); - - return 0; -} - -static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(NULL, buf->bytes, buf->area, 0); - buf->area = NULL; - } -} - static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; pr_debug("%s enter\n", __func__); if (!card->dev->dma_mask) @@ -269,27 +336,13 @@ static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, card->dev, size, size); } static struct snd_soc_platform_driver bf5xx_i2s_soc_platform = { .ops = &bf5xx_pcm_i2s_ops, .pcm_new = bf5xx_pcm_i2s_new, - .pcm_free = bf5xx_pcm_free_dma_buffers, }; static int bfin_i2s_soc_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h index 0c2c5a6..1f04352 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.h @@ -1,26 +1,17 @@ /* - * linux/sound/arm/bf5xx-i2s-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * * 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 _BF5XX_I2S_PCM_H -#define _BF5XX_I2S_PCM_H +#ifndef _BF5XX_TDM_PCM_H +#define _BF5XX_TDM_PCM_H -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; +#define BFIN_TDM_DAI_MAX_SLOTS 8 -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; +struct bf5xx_i2s_pcm_data { + unsigned int map[BFIN_TDM_DAI_MAX_SLOTS]; + bool tdm_mode; }; #endif diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index dd0c2a4..9a174fc 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -42,6 +42,7 @@ #include <linux/gpio.h> #include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" struct bf5xx_i2s_port { u16 tcr1; @@ -49,6 +50,13 @@ struct bf5xx_i2s_port { u16 tcr2; u16 rcr2; int configured; + + unsigned int slots; + unsigned int tx_mask; + unsigned int rx_mask; + + struct bf5xx_i2s_pcm_data tx_dma_data; + struct bf5xx_i2s_pcm_data rx_dma_data; }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -74,7 +82,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: - printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); + dev_err(cpu_dai->dev, "%s: Unknown DAI format type\n", + __func__); ret = -EINVAL; break; } @@ -88,7 +97,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: - printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); + dev_err(cpu_dai->dev, "%s: Unknown DAI master type\n", + __func__); ret = -EINVAL; break; } @@ -141,14 +151,14 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, bf5xx_i2s->rcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, bf5xx_i2s->tcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } } @@ -162,18 +172,76 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - pr_debug("%s enter\n", __func__); + dev_dbg(dai->dev, "%s enter\n", __func__); /* No active stream, SPORT is allowed to be configured again. */ if (!dai->active) bf5xx_i2s->configured = 0; } +static int bf5xx_i2s_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + unsigned int tx_mapped = 0, rx_mapped = 0; + unsigned int slot; + int i; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_i2s->tx_dma_data.map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_i2s->rx_dma_data.map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + +static int bf5xx_i2s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + + if (slots % 8 != 0 || slots > 8) + return -EINVAL; + + if (width != 32) + return -EINVAL; + + bf5xx_i2s->slots = slots; + bf5xx_i2s->tx_mask = tx_mask; + bf5xx_i2s->rx_mask = rx_mask; + + bf5xx_i2s->tx_dma_data.tdm_mode = slots != 0; + bf5xx_i2s->rx_dma_data.tdm_mode = slots != 0; + + return sport_set_multichannel(sport_handle, slots, tx_mask, rx_mask, 0); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - pr_debug("%s : sport %d\n", __func__, dai->id); + dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); if (dai->capture_active) sport_rx_stop(sport_handle); @@ -188,23 +256,24 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; int ret; - pr_debug("%s : sport %d\n", __func__, dai->id); + dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, bf5xx_i2s->rcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, bf5xx_i2s->tcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } - return 0; + return sport_set_multichannel(sport_handle, bf5xx_i2s->slots, + bf5xx_i2s->tx_mask, bf5xx_i2s->rx_mask, 0); } #else @@ -212,6 +281,23 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) #define bf5xx_i2s_resume NULL #endif +static int bf5xx_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + unsigned int i; + + for (i = 0; i < BFIN_TDM_DAI_MAX_SLOTS; i++) { + bf5xx_i2s->tx_dma_data.map[i] = i; + bf5xx_i2s->rx_dma_data.map[i] = i; + } + + dai->playback_dma_data = &bf5xx_i2s->tx_dma_data; + dai->capture_dma_data = &bf5xx_i2s->rx_dma_data; + + return 0; +} + #define BF5XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ @@ -224,22 +310,25 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, + .shutdown = bf5xx_i2s_shutdown, + .hw_params = bf5xx_i2s_hw_params, + .set_fmt = bf5xx_i2s_set_dai_fmt, + .set_tdm_slot = bf5xx_i2s_set_tdm_slot, + .set_channel_map = bf5xx_i2s_set_channel_map, }; static struct snd_soc_dai_driver bf5xx_i2s_dai = { + .probe = bf5xx_i2s_dai_probe, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .ops = &bf5xx_i2s_dai_ops, @@ -255,7 +344,7 @@ static int bf5xx_i2s_probe(struct platform_device *pdev) int ret; /* configure SPORT for I2S */ - sport_handle = sport_init(pdev, 4, 2 * sizeof(u32), + sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), sizeof(struct bf5xx_i2s_port)); if (!sport_handle) return -ENODEV; @@ -264,7 +353,7 @@ static int bf5xx_i2s_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &bf5xx_i2s_component, &bf5xx_i2s_dai, 1); if (ret) { - pr_err("Failed to register DAI: %d\n", ret); + dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret); sport_done(sport_handle); return ret; } @@ -276,7 +365,7 @@ static int bf5xx_i2s_remove(struct platform_device *pdev) { struct sport_device *sport_handle = platform_get_drvdata(pdev); - pr_debug("%s enter\n", __func__); + dev_dbg(&pdev->dev, "%s enter\n", __func__); snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 2fd9f2a..6953512 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -46,10 +46,10 @@ /* note: multichannel is in units of 8 channels, * tdm_count is # channels NOT / 8 ! */ int sport_set_multichannel(struct sport_device *sport, - int tdm_count, u32 mask, int packed) + int tdm_count, u32 tx_mask, u32 rx_mask, int packed) { - pr_debug("%s tdm_count=%d mask:0x%08x packed=%d\n", __func__, - tdm_count, mask, packed); + pr_debug("%s tdm_count=%d tx_mask:0x%08x rx_mask:0x%08x packed=%d\n", + __func__, tdm_count, tx_mask, rx_mask, packed); if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) return -EBUSY; @@ -65,8 +65,8 @@ int sport_set_multichannel(struct sport_device *sport, sport->regs->mcmc2 = FRAME_DELAY | MCMEN | \ (packed ? (MCDTXPE|MCDRXPE) : 0); - sport->regs->mtcs0 = mask; - sport->regs->mrcs0 = mask; + sport->regs->mtcs0 = tx_mask; + sport->regs->mrcs0 = rx_mask; sport->regs->mtcs1 = 0; sport->regs->mrcs1 = 0; sport->regs->mtcs2 = 0; diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 5ab60bd..9fc2192 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -128,7 +128,7 @@ void sport_done(struct sport_device *sport); /* note: multichannel is in units of 8 channels, tdm_count is number of channels * NOT / 8 ! all channels are enabled by default */ int sport_set_multichannel(struct sport_device *sport, int tdm_count, - u32 mask, int packed); + u32 tx_mask, u32 rx_mask, int packed); int sport_config_rx(struct sport_device *sport, unsigned int rcr1, unsigned int rcr2, diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 7dbeef1..9c19ccc 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -40,7 +40,6 @@ #include <linux/gpio.h> #include "../codecs/ssm2602.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" static struct snd_soc_card bf5xx_ssm2602; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c deleted file mode 100644 index 0e6b888..0000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ /dev/null @@ -1,345 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-tdm-pcm.c - * Author: Barry Song <Barry.Song@analog.com> - * - * Created: Tue June 06 2009 - * Description: DMA driver for tdm codec - * - * Modified: - * Copyright 2009 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/init.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/gfp.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "bf5xx-tdm-pcm.h" -#include "bf5xx-tdm.h" -#include "bf5xx-sport.h" - -#define PCM_BUFFER_MAX 0x8000 -#define FRAGMENT_SIZE_MIN (4*1024) -#define FRAGMENTS_MIN 2 -#define FRAGMENTS_MAX 32 - -static void bf5xx_dma_irq(void *data) -{ - struct snd_pcm_substream *pcm = data; - snd_pcm_period_elapsed(pcm); -} - -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_48000, - .channels_min = 2, - .channels_max = 8, - .buffer_bytes_max = PCM_BUFFER_MAX, - .period_bytes_min = FRAGMENT_SIZE_MIN, - .period_bytes_max = PCM_BUFFER_MAX/2, - .periods_min = FRAGMENTS_MIN, - .periods_max = FRAGMENTS_MAX, -}; - -static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - snd_pcm_lib_malloc_pages(substream, size * 4); - - return 0; -} - -static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - - return 0; -} - -static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int fragsize_bytes = frames_to_bytes(runtime, runtime->period_size); - - fragsize_bytes /= runtime->channels; - /* inflate the fragsize to match the dma width of SPORT */ - fragsize_bytes *= 8; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, runtime->dma_area, - runtime->periods, fragsize_bytes); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, runtime->dma_area, - runtime->periods, fragsize_bytes); - } - - return 0; -} - -static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_start(sport); - else - sport_rx_start(sport); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_stop(sport); - else - sport_rx_stop(sport); - break; - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int diff; - snd_pcm_uframes_t frames; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - diff = sport_curr_offset_tx(sport); - frames = diff / (8*4); /* 32 bytes per frame */ - } else { - diff = sport_curr_offset_rx(sport); - frames = diff / (8*4); - } - return frames; -} - -static int bf5xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - if (sport_handle != NULL) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - - runtime->private_data = sport_handle; - } else { - pr_err("sport_handle is NULL\n"); - ret = -ENODEV; - } -out: - return ret; -} - -static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - struct bf5xx_tdm_port *tdm_port = sport->private_data; - unsigned int *src; - unsigned int *dst; - int i; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = buf; - dst = (unsigned int *)substream->runtime->dma_area; - - dst += pos * 8; - while (count--) { - for (i = 0; i < substream->runtime->channels; i++) - *(dst + tdm_port->tx_map[i]) = *src++; - dst += 8; - } - } else { - src = (unsigned int *)substream->runtime->dma_area; - dst = buf; - - src += pos * 8; - while (count--) { - for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src + tdm_port->rx_map[i]); - src += 8; - } - } - - return 0; -} - -static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, - int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) -{ - unsigned char *buf = substream->runtime->dma_area; - buf += pos * 8 * 4; - memset(buf, '\0', count * 8 * 4); - - return 0; -} - - -struct snd_pcm_ops bf5xx_pcm_tdm_ops = { - .open = bf5xx_pcm_open, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = bf5xx_pcm_hw_params, - .hw_free = bf5xx_pcm_hw_free, - .prepare = bf5xx_pcm_prepare, - .trigger = bf5xx_pcm_trigger, - .pointer = bf5xx_pcm_pointer, - .copy = bf5xx_pcm_copy, - .silence = bf5xx_pcm_silence, -}; - -static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, - &buf->addr, GFP_KERNEL); - if (!buf->area) { - pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } - buf->bytes = size; - - return 0; -} - -static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(NULL, buf->bytes, buf->area, 0); - buf->area = NULL; - } -} - -static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); - -static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &bf5xx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } -out: - return ret; -} - -static struct snd_soc_platform_driver bf5xx_tdm_soc_platform = { - .ops = &bf5xx_pcm_tdm_ops, - .pcm_new = bf5xx_pcm_tdm_new, - .pcm_free = bf5xx_pcm_free_dma_buffers, -}; - -static int bf5xx_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &bf5xx_tdm_soc_platform); -} - -static int bf5xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver bfin_tdm_driver = { - .driver = { - .name = "bfin-tdm-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = bf5xx_soc_platform_probe, - .remove = bf5xx_soc_platform_remove, -}; - -module_platform_driver(bfin_tdm_driver); - -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h deleted file mode 100644 index 7f8cc01..0000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2009 Analog Device Inc. - * - * 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 _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c deleted file mode 100644 index 69e9a3e..0000000 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ /dev/null @@ -1,328 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-tdm.c - * Author: Barry Song <Barry.Song@analog.com> - * - * Created: Thurs June 04 2009 - * Description: Blackfin I2S(TDM) CPU DAI driver - * Even though TDM mode can be as part of I2S DAI, but there - * are so much difference in configuration and data flow, - * it's very ugly to integrate I2S and TDM into a module - * - * Modified: - * Copyright 2009 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/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <asm/irq.h> -#include <asm/portmux.h> -#include <linux/mutex.h> -#include <linux/gpio.h> - -#include "bf5xx-sport.h" -#include "bf5xx-tdm.h" - -static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - int ret = 0; - - /* interface format:support TDM,slave mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - 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_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; - } - - return ret; -} - -static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; - int ret = 0; - - bf5xx_tdm->tcr2 &= ~0x1f; - bf5xx_tdm->rcr2 &= ~0x1f; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S32_LE: - bf5xx_tdm->tcr2 |= 31; - bf5xx_tdm->rcr2 |= 31; - sport_handle->wdsize = 4; - break; - /* at present, we only support 32bit transfer */ - default: - pr_err("not supported PCM format yet\n"); - return -EINVAL; - break; - } - - if (!bf5xx_tdm->configured) { - /* - * TX and RX are not independent,they are enabled at the - * same time, even if only one side is running. So, we - * need to configure both of them at the time when the first - * stream is opened. - * - * CPU DAI:slave mode. - */ - ret = sport_config_rx(sport_handle, bf5xx_tdm->rcr1, - bf5xx_tdm->rcr2, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport_handle, bf5xx_tdm->tcr1, - bf5xx_tdm->tcr2, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - bf5xx_tdm->configured = 1; - } - - return 0; -} - -static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; - - /* No active stream, SPORT is allowed to be configured again. */ - if (!dai->active) - bf5xx_tdm->configured = 0; -} - -static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; - int i; - unsigned int slot; - unsigned int tx_mapped = 0, rx_mapped = 0; - - if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || - (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) - return -EINVAL; - - for (i = 0; i < tx_num; i++) { - slot = tx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(tx_mapped & (1 << slot)))) { - bf5xx_tdm->tx_map[i] = slot; - tx_mapped |= 1 << slot; - } else - return -EINVAL; - } - for (i = 0; i < rx_num; i++) { - slot = rx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(rx_mapped & (1 << slot)))) { - bf5xx_tdm->rx_map[i] = slot; - rx_mapped |= 1 << slot; - } else - return -EINVAL; - } - - return 0; -} - -#ifdef CONFIG_PM -static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - if (dai->playback_active) - sport_tx_stop(sport); - if (dai->capture_active) - sport_rx_stop(sport); - - /* isolate sync/clock pins from codec while sports resume */ - peripheral_free_list(sport->pin_req); - - return 0; -} - -static int bf5xx_tdm_resume(struct snd_soc_dai *dai) -{ - int ret; - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - ret = sport_set_multichannel(sport, 8, 0xFF, 1); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - } - - ret = sport_config_rx(sport, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - } - - ret = sport_config_tx(sport, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - } - - peripheral_request_list(sport->pin_req, "soc-audio"); - - return 0; -} - -#else -#define bf5xx_tdm_suspend NULL -#define bf5xx_tdm_resume NULL -#endif - -static const struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { - .hw_params = bf5xx_tdm_hw_params, - .set_fmt = bf5xx_tdm_set_dai_fmt, - .shutdown = bf5xx_tdm_shutdown, - .set_channel_map = bf5xx_tdm_set_channel_map, -}; - -static struct snd_soc_dai_driver bf5xx_tdm_dai = { - .suspend = bf5xx_tdm_suspend, - .resume = bf5xx_tdm_resume, - .playback = { - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, - .ops = &bf5xx_tdm_dai_ops, -}; - -static const struct snd_soc_component_driver bf5xx_tdm_component = { - .name = "bf5xx-tdm", -}; - -static int bfin_tdm_probe(struct platform_device *pdev) -{ - struct sport_device *sport_handle; - int ret; - - /* configure SPORT for TDM */ - sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), - sizeof(struct bf5xx_tdm_port)); - if (!sport_handle) - return -ENODEV; - - /* SPORT works in TDM mode */ - ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = snd_soc_register_component(&pdev->dev, &bf5xx_tdm_component, - &bf5xx_tdm_dai, 1); - if (ret) { - pr_err("Failed to register DAI: %d\n", ret); - goto sport_config_err; - } - - return 0; - -sport_config_err: - sport_done(sport_handle); - return ret; -} - -static int bfin_tdm_remove(struct platform_device *pdev) -{ - struct sport_device *sport_handle = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - sport_done(sport_handle); - - return 0; -} - -static struct platform_driver bfin_tdm_driver = { - .probe = bfin_tdm_probe, - .remove = bfin_tdm_remove, - .driver = { - .name = "bfin-tdm", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(bfin_tdm_driver); - -/* Module information */ -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("TDM driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h deleted file mode 100644 index e986a3e..0000000 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_TDM_H -#define _BF5XX_TDM_H - -#define BFIN_TDM_DAI_MAX_SLOTS 8 -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; - unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; - int configured; -}; - -#endif diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 88143db..2c20f01 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,7 +1,7 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" depends on ARCH_EP93XX && SND_SOC - select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the EP93xx I2S or AC97 interfaces. diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 7798fbd..efa75b5 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -102,13 +102,13 @@ static struct ep93xx_ac97_info *ep93xx_ac97_info; static struct ep93xx_dma_data ep93xx_ac97_pcm_out = { .name = "ac97-pcm-out", - .dma_port = EP93XX_DMA_AAC1, + .port = EP93XX_DMA_AAC1, .direction = DMA_MEM_TO_DEV, }; static struct ep93xx_dma_data ep93xx_ac97_pcm_in = { .name = "ac97-pcm-in", - .dma_port = EP93XX_DMA_AAC1, + .port = EP93XX_DMA_AAC1, .direction = DMA_DEV_TO_MEM, }; @@ -237,13 +237,12 @@ static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ep93xx_ac97_ops = { .read = ep93xx_ac97_read, .write = ep93xx_ac97_write, .reset = ep93xx_ac97_cold_reset, .warm_reset = ep93xx_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -314,22 +313,15 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, return 0; } -static int ep93xx_ac97_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai) { - struct ep93xx_dma_data *dma_data; + dai->playback_dma_data = &ep93xx_ac97_pcm_out; + dai->capture_dma_data = &ep93xx_ac97_pcm_in; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &ep93xx_ac97_pcm_out; - else - dma_data = &ep93xx_ac97_pcm_in; - - snd_soc_dai_set_dma_data(dai, substream, dma_data); return 0; } static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { - .startup = ep93xx_ac97_startup, .trigger = ep93xx_ac97_trigger, }; @@ -337,6 +329,7 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { .name = "ep93xx-ac97", .id = 0, .ac97_control = 1, + .probe = ep93xx_ac97_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -370,9 +363,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -ENODEV; - info->regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); @@ -395,6 +385,10 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); + ret = snd_soc_set_ac97_ops(&ep93xx_ac97_ops); + if (ret) + goto fail; + ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component, &ep93xx_ac97_dai, 1); if (ret) @@ -403,9 +397,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return 0; fail: - platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); + snd_soc_set_ac97_ops(NULL); return ret; } @@ -418,9 +411,9 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) /* disable the AC97 controller */ ep93xx_ac97_write_reg(info, AC97GCR, 0); - platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); + + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 5c1102e..a57643d 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -60,11 +60,10 @@ struct ep93xx_i2s_info { struct clk *mclk; struct clk *sclk; struct clk *lrclk; - struct ep93xx_dma_data *dma_data; void __iomem *regs; }; -struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { +static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { [SNDRV_PCM_STREAM_PLAYBACK] = { .name = "i2s-pcm-out", .port = EP93XX_DMA_I2S1, @@ -139,15 +138,11 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) } } -static int ep93xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; - snd_soc_dai_set_dma_data(cpu_dai, substream, - &info->dma_data[substream->stream]); return 0; } @@ -338,7 +333,6 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) #endif static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { - .startup = ep93xx_i2s_startup, .shutdown = ep93xx_i2s_shutdown, .hw_params = ep93xx_i2s_hw_params, .set_sysclk = ep93xx_i2s_set_sysclk, @@ -349,6 +343,7 @@ static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { static struct snd_soc_dai_driver ep93xx_i2s_dai = { .symmetric_rates= 1, + .probe = ep93xx_i2s_dai_probe, .suspend = ep93xx_i2s_suspend, .resume = ep93xx_i2s_resume, .playback = { @@ -381,9 +376,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -ENODEV; - info->regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); @@ -407,7 +399,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, info); - info->dma_data = ep93xx_i2s_dma_data; err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, &ep93xx_i2s_dai, 1); @@ -417,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return 0; fail_put_lrclk: - dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); @@ -432,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev) struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 4880326..0e9f56e 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -14,20 +14,14 @@ #include <linux/module.h> #include <linux/init.h> -#include <linux/device.h> -#include <linux/slab.h> +#include <linux/platform_device.h> #include <linux/dmaengine.h> -#include <linux/dma-mapping.h> -#include <sound/core.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> #include <linux/platform_data/dma-ep93xx.h> -#include <mach/hardware.h> -#include <mach/ep93xx-regs.h> static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | @@ -63,134 +57,24 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } -static int ep93xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); - - return snd_dmaengine_pcm_open_request_chan(substream, - ep93xx_pcm_dma_filter, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); -} - -static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - return 0; -} - -static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops ep93xx_pcm_ops = { - .open = ep93xx_pcm_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = ep93xx_pcm_hw_params, - .hw_free = ep93xx_pcm_hw_free, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = ep93xx_pcm_mmap, -}; - -static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ep93xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - buf->bytes = size; - - return (buf->area == NULL) ? -ENOMEM : 0; -} - -static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, - buf->addr); - buf->area = NULL; - } -} - -static u64 ep93xx_pcm_dmamask = DMA_BIT_MASK(32); - -static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &ep93xx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = ep93xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = ep93xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver ep93xx_soc_platform = { - .ops = &ep93xx_pcm_ops, - .pcm_new = &ep93xx_pcm_new, - .pcm_free = &ep93xx_pcm_free_dma_buffers, +static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { + .pcm_hardware = &ep93xx_pcm_hardware, + .compat_filter_fn = ep93xx_pcm_dma_filter, + .prealloc_buffer_size = 131072, }; static int ep93xx_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &ep93xx_soc_platform); + return snd_dmaengine_pcm_register(&pdev->dev, + &ep93xx_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); } static int ep93xx_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_platform(&pdev->dev); + snd_dmaengine_pcm_unregister(&pdev->dev); return 0; } diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c0..8af0434 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -120,10 +120,8 @@ * before DAC & PGA in DAPM power-off sequence. */ #define PM860X_DAPM_OUTPUT(wname, wevent) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = 0, .invert = 0, .kcontrol_news = NULL, \ - .num_kcontrols = 0, .event = wevent, \ - .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, } + SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, 0, 0, NULL, 0, wevent, \ + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD) struct pm860x_det { struct snd_soc_jack *hp_jack; @@ -1444,7 +1442,7 @@ static int pm860x_codec_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_IRQ, i); if (!res) { dev_err(&pdev->dev, "Failed to get IRQ resources\n"); - goto out; + return -EINVAL; } pm860x->irq[i] = res->start + chip->irq_base; strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); @@ -1454,19 +1452,14 @@ static int pm860x_codec_probe(struct platform_device *pdev) pm860x_dai, ARRAY_SIZE(pm860x_dai)); if (ret) { dev_err(&pdev->dev, "Failed to register codec\n"); - goto out; + return -EINVAL; } return ret; - -out: - platform_set_drvdata(pdev, NULL); - return -EINVAL; } static int pm860x_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..b33b45d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + depends on COMPILE_TEST select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE @@ -19,7 +20,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAV80X + select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -40,7 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -53,13 +55,17 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI + select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C + select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C + select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -101,7 +107,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C && GENERIC_HARDIRQS + select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8955 if I2C @@ -120,6 +126,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C + select SND_SOC_WM8997 if MFD_WM8997 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -143,8 +150,10 @@ config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y + default y if SND_SOC_WM8997=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m + default m if SND_SOC_WM8997=m config SND_SOC_WM_HUBS tristate @@ -196,6 +205,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate @@ -263,7 +275,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC @@ -287,7 +299,13 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_OMAP_HDMI_CODEC +config SND_SOC_HDMI_CODEC + tristate + +config SND_SOC_PCM1681 + tristate + +config SND_SOC_PCM1792A tristate config SND_SOC_PCM3008 @@ -296,6 +314,9 @@ config SND_SOC_PCM3008 config SND_SOC_RT5631 tristate +config SND_SOC_RT5640 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate @@ -313,6 +334,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate @@ -492,6 +516,9 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate +config SND_SOC_WM8997 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..bc12676 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4554-objs := ak4554.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -27,7 +28,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -41,17 +42,21 @@ snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-omap-hdmi-codec-objs := omap-hdmi.o +snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o +snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o +snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -112,6 +117,7 @@ snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o +snd-soc-wm8997-objs := wm8997.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -136,6 +142,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o @@ -154,7 +161,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o @@ -168,14 +175,18 @@ obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o -obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o +obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o +obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o @@ -235,6 +246,7 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o +obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index a153b16..b8ba0ad 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1496,6 +1496,12 @@ static const char * const enum_ad_to_slot_map[] = {"AD_OUT1", "AD_OUT7", "AD_OUT8", "zeroes", + "zeroes", + "zeroes", + "zeroes", + "tristate", + "tristate", + "tristate", "tristate"}; static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map, AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT, @@ -2230,7 +2236,7 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int val, mask, slots_active; + unsigned int val, mask, slot, slots_active; mask = BIT(AB8500_DIGIFCONF2_IF0WL0) | BIT(AB8500_DIGIFCONF2_IF0WL1); @@ -2286,27 +2292,34 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); /* Setup TDM DA according to active tx slots */ + + if (tx_mask & ~0xff) + return -EINVAL; + mask = AB8500_DASLOTCONFX_SLTODAX_MASK; + tx_mask = tx_mask << AB8500_DA_DATA0_OFFSET; slots_active = hweight32(tx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__, slots_active); + switch (slots_active) { case 0: break; case 1: - /* Slot 9 -> DA_IN1 & DA_IN3 */ - snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + slot = find_first_bit((unsigned long *)&tx_mask, 32); + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; case 2: - /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */ - snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9); - snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9); - snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); - + slot = find_first_bit((unsigned long *)&tx_mask, 32); + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); + slot = find_next_bit((unsigned long *)&tx_mask, 32, slot + 1); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; case 8: dev_dbg(dai->codec->dev, @@ -2321,25 +2334,36 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, } /* Setup TDM AD according to active RX-slots */ + + if (rx_mask & ~0xff) + return -EINVAL; + + rx_mask = rx_mask << AB8500_AD_DATA0_OFFSET; slots_active = hweight32(rx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__, slots_active); + switch (slots_active) { case 0: break; case 1: - /* AD_OUT3 -> slot 0 & 1 */ - snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL, - AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | - AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD); + slot = find_first_bit((unsigned long *)&rx_mask, 32); + snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), + AB8500_MASK_SLOT(slot), + AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); break; case 2: - /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */ + slot = find_first_bit((unsigned long *)&rx_mask, 32); snd_soc_update_bits(codec, - AB8500_ADSLOTSEL1, - AB8500_MASK_ALL, - AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | - AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD); + AB8500_ADSLOTSEL(slot), + AB8500_MASK_SLOT(slot), + AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); + slot = find_next_bit((unsigned long *)&rx_mask, 32, slot + 1); + snd_soc_update_bits(codec, + AB8500_ADSLOTSEL(slot), + AB8500_MASK_SLOT(slot), + AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT2, slot)); break; case 8: dev_dbg(dai->codec->dev, @@ -2356,6 +2380,11 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static const struct snd_soc_dai_ops ab8500_codec_ops = { + .set_fmt = ab8500_codec_set_dai_fmt, + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, +}; + static struct snd_soc_dai_driver ab8500_codec_dai[] = { { .name = "ab8500-codec-dai.0", @@ -2367,12 +2396,7 @@ static struct snd_soc_dai_driver ab8500_codec_dai[] = { .rates = AB8500_SUPPORTED_RATE, .formats = AB8500_SUPPORTED_FMT, }, - .ops = (struct snd_soc_dai_ops[]) { - { - .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, - .set_fmt = ab8500_codec_set_dai_fmt, - } - }, + .ops = &ab8500_codec_ops, .symmetric_rates = 1 }, { @@ -2385,12 +2409,7 @@ static struct snd_soc_dai_driver ab8500_codec_dai[] = { .rates = AB8500_SUPPORTED_RATE, .formats = AB8500_SUPPORTED_FMT, }, - .ops = (struct snd_soc_dai_ops[]) { - { - .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, - .set_fmt = ab8500_codec_set_dai_fmt, - } - }, + .ops = &ab8500_codec_ops, .symmetric_rates = 1 } }; diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 306d0bc..e2e5442 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -24,6 +24,13 @@ #define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000) #define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE) +/* AB8500 interface slot offset definitions */ + +#define AB8500_AD_DATA0_OFFSET 0 +#define AB8500_DA_DATA0_OFFSET 8 +#define AB8500_AD_DATA1_OFFSET 16 +#define AB8500_DA_DATA1_OFFSET 24 + /* AB8500 audio bank (0x0d) register definitions */ #define AB8500_POWERUP 0x00 @@ -73,6 +80,7 @@ #define AB8500_ADSLOTSEL14 0x2C #define AB8500_ADSLOTSEL15 0x2D #define AB8500_ADSLOTSEL16 0x2E +#define AB8500_ADSLOTSEL(slot) (AB8500_ADSLOTSEL1 + (slot >> 1)) #define AB8500_ADSLOTHIZCTRL1 0x2F #define AB8500_ADSLOTHIZCTRL2 0x30 #define AB8500_ADSLOTHIZCTRL3 0x31 @@ -144,6 +152,7 @@ #define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1) #define AB8500_MASK_ALL 0xFF +#define AB8500_MASK_SLOT(slot) ((slot & 1) ? 0xF0 : 0x0F) #define AB8500_MASK_NONE 0x00 /* AB8500_POWERUP */ @@ -347,28 +356,21 @@ #define AB8500_DIGIFCONF4_IF1WL0 0 /* AB8500_ADSLOTSELX */ -#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x10 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x20 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x30 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x40 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x50 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x60 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x70 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x80 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0xF0 -#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x01 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x02 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x03 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x04 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x05 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x06 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x07 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x08 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0x0F +#define AB8500_AD_OUT1 0x0 +#define AB8500_AD_OUT2 0x1 +#define AB8500_AD_OUT3 0x2 +#define AB8500_AD_OUT4 0x3 +#define AB8500_AD_OUT5 0x4 +#define AB8500_AD_OUT6 0x5 +#define AB8500_AD_OUT7 0x6 +#define AB8500_AD_OUT8 0x7 +#define AB8500_ZEROES 0x8 +#define AB8500_TRISTATE 0xF #define AB8500_ADSLOTSELX_EVEN_SHIFT 0 #define AB8500_ADSLOTSELX_ODD_SHIFT 4 +#define AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(out, slot) \ + ((out) << (((slot) & 1) ? \ + AB8500_ADSLOTSELX_ODD_SHIFT : AB8500_ADSLOTSELX_EVEN_SHIFT)) /* AB8500_ADSLOTHIZCTRL1 */ /* AB8500_ADSLOTHIZCTRL2 */ diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ef2ae32..8d9ba4b 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget ac97_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ac97_routes[] = { + { "AC97 Capture", NULL, "RX" }, + { "TX", NULL, "AC97 Playback" }, +}; + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -62,13 +72,13 @@ static struct snd_soc_dai_driver ac97_dai = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); return 0; } @@ -79,7 +89,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, &soc_ac97_ops, NULL, &ac97_bus); + ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, + &ac97_bus); if (ret < 0) return ret; @@ -116,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, + + .dapm_widgets = ac97_widgets, + .num_dapm_widgets = ARRAY_SIZE(ac97_widgets), + .dapm_routes = ac97_routes, + .num_dapm_routes = ARRAY_SIZE(ac97_routes), }; static int ac97_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index f385342..7257a88 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; +static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_INPUT("CD_L"), +SND_SOC_DAPM_INPUT("CD_R"), +SND_SOC_DAPM_INPUT("AUX_L"), +SND_SOC_DAPM_INPUT("AUX_R"), +SND_SOC_DAPM_INPUT("LINE_IN_L"), +SND_SOC_DAPM_INPUT("LINE_IN_R"), + +SND_SOC_DAPM_OUTPUT("LFE_OUT"), +SND_SOC_DAPM_OUTPUT("CENTER_OUT"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_L"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_R"), +SND_SOC_DAPM_OUTPUT("MONO_OUT"), +SND_SOC_DAPM_OUTPUT("HP_OUT_L"), +SND_SOC_DAPM_OUTPUT("HP_OUT_R"), +}; + +static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { + { "Capture", NULL, "MIC1" }, + { "Capture", NULL, "MIC2" }, + { "Capture", NULL, "CD_L" }, + { "Capture", NULL, "CD_R" }, + { "Capture", NULL, "AUX_L" }, + { "Capture", NULL, "AUX_R" }, + { "Capture", NULL, "LINE_IN_L" }, + { "Capture", NULL, "LINE_IN_R" }, + + { "LFE_OUT", NULL, "Playback" }, + { "CENTER_OUT", NULL, "Playback" }, + { "LINE_OUT_L", NULL, "Playback" }, + { "LINE_OUT_R", NULL, "Playback" }, + { "MONO_OUT", NULL, "Playback" }, + { "HP_OUT_L", NULL, "Playback" }, + { "HP_OUT_R", NULL, "Playback" }, +}; + static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -108,7 +146,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, case AC97_EXTENDED_STATUS: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -124,7 +162,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; @@ -154,13 +192,13 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) u16 retry_cnt = 0; retry: - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, AC97_RESET) == 0x0090) return 1; } - soc_ac97_ops.reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); /* Set bit 16slot in register 74h, then every slot will has only 16 * bits. This command is sent out in 20bit mode, in which case the * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ @@ -186,7 +224,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); return ret; @@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .reg_cache_step = 2, .write = ac97_write, .read = ac97_read, + + .dapm_widgets = ad1980_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), + .dapm_routes = ad1980_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes), }; static int ad1980_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b1f2baf..5fac8ad 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -23,6 +23,21 @@ #include "ad73311.h" +static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINP"), +SND_SOC_DAPM_INPUT("VINN"), +SND_SOC_DAPM_OUTPUT("VOUTN"), +SND_SOC_DAPM_OUTPUT("VOUTP"), +}; + +static const struct snd_soc_dapm_route ad73311_dapm_routes[] = { + { "Capture", NULL, "VINP" }, + { "Capture", NULL, "VINN" }, + + { "VOUTN", NULL, "Playback" }, + { "VOUTP", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ad73311_dai = { .name = "ad73311-hifi", .playback = { @@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -static struct snd_soc_codec_driver soc_codec_dev_ad73311; +static struct snd_soc_codec_driver soc_codec_dev_ad73311 = { + .dapm_widgets = ad73311_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets), + .dapm_routes = ad73311_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes), +}; static int ad73311_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index dafdbe8..ebff112 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -13,6 +13,10 @@ #include <linux/i2c.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -21,16 +25,19 @@ #include "sigmadsp.h" #include "adau1701.h" -#define ADAU1701_DSPCTRL 0x1c -#define ADAU1701_SEROCTL 0x1e -#define ADAU1701_SERICTL 0x1f +#define ADAU1701_DSPCTRL 0x081c +#define ADAU1701_SEROCTL 0x081e +#define ADAU1701_SERICTL 0x081f -#define ADAU1701_AUXNPOW 0x22 +#define ADAU1701_AUXNPOW 0x0822 +#define ADAU1701_PINCONF_0 0x0820 +#define ADAU1701_PINCONF_1 0x0821 +#define ADAU1701_AUXNPOW 0x0822 -#define ADAU1701_OSCIPOW 0x26 -#define ADAU1701_DACSET 0x27 +#define ADAU1701_OSCIPOW 0x0826 +#define ADAU1701_DACSET 0x0827 -#define ADAU1701_NUM_REGS 0x28 +#define ADAU1701_MAX_REGISTER 0x0828 #define ADAU1701_DSPCTRL_CR (1 << 2) #define ADAU1701_DSPCTRL_DAM (1 << 3) @@ -84,10 +91,18 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 +#define ADAU1707_CLKDIV_UNSET (-1U) + #define ADAU1701_FIRMWARE "adau1701.bin" struct adau1701 { + int gpio_nreset; + int gpio_pll_mode[2]; unsigned int dai_fmt; + unsigned int pll_clkdiv; + unsigned int sysclk; + struct regmap *regmap; + u8 pin_config[12]; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -119,10 +134,13 @@ static const struct snd_soc_dapm_route adau1701_dapm_routes[] = { { "ADC", NULL, "IN1" }, }; -static unsigned int adau1701_register_size(struct snd_soc_codec *codec, +static unsigned int adau1701_register_size(struct device *dev, unsigned int reg) { switch (reg) { + case ADAU1701_PINCONF_0: + case ADAU1701_PINCONF_1: + return 3; case ADAU1701_DSPCTRL: case ADAU1701_SEROCTL: case ADAU1701_AUXNPOW: @@ -133,33 +151,42 @@ static unsigned int adau1701_register_size(struct snd_soc_codec *codec, return 1; } - dev_err(codec->dev, "Unsupported register address: %d\n", reg); + dev_err(dev, "Unsupported register address: %d\n", reg); return 0; } -static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) +static bool adau1701_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1701_DACSET: + return true; + default: + return false; + } +} + +static int adau1701_reg_write(void *context, unsigned int reg, + unsigned int value) { + struct i2c_client *client = context; unsigned int i; unsigned int size; - uint8_t buf[4]; + uint8_t buf[5]; int ret; - size = adau1701_register_size(codec, reg); + size = adau1701_register_size(&client->dev, reg); if (size == 0) return -EINVAL; - snd_soc_cache_write(codec, reg, value); - - buf[0] = 0x08; - buf[1] = reg; + buf[0] = reg >> 8; + buf[1] = reg & 0xff; for (i = size + 1; i >= 2; --i) { buf[i] = value; value >>= 8; } - ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2); + ret = i2c_master_send(client, buf, size + 2); if (ret == size + 2) return 0; else if (ret < 0) @@ -168,21 +195,107 @@ static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } -static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) +static int adau1701_reg_read(void *context, unsigned int reg, + unsigned int *value) { - unsigned int value; - unsigned int ret; + int ret; + unsigned int i; + unsigned int size; + uint8_t send_buf[2], recv_buf[3]; + struct i2c_client *client = context; + struct i2c_msg msgs[2]; + + size = adau1701_register_size(&client->dev, reg); + if (size == 0) + return -EINVAL; - ret = snd_soc_cache_read(codec, reg, &value); - if (ret) + send_buf[0] = reg >> 8; + send_buf[1] = reg & 0xff; + + msgs[0].addr = client->addr; + msgs[0].len = sizeof(send_buf); + msgs[0].buf = send_buf; + msgs[0].flags = 0; + + msgs[1].addr = client->addr; + msgs[1].len = size; + msgs[1].buf = recv_buf; + msgs[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); + if (ret < 0) return ret; + else if (ret != ARRAY_SIZE(msgs)) + return -EIO; - return value; + *value = 0; + + for (i = 0; i < size; i++) + *value |= recv_buf[i] << (i * 8); + + return 0; } -static int adau1701_load_firmware(struct snd_soc_codec *codec) +static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) { - return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret; + + if (clkdiv != ADAU1707_CLKDIV_UNSET && + gpio_is_valid(adau1701->gpio_pll_mode[0]) && + gpio_is_valid(adau1701->gpio_pll_mode[1])) { + switch (clkdiv) { + case 64: + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); + break; + case 256: + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); + break; + case 384: + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); + break; + case 0: /* fallback */ + case 512: + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); + break; + } + } + + adau1701->pll_clkdiv = clkdiv; + + if (gpio_is_valid(adau1701->gpio_nreset)) { + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); + /* minimum reset time is 20ns */ + udelay(1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); + /* power-up time may be as long as 85ms */ + mdelay(85); + } + + /* + * Postpone the firmware download to a point in time when we + * know the correct PLL setup + */ + if (clkdiv != ADAU1707_CLKDIV_UNSET) { + ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); + if (ret) { + dev_warn(codec->dev, "Failed to load firmware\n"); + return ret; + } + } + + regmap_write(adau1701->regmap, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + regmap_write(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + + regcache_mark_dirty(adau1701->regmap); + regcache_sync(adau1701->regmap); + + return 0; } static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, @@ -221,7 +334,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; } - snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, mask, val); return 0; } @@ -249,7 +362,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_SERICTL, + regmap_update_bits(adau1701->regmap, ADAU1701_SERICTL, ADAU1701_SERICTL_MODE_MASK, val); return 0; @@ -259,8 +372,22 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int clkdiv = adau1701->sysclk / params_rate(params); snd_pcm_format_t format; unsigned int val; + int ret; + + /* + * If the mclk/lrclk ratio changes, the chip needs updated PLL + * mode GPIO settings, and a full reset cycle, including a new + * firmware upload. + */ + if (clkdiv != adau1701->pll_clkdiv) { + ret = adau1701_reset(codec, clkdiv); + if (ret < 0) + return ret; + } switch (params_rate(params)) { case 192000: @@ -276,7 +403,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); format = params_format(params); @@ -352,8 +479,8 @@ static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - snd_soc_write(codec, ADAU1701_SERICTL, serictl); - snd_soc_update_bits(codec, ADAU1701_SEROCTL, + regmap_write(adau1701->regmap, ADAU1701_SERICTL, serictl); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl); return 0; @@ -363,6 +490,7 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -371,11 +499,13 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Enable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, 0x00); break; case SND_SOC_BIAS_OFF: /* Disable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, mask); break; } @@ -387,6 +517,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; unsigned int mask = ADAU1701_DSPCTRL_DAM; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (mute) @@ -394,7 +525,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) else val = mask; - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, mask, val); return 0; } @@ -403,6 +534,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { unsigned int val; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (clk_id) { case ADAU1701_CLK_SRC_OSC: @@ -415,7 +547,9 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + regmap_update_bits(adau1701->regmap, ADAU1701_OSCIPOW, + ADAU1701_OSCIPOW_OPD, val); + adau1701->sysclk = freq; return 0; } @@ -452,18 +586,45 @@ static struct snd_soc_dai_driver adau1701_dai = { .symmetric_rates = 1, }; +#ifdef CONFIG_OF +static const struct of_device_id adau1701_dt_ids[] = { + { .compatible = "adi,adau1701", }, + { } +}; +MODULE_DEVICE_TABLE(of, adau1701_dt_ids); +#endif + static int adau1701_probe(struct snd_soc_codec *codec) { - int ret; + int i, ret; + unsigned int val; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + + /* + * Let the pll_clkdiv variable default to something that won't happen + * at runtime. That way, we can postpone the firmware download from + * adau1701_reset() to a point in time when we know the correct PLL + * mode parameters. + */ + adau1701->pll_clkdiv = ADAU1707_CLKDIV_UNSET; + + /* initalize with pre-configured pll mode settings */ + ret = adau1701_reset(codec, adau1701->pll_clkdiv); + if (ret < 0) + return ret; + + /* set up pin config */ + val = 0; + for (i = 0; i < 6; i++) + val |= adau1701->pin_config[i] << (i * 4); - codec->control_data = to_i2c_client(codec->dev); + regmap_write(adau1701->regmap, ADAU1701_PINCONF_0, val); - ret = adau1701_load_firmware(codec); - if (ret) - dev_warn(codec->dev, "Failed to load firmware\n"); + val = 0; + for (i = 0; i < 6; i++) + val |= adau1701->pin_config[i + 6] << (i * 4); - snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); - snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + regmap_write(adau1701->regmap, ADAU1701_PINCONF_1, val); return 0; } @@ -473,9 +634,6 @@ static struct snd_soc_codec_driver adau1701_codec_drv = { .set_bias_level = adau1701_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ADAU1701_NUM_REGS, - .reg_word_size = sizeof(u16), - .controls = adau1701_controls, .num_controls = ARRAY_SIZE(adau1701_controls), .dapm_widgets = adau1701_dapm_widgets, @@ -483,22 +641,86 @@ static struct snd_soc_codec_driver adau1701_codec_drv = { .dapm_routes = adau1701_dapm_routes, .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes), - .write = adau1701_write, - .read = adau1701_read, - .set_sysclk = adau1701_set_sysclk, }; +static const struct regmap_config adau1701_regmap = { + .reg_bits = 16, + .val_bits = 32, + .max_register = ADAU1701_MAX_REGISTER, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = adau1701_volatile_reg, + .reg_write = adau1701_reg_write, + .reg_read = adau1701_reg_read, +}; + static int adau1701_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct adau1701 *adau1701; + struct device *dev = &client->dev; + int gpio_nreset = -EINVAL; + int gpio_pll_mode[2] = { -EINVAL, -EINVAL }; int ret; - adau1701 = devm_kzalloc(&client->dev, sizeof(*adau1701), GFP_KERNEL); + adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; + adau1701->regmap = devm_regmap_init(dev, NULL, client, + &adau1701_regmap); + if (IS_ERR(adau1701->regmap)) + return PTR_ERR(adau1701->regmap); + + if (dev->of_node) { + gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); + if (gpio_nreset < 0 && gpio_nreset != -ENOENT) + return gpio_nreset; + + gpio_pll_mode[0] = of_get_named_gpio(dev->of_node, + "adi,pll-mode-gpios", 0); + if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) + return gpio_pll_mode[0]; + + gpio_pll_mode[1] = of_get_named_gpio(dev->of_node, + "adi,pll-mode-gpios", 1); + if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) + return gpio_pll_mode[1]; + + of_property_read_u32(dev->of_node, "adi,pll-clkdiv", + &adau1701->pll_clkdiv); + + of_property_read_u8_array(dev->of_node, "adi,pin-config", + adau1701->pin_config, + ARRAY_SIZE(adau1701->pin_config)); + } + + if (gpio_is_valid(gpio_nreset)) { + ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, + "ADAU1701 Reset"); + if (ret < 0) + return ret; + } + + if (gpio_is_valid(gpio_pll_mode[0]) && + gpio_is_valid(gpio_pll_mode[1])) { + ret = devm_gpio_request_one(dev, gpio_pll_mode[0], + GPIOF_OUT_INIT_LOW, + "ADAU1701 PLL mode 0"); + if (ret < 0) + return ret; + + ret = devm_gpio_request_one(dev, gpio_pll_mode[1], + GPIOF_OUT_INIT_LOW, + "ADAU1701 PLL mode 1"); + if (ret < 0) + return ret; + } + + adau1701->gpio_nreset = gpio_nreset; + adau1701->gpio_pll_mode[0] = gpio_pll_mode[0]; + adau1701->gpio_pll_mode[1] = gpio_pll_mode[1]; + i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); @@ -512,7 +734,10 @@ static int adau1701_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1401", 0 }, + { "adau1401a", 0 }, { "adau1701", 0 }, + { "adau1702", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); @@ -521,6 +746,7 @@ static struct i2c_driver adau1701_i2c_driver = { .driver = { .name = "adau1701", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(adau1701_dt_ids), }, .probe = adau1701_i2c_probe, .remove = adau1701_i2c_remove, diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 3c839cc..15b012d0 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + static int adav80x_spi_probe(struct spi_device *spi) { return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); @@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = { }, .probe = adav80x_spi_probe, .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, }; #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static const struct i2c_device_id adav80x_id[] = { +static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } }; -MODULE_DEVICE_TABLE(i2c, adav80x_id); +MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) @@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = { }, .probe = adav80x_i2c_probe, .remove = adav80x_i2c_remove, - .id_table = adav80x_id, + .id_table = adav80x_i2c_id, }; #endif diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 506d474..8f388ed 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -23,6 +23,28 @@ #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("Input1"), +SND_SOC_DAPM_INPUT("Input2"), +SND_SOC_DAPM_INPUT("Input3"), +SND_SOC_DAPM_INPUT("Input4"), +SND_SOC_DAPM_INPUT("Input5"), +SND_SOC_DAPM_INPUT("Input6"), +SND_SOC_DAPM_INPUT("Input7"), +SND_SOC_DAPM_INPUT("Input8"), +}; + +static const struct snd_soc_dapm_route ads117x_dapm_routes[] = { + { "Capture", NULL, "Input1" }, + { "Capture", NULL, "Input2" }, + { "Capture", NULL, "Input3" }, + { "Capture", NULL, "Input4" }, + { "Capture", NULL, "Input5" }, + { "Capture", NULL, "Input6" }, + { "Capture", NULL, "Input7" }, + { "Capture", NULL, "Input8" }, +}; + static struct snd_soc_dai_driver ads117x_dai = { /* ADC */ .name = "ads117x-hifi", @@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x; +static struct snd_soc_codec_driver soc_codec_dev_ads117x = { + .dapm_widgets = ads117x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), + .dapm_routes = ads117x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes), +}; static int ads117x_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c7cfdf9..71059c0 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -51,6 +51,17 @@ struct ak4104_private { struct regmap *regmap; }; +static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_PGA("TXE", AK4104_REG_TX, AK4104_TX_TXE, 0, NULL, 0), + +SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { + { "TXE", NULL, "Playback" }, + { "TX", NULL, "TXE" }, +}; + static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { @@ -138,29 +149,11 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* enable transmitter */ - ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); - if (ret < 0) - return ret; - return 0; } -static int ak4104_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); - - /* disable transmitter */ - return regmap_update_bits(ak4104->regmap, AK4104_REG_TX, - AK4104_TX_TXE, 0); -} - static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, - .hw_free = ak4104_hw_free, .set_fmt = ak4104_set_dai_fmt, }; @@ -214,6 +207,11 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, + + .dapm_widgets = ak4104_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets), + .dapm_routes = ak4104_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes), }; static const struct regmap_config ak4104_regmap = { diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c new file mode 100644 index 0000000..79e9555 --- /dev/null +++ b/sound/soc/codecs/ak4554.c @@ -0,0 +1,106 @@ +/* + * ak4554.c + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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. + */ + +#include <linux/module.h> +#include <sound/soc.h> + +/* + * ak4554 is very simple DA/AD converter which has no setting register. + * + * CAUTION + * + * ak4554 playback format is SND_SOC_DAIFMT_RIGHT_J, + * and, capture format is SND_SOC_DAIFMT_LEFT_J + * on same bit clock, LR clock. + * But, this driver doesn't have snd_soc_dai_ops :: set_fmt + * + * CPU/Codec DAI image + * + * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554 + * | + * CPU-DAI2 (capture only fmt = LEFT_J) ---+ + */ + +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, + + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, +}; + +static struct snd_soc_dai_driver ak4554_dai = { + .name = "ak4554-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), +}; + +static int ak4554_soc_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ak4554, + &ak4554_dai, 1); +} + +static int ak4554_soc_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct of_device_id ak4554_of_match[] = { + { .compatible = "asahi-kasei,ak4554" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4554_of_match); + +static struct platform_driver ak4554_driver = { + .driver = { + .name = "ak4554-adc-dac", + .owner = THIS_MODULE, + .of_match_table = ak4554_of_match, + }, + .probe = ak4554_soc_probe, + .remove = ak4554_soc_remove, +}; +module_platform_driver(ak4554_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoC AK4554 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f30398..72e953b 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak5386_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), + .dapm_routes = ak5386_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 389f232..657808b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -19,6 +19,7 @@ #include <sound/tlv.h> #include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/gpio.h> #include <linux/mfd/arizona/registers.h> #include "arizona.h" @@ -199,9 +200,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); - if (ret != 0) - return ret; + switch (arizona->type) { + case WM8997: + break; + default: + ret = snd_soc_dapm_new_controls(&codec->dapm, + &arizona_spkr, 1); + if (ret != 0) + return ret; + break; + } ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, "Thermal warning", arizona_thermal_warn, @@ -223,6 +231,41 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +int arizona_init_gpio(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + switch (arizona->type) { + case WM5110: + snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + break; + default: + break; + } + + snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); + + for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) { + switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) { + case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT: + snd_soc_dapm_enable_pin(&codec->dapm, + "DRC1 Signal Activity"); + break; + case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT: + snd_soc_dapm_enable_pin(&codec->dapm, + "DRC2 Signal Activity"); + break; + default: + break; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_gpio); + const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", @@ -517,6 +560,26 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_dmic_osr_text[] = { + "1.536MHz", "3.072MHz", "6.144MHz", +}; + +const struct soc_enum arizona_in_dmic_osr[] = { + SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), +}; +EXPORT_SYMBOL_GPL(arizona_in_dmic_osr); + static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); @@ -1198,6 +1261,13 @@ const struct snd_soc_dai_ops arizona_dai_ops = { }; EXPORT_SYMBOL_GPL(arizona_dai_ops); +const struct snd_soc_dai_ops arizona_simple_dai_ops = { + .startup = arizona_startup, + .hw_params = arizona_hw_params_rate, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_simple_dai_ops); + int arizona_init_dai(struct arizona_priv *priv, int id) { struct arizona_dai_priv *dai_priv = &priv->dai[id]; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index af39f10..9e81b63 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -57,7 +57,7 @@ #define ARIZONA_CLK_98MHZ 5 #define ARIZONA_CLK_147MHZ 6 -#define ARIZONA_MAX_DAI 4 +#define ARIZONA_MAX_DAI 6 #define ARIZONA_MAX_ADSP 4 struct arizona; @@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \ ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux) -#define ARIZONA_MUX_ROUTES(name) \ +#define ARIZONA_MUX_ROUTES(widget, name) \ + { widget, NULL, name " Input" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Input") #define ARIZONA_MIXER_ROUTES(widget, name) \ @@ -198,6 +199,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, @@ -213,6 +215,7 @@ extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); extern const struct snd_soc_dai_ops arizona_dai_ops; +extern const struct snd_soc_dai_ops arizona_simple_dai_ops; #define ARIZONA_FLL_NAME_LEN 20 @@ -241,6 +244,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); extern int arizona_init_spk(struct snd_soc_codec *codec); +extern int arizona_init_gpio(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c new file mode 100644 index 0000000..c4cf069 --- /dev/null +++ b/sound/soc/codecs/bt-sco.c @@ -0,0 +1,91 @@ +/* + * Driver for generic Bluetooth SCO link + * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de> + * + * 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/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/soc.h> + +static const struct snd_soc_dapm_widget bt_sco_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route bt_sco_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; + +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", + .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, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_bt_sco = { + .dapm_widgets = bt_sco_widgets, + .num_dapm_widgets = ARRAY_SIZE(bt_sco_widgets), + .dapm_routes = bt_sco_routes, + .num_dapm_routes = ARRAY_SIZE(bt_sco_routes), +}; + +static int bt_sco_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); +} + +static int bt_sco_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + { + .name = "bt-sco", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { + .driver = { + .name = "bt-sco", + .owner = THIS_MODULE, + }, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, +}; + +module_platform_driver(bt_sco_driver); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8e47798..83c835d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -139,6 +139,22 @@ struct cs4270_private { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; +static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA", NULL, "Playback" }, + { "AOUTB", NULL, "Playback" }, +}; + /** * struct cs4270_mode_ratios - clock ratio tables * @ratio: the ratio of MCLK to the sample rate @@ -612,6 +628,10 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .controls = cs4270_snd_controls, .num_controls = ARRAY_SIZE(cs4270_snd_controls), + .dapm_widgets = cs4270_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4270_dapm_widgets), + .dapm_routes = cs4270_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4270_dapm_routes), }; /* diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 03036b3..a20f1bb 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -173,6 +173,26 @@ struct cs4271_private { bool enable_soft_reset; }; +static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINA"), +SND_SOC_DAPM_INPUT("AINB"), + +SND_SOC_DAPM_OUTPUT("AOUTA+"), +SND_SOC_DAPM_OUTPUT("AOUTA-"), +SND_SOC_DAPM_OUTPUT("AOUTB+"), +SND_SOC_DAPM_OUTPUT("AOUTB-"), +}; + +static const struct snd_soc_dapm_route cs4271_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA+", NULL, "Playback" }, + { "AOUTA-", NULL, "Playback" }, + { "AOUTB+", NULL, "Playback" }, + { "AOUTB-", NULL, "Playback" }, +}; + /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -576,8 +596,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) CS4271_MODE2_MUTECAEQUB, CS4271_MODE2_MUTECAEQUB); - return snd_soc_add_codec_controls(codec, cs4271_snd_controls, - ARRAY_SIZE(cs4271_snd_controls)); + return 0; } static int cs4271_remove(struct snd_soc_codec *codec) @@ -596,6 +615,13 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, + + .controls = cs4271_snd_controls, + .num_controls = ARRAY_SIZE(cs4271_snd_controls), + .dapm_widgets = cs4271_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4271_dapm_widgets), + .dapm_routes = cs4271_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728..be2ba1b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, + 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c deleted file mode 100644 index 4f4f7f4..0000000 --- a/sound/soc/codecs/dfbmcs320.c +++ /dev/null @@ -1,62 +0,0 @@ -/* - * Driver for the DFBM-CS320 bluetooth module - * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de> - * - * 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/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> - -#include <sound/soc.h> - -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}; - -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; - -static int dfbmcs320_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); -} - -static int dfbmcs320_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - - return 0; -} - -static struct platform_driver dfmcs320_driver = { - .driver = { - .name = "dfbmcs320", - .owner = THIS_MODULE, - }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, -}; - -module_platform_driver(dfmcs320_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 66967ba..b2090b2 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"DMIC AIF", NULL, "DMic"}, }; -static int dmic_probe(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, - ARRAY_SIZE(dmic_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - static struct snd_soc_codec_driver soc_dmic = { - .probe = dmic_probe, + .dapm_widgets = dmic_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int dmic_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/hdmi.c index 529d064..68342b1 100644 --- a/sound/soc/codecs/omap-hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -1,5 +1,5 @@ /* - * ALSA SoC codec driver for HDMI audio on OMAP processors. + * ALSA SoC codec driver for HDMI audio codecs. * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ * Author: Ricardo Neri <ricardo.neri@ti.com> * @@ -23,11 +23,20 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver omap_hdmi_codec; +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; -static struct snd_soc_dai_driver omap_hdmi_codec_dai = { - .name = "omap-hdmi-hifi", +static struct snd_soc_dai_driver hdmi_codec_dai = { + .name = "hdmi-hifi", .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | @@ -37,33 +46,52 @@ static struct snd_soc_dai_driver omap_hdmi_codec_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, + +}; + +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), }; -static int omap_hdmi_codec_probe(struct platform_device *pdev) +static int hdmi_codec_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, - &omap_hdmi_codec_dai, 1); + return snd_soc_register_codec(&pdev->dev, &hdmi_codec, + &hdmi_codec_dai, 1); } -static int omap_hdmi_codec_remove(struct platform_device *pdev) +static int hdmi_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver omap_hdmi_codec_driver = { +static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, }, - .probe = omap_hdmi_codec_probe, - .remove = omap_hdmi_codec_remove, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, }; -module_platform_driver(omap_hdmi_codec_driver); +module_platform_driver(hdmi_codec_driver); MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); -MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 5f607b3..bcebd1a 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -384,8 +384,6 @@ static int jz4740_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); - platform_set_drvdata(pdev, NULL); - return 0; } diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f595..0e5743e 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> @@ -23,12 +24,15 @@ #include <sound/tlv.h> struct lm4857 { - struct i2c_client *i2c; + struct regmap *regmap; uint8_t mode; }; -static const uint8_t lm4857_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, +static const struct reg_default lm4857_default_regs[] = { + { 0x0, 0x00 }, + { 0x1, 0x00 }, + { 0x2, 0x00 }, + { 0x3, 0x00 }, }; /* The register offsets in the cache array */ @@ -42,39 +46,6 @@ static const uint8_t lm4857_default_regs[] = { #define LM4857_WAKEUP 5 #define LM4857_EPGAIN 4 -static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - uint8_t data; - int ret; - - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return ret; - - data = (reg << 6) | value; - ret = i2c_master_send(codec->control_data, &data, 1); - if (ret != 1) { - dev_err(codec->dev, "Failed to write register: %d\n", ret); - return ret; - } - - return 0; -} - -static unsigned int lm4857_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned int val; - int ret; - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret) - return -1; - - return val; -} - static int lm4857_get_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -96,7 +67,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, lm4857->mode = value; if (codec->dapm.bias_level == SND_SOC_BIAS_ON) - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6); return 1; } @@ -108,10 +79,11 @@ static int lm4857_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, + lm4857->mode + 6); break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0); break; default: break; @@ -171,49 +143,32 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { {"EP", NULL, "IN"}, }; -static int lm4857_probe(struct snd_soc_codec *codec) -{ - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - codec->control_data = lm4857->i2c; - - ret = snd_soc_add_codec_controls(codec, lm4857_controls, - ARRAY_SIZE(lm4857_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, - ARRAY_SIZE(lm4857_dapm_widgets)); - if (ret) - return ret; +static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { + .set_bias_level = lm4857_set_bias_level, - ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, - ARRAY_SIZE(lm4857_routes)); - if (ret) - return ret; + .controls = lm4857_controls, + .num_controls = ARRAY_SIZE(lm4857_controls), + .dapm_widgets = lm4857_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets), + .dapm_routes = lm4857_routes, + .num_dapm_routes = ARRAY_SIZE(lm4857_routes), +}; - snd_soc_dapm_new_widgets(dapm); +static const struct regmap_config lm4857_regmap_config = { + .val_bits = 6, + .reg_bits = 2, - return 0; -} + .max_register = LM4857_CTRL, -static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { - .write = lm4857_write, - .read = lm4857_read, - .probe = lm4857_probe, - .reg_cache_size = ARRAY_SIZE(lm4857_default_regs), - .reg_word_size = sizeof(uint8_t), - .reg_cache_default = lm4857_default_regs, - .set_bias_level = lm4857_set_bias_level, + .cache_type = REGCACHE_FLAT, + .reg_defaults = lm4857_default_regs, + .num_reg_defaults = ARRAY_SIZE(lm4857_default_regs), }; static int lm4857_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct lm4857 *lm4857; - int ret; lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL); if (!lm4857) @@ -221,11 +176,11 @@ static int lm4857_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, lm4857); - lm4857->i2c = i2c; - - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); + lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); + if (IS_ERR(lm4857->regmap)) + return PTR_ERR(lm4857->regmap); - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); } static int lm4857_i2c_remove(struct i2c_client *i2c) diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index a6ac231..31f9156 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -118,6 +118,18 @@ static const struct snd_kcontrol_new max9768_mute[] = { SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; +static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN"), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +}; + +static const struct snd_soc_dapm_route max9768_dapm_routes[] = { + { "OUT+", NULL, "IN" }, + { "OUT-", NULL, "IN" }, +}; + static int max9768_probe(struct snd_soc_codec *codec) { struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); @@ -148,6 +160,10 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .probe = max9768_probe, .controls = max9768_volume, .num_controls = ARRAY_SIZE(max9768_volume), + .dapm_widgets = max9768_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets), + .dapm_routes = max9768_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes), }; static const struct regmap_config max9768_i2c_regmap_config = { diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada5..566a367 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 8d14a76..0569a4c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -857,6 +857,14 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); +static const char *dmic_mux_text[] = { "ADC", "DMIC" }; + +static const struct soc_enum dmic_mux_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_mux_text), dmic_mux_text); + +static const struct snd_kcontrol_new max98090_dmic_mux = + SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum); + static const char *max98090_micpre_text[] = { "Off", "On" }; static const struct soc_enum max98090_pa1en_enum = @@ -1144,6 +1152,9 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIC2 Mux", SND_SOC_NOPM, 0, 0, &max98090_mic2_mux), + SND_SOC_DAPM_VIRT_MUX("DMIC Mux", SND_SOC_NOPM, + 0, 0, &max98090_dmic_mux), + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -1336,11 +1347,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"ADCL", NULL, "SHDN"}, {"ADCR", NULL, "SHDN"}, - {"LBENL Mux", "Normal", "ADCL"}, - {"LBENL Mux", "Normal", "DMICL"}, + {"DMIC Mux", "ADC", "ADCL"}, + {"DMIC Mux", "ADC", "ADCR"}, + {"DMIC Mux", "DMIC", "DMICL"}, + {"DMIC Mux", "DMIC", "DMICR"}, + + {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, - {"LBENR Mux", "Normal", "ADCR"}, - {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Normal", "DMIC Mux"}, {"LBENR Mux", "Loopback", "LTENR Mux"}, {"AIFOUTL", NULL, "LBENL Mux"}, @@ -2070,8 +2084,9 @@ static irqreturn_t max98090_interrupt(int irq, void *data) pm_wakeup_event(codec->dev, 100); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); } if (active & M98090_DRCACT_MASK) @@ -2118,8 +2133,9 @@ int max98090_mic_detect(struct snd_soc_codec *codec, snd_soc_jack_report(max98090->jack, 0, SND_JACK_HEADSET | SND_JACK_BTN_0); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); return 0; } @@ -2336,6 +2352,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2355,6 +2372,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74c..29549cd 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -14,170 +14,21 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <sound/tlv.h> #include "max9877.h" -static struct i2c_client *i2c; +static struct regmap *regmap; -static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; - -static void max9877_write_regs(void) -{ - unsigned int i; - u8 data[6]; - - data[0] = MAX9877_INPUT_MODE; - for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) - data[i + 1] = max9877_regs[i]; - - if (i2c_master_send(i2c, data, 6) != 6) - dev_err(&i2c->dev, "i2c write failed\n"); -} - -static int max9877_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int invert = mc->invert; - - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - - if (invert) - ucontrol->value.integer.value[0] = - mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int max9877_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int invert = mc->invert; - unsigned int val = (ucontrol->value.integer.value[0] & mask); - - if (invert) - val = mask - val; - - if (((max9877_regs[reg] >> shift) & mask) == val) - return 0; - - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_write_regs(); - - return 1; -} - -static int max9877_get_2reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; - - return 0; -} - -static int max9877_set_2reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int val = (ucontrol->value.integer.value[0] & mask); - unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 0; - - if (((max9877_regs[reg] >> shift) & mask) != val) - change = 1; - - if (((max9877_regs[reg2] >> shift) & mask) != val2) - change = 1; - - if (change) { - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= val2 << shift; - max9877_write_regs(); - } - - return change; -} - -static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; - - if (value) - value -= 1; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - value += 1; - - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) - return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; -} - -static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); - - value = value >> MAX9877_OSC_OFFSET; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - value = value << MAX9877_OSC_OFFSET; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) - return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; -} +static struct reg_default max9877_regs[] = { + { 0, 0x40 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x49 }, +}; static const unsigned int max9877_pgain_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -212,65 +63,104 @@ static const char *max9877_osc_mode[] = { }; static const struct soc_enum max9877_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), + SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, 0, ARRAY_SIZE(max9877_out_mode), + max9877_out_mode), + SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, MAX9877_OSC_OFFSET, + ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), }; static const struct snd_kcontrol_new max9877_controls[] = { - SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", - MAX9877_INPUT_MODE, 0, 2, 0, - max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), - SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", - MAX9877_INPUT_MODE, 2, 2, 0, - max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), - SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", - MAX9877_SPK_VOLUME, 0, 31, 0, - max9877_get_reg, max9877_set_reg, max9877_output_tlv), - SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", - MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, - max9877_get_2reg, max9877_set_2reg, max9877_output_tlv), - SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", - MAX9877_INPUT_MODE, 4, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", - MAX9877_INPUT_MODE, 5, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", - MAX9877_INPUT_MODE, 6, 1, 0, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", - MAX9877_OUTPUT_MODE, 6, 1, 0, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", - MAX9877_OUTPUT_MODE, 7, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], - max9877_get_out_mode, max9877_set_out_mode), - SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], - max9877_get_osc_mode, max9877_set_osc_mode), + SOC_SINGLE_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, max9877_pgain_tlv), + SOC_SINGLE_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, max9877_pgain_tlv), + SOC_SINGLE_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, max9877_output_tlv), + SOC_DOUBLE_R_TLV("MAX9877 Amp HP Playback Volume", + MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_output_tlv), + SOC_SINGLE("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1), + SOC_SINGLE("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1), + SOC_SINGLE("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0), + SOC_SINGLE("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0), + SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]), + SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]), }; -/* This function is called from ASoC machine driver */ -int max9877_add_controls(struct snd_soc_codec *codec) -{ - return snd_soc_add_codec_controls(codec, max9877_controls, - ARRAY_SIZE(max9877_controls)); -} -EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct snd_soc_dapm_widget max9877_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("INA1"), +SND_SOC_DAPM_INPUT("INA2"), +SND_SOC_DAPM_INPUT("INB1"), +SND_SOC_DAPM_INPUT("INB2"), +SND_SOC_DAPM_INPUT("RXIN+"), +SND_SOC_DAPM_INPUT("RXIN-"), + +SND_SOC_DAPM_PGA("SHDN", MAX9877_OUTPUT_MODE, 7, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_route max9877_dapm_routes[] = { + { "SHDN", NULL, "INA1" }, + { "SHDN", NULL, "INA2" }, + { "SHDN", NULL, "INB1" }, + { "SHDN", NULL, "INB2" }, + + { "OUT+", NULL, "RXIN+" }, + { "OUT+", NULL, "SHDN" }, + + { "OUT-", NULL, "SHDN" }, + { "OUT-", NULL, "RXIN-" }, + + { "HPL", NULL, "SHDN" }, + { "HPR", NULL, "SHDN" }, +}; + +static const struct snd_soc_codec_driver max9877_codec = { + .controls = max9877_controls, + .num_controls = ARRAY_SIZE(max9877_controls), + + .dapm_widgets = max9877_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9877_dapm_widgets), + .dapm_routes = max9877_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9877_dapm_routes), +}; + +static const struct regmap_config max9877_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max9877_regs, + .num_reg_defaults = ARRAY_SIZE(max9877_regs), + .cache_type = REGCACHE_RBTREE, +}; static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - i2c = client; + int i; - max9877_write_regs(); + regmap = devm_regmap_init_i2c(client, &max9877_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - return 0; + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); + + return snd_soc_register_codec(&client->dev, &max9877_codec, NULL, 0); } static int max9877_i2c_remove(struct i2c_client *client) { - i2c = NULL; + snd_soc_unregister_codec(&client->dev); return 0; } diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 5402dfb..ea141e1 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -94,7 +94,6 @@ #define AUDIO_DAC_CFS_DLY_B (1 << 10) struct mc13783_priv { - struct snd_soc_codec codec; struct mc13xxx *mc13xxx; enum mc13783_ssi_port adc_ssi_port; @@ -126,6 +125,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..651ce09 --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,339 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko@streamunlimited.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. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static const int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, enable = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + if (val != -1) { + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + enable = 1; + } else + enable = 0; + + /* enable/disable deemphasis functionality */ + return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, enable); +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + else + val = 0; + + return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + if (ret < 0) + return ret; + + return pcm1681_set_deemph(codec); +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUT1"), +SND_SOC_DAPM_OUTPUT("VOUT2"), +SND_SOC_DAPM_OUTPUT("VOUT3"), +SND_SOC_DAPM_OUTPUT("VOUT4"), +SND_SOC_DAPM_OUTPUT("VOUT5"), +SND_SOC_DAPM_OUTPUT("VOUT6"), +SND_SOC_DAPM_OUTPUT("VOUT7"), +SND_SOC_DAPM_OUTPUT("VOUT8"), +}; + +static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = { + { "VOUT1", NULL, "Playback" }, + { "VOUT2", NULL, "Playback" }, + { "VOUT3", NULL, "Playback" }, + { "VOUT4", NULL, "Playback" }, + { "VOUT5", NULL, "Playback" }, + { "VOUT6", NULL, "Playback" }, + { "VOUT7", NULL, "Playback" }, + { "VOUT8", NULL, "Playback" }, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +static struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), + .dapm_widgets = pcm1681_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets), + .dapm_routes = pcm1681_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes), +}; + +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko@streamunlimited.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c new file mode 100644 index 0000000..2a8eccf --- /dev/null +++ b/sound/soc/codecs/pcm1792a.c @@ -0,0 +1,257 @@ +/* + * PCM1792A ASoC codec driver + * + * Copyright (c) Amarula Solutions B.V. 2013 + * + * Michael Trimarchi <michael@amarulasolutions.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. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/spi/spi.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/of_device.h> + +#include "pcm1792a.h" + +#define PCM1792A_DAC_VOL_LEFT 0x10 +#define PCM1792A_DAC_VOL_RIGHT 0x11 +#define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL + +#define PCM1792A_FMT_MASK 0x70 +#define PCM1792A_FMT_SHIFT 4 +#define PCM1792A_MUTE_MASK 0x01 +#define PCM1792A_MUTE_SHIFT 0 +#define PCM1792A_ATLD_ENABLE (1 << 7) + +static const struct reg_default pcm1792a_reg_defaults[] = { + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x50 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x01 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg) +{ + return reg >= 0x10 && reg <= 0x17; +} + +static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg) +{ + bool accessible; + + accessible = pcm1792a_accessible_reg(dev, reg); + + return accessible && reg != 0x16 && reg != 0x17; +} + +struct pcm1792a_private { + struct regmap *regmap; + unsigned int format; + unsigned int rate; +}; + +static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->format = format; + + return 0; +} + +static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + PCM1792A_MUTE_MASK, !!mute); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1792a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x02; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x05; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x04; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; + + ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_soc_dai_ops pcm1792a_dai_ops = { + .set_fmt = pcm1792a_set_dai_fmt, + .hw_params = pcm1792a_hw_params, + .digital_mute = pcm1792a_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1); + +static const struct snd_kcontrol_new pcm1792a_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, + PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, + pcm1792a_dac_tlv), +}; + +static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("IOUTL+"), +SND_SOC_DAPM_OUTPUT("IOUTL-"), +SND_SOC_DAPM_OUTPUT("IOUTR+"), +SND_SOC_DAPM_OUTPUT("IOUTR-"), +}; + +static const struct snd_soc_dapm_route pcm1792a_dapm_routes[] = { + { "IOUTL+", NULL, "Playback" }, + { "IOUTL-", NULL, "Playback" }, + { "IOUTR+", NULL, "Playback" }, + { "IOUTR-", NULL, "Playback" }, +}; + +static struct snd_soc_dai_driver pcm1792a_dai = { + .name = "pcm1792a-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM1792A_RATES, + .formats = PCM1792A_FORMATS, }, + .ops = &pcm1792a_dai_ops, +}; + +static const struct of_device_id pcm1792a_of_match[] = { + { .compatible = "ti,pcm1792a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1792a_of_match); + +static const struct regmap_config pcm1792a_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 24, + .reg_defaults = pcm1792a_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), + .writeable_reg = pcm1792a_writeable_reg, + .readable_reg = pcm1792a_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { + .controls = pcm1792a_controls, + .num_controls = ARRAY_SIZE(pcm1792a_controls), + .dapm_widgets = pcm1792a_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1792a_dapm_widgets), + .dapm_routes = pcm1792a_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1792a_dapm_routes), +}; + +static int pcm1792a_spi_probe(struct spi_device *spi) +{ + struct pcm1792a_private *pcm1792a; + int ret; + + pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private), + GFP_KERNEL); + if (!pcm1792a) + return -ENOMEM; + + spi_set_drvdata(spi, pcm1792a); + + pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap); + if (IS_ERR(pcm1792a->regmap)) { + ret = PTR_ERR(pcm1792a->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&spi->dev, + &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1); +} + +static int pcm1792a_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm1792a_spi_ids[] = { + { "pcm1792a", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids); + +static struct spi_driver pcm1792a_codec_driver = { + .driver = { + .name = "pcm1792a", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1792a_of_match), + }, + .id_table = pcm1792a_spi_ids, + .probe = pcm1792a_spi_probe, + .remove = pcm1792a_spi_remove, +}; + +module_spi_driver(pcm1792a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1792A driver"); +MODULE_AUTHOR("Michael Trimarchi <michael@amarulasolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h new file mode 100644 index 0000000..7a83d1f --- /dev/null +++ b/sound/soc/codecs/pcm1792a.h @@ -0,0 +1,26 @@ +/* + * definitions for PCM1792A + * + * Copyright 2013 Amarula Solutions + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __PCM1792A_H__ +#define __PCM1792A_H__ + +#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE) + +#endif diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f2a6282..b6618c4 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,7 +28,54 @@ #include "pcm3008.h" -#define PCM3008_VERSION "0.2" +static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value_cansleep(setup->pdda_pin, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value_cansleep(setup->pdad_pin, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_dac_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_adc_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "ADC" }, + { "ADC", NULL, "VINL" }, + { "ADC", NULL, "VINR" }, + + { "DAC", NULL, "PCM3008 Playback" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -51,20 +98,20 @@ static struct snd_soc_dai_driver pcm3008_dai = { }, }; -static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) -{ - gpio_free(setup->dem0_pin); - gpio_free(setup->dem1_pin); - gpio_free(setup->pdad_pin); - gpio_free(setup->pdda_pin); -} +static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), +}; -static int pcm3008_soc_probe(struct snd_soc_codec *codec) +static int pcm3008_codec_probe(struct platform_device *pdev) { - struct pcm3008_setup_data *setup = codec->dev->platform_data; - int ret = 0; + struct pcm3008_setup_data *setup = pdev->dev.platform_data; + int ret; - printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); + if (!setup) + return -EINVAL; /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON @@ -74,83 +121,29 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) */ /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ - ret = gpio_request(setup->dem0_pin, "codec_dem0"); - if (ret == 0) - ret = gpio_direction_output(setup->dem0_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->dem0_pin, + GPIOF_OUT_INIT_HIGH, "codec_dem0"); if (ret != 0) - goto gpio_err; + return ret; /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ - ret = gpio_request(setup->dem1_pin, "codec_dem1"); - if (ret == 0) - ret = gpio_direction_output(setup->dem1_pin, 0); + ret = devm_gpio_request_one(&pdev->dev, setup->dem1_pin, + GPIOF_OUT_INIT_LOW, "codec_dem1"); if (ret != 0) - goto gpio_err; + return ret; /* Configure PDAD GPIO. */ - ret = gpio_request(setup->pdad_pin, "codec_pdad"); - if (ret == 0) - ret = gpio_direction_output(setup->pdad_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin, + GPIOF_OUT_INIT_LOW, "codec_pdad"); if (ret != 0) - goto gpio_err; + return ret; /* Configure PDDA GPIO. */ - ret = gpio_request(setup->pdda_pin, "codec_pdda"); - if (ret == 0) - ret = gpio_direction_output(setup->pdda_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin, + GPIOF_OUT_INIT_LOW, "codec_pdda"); if (ret != 0) - goto gpio_err; - - return ret; - -gpio_err: - pcm3008_gpio_free(setup); + return ret; - return ret; -} - -static int pcm3008_soc_remove(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - pcm3008_gpio_free(setup); - return 0; -} - -#ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value(setup->pdad_pin, 0); - gpio_set_value(setup->pdda_pin, 0); - - return 0; -} - -static int pcm3008_soc_resume(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value(setup->pdad_pin, 1); - gpio_set_value(setup->pdda_pin, 1); - - return 0; -} -#else -#define pcm3008_soc_suspend NULL -#define pcm3008_soc_resume NULL -#endif - -static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { - .probe = pcm3008_soc_probe, - .remove = pcm3008_soc_remove, - .suspend = pcm3008_soc_suspend, - .resume = pcm3008_soc_resume, -}; - -static int pcm3008_codec_probe(struct platform_device *pdev) -{ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3008, &pcm3008_dai, 1); } @@ -158,6 +151,7 @@ static int pcm3008_codec_probe(struct platform_device *pdev) static int pcm3008_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); + return 0; } diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c new file mode 100644 index 0000000..c26a8f8 --- /dev/null +++ b/sound/soc/codecs/rt5640.c @@ -0,0 +1,2211 @@ +/* + * rt5640.c -- RT5640 ALSA SoC audio codec driver + * + * Copyright 2011 Realtek Semiconductor Corp. + * Author: Johnny Hsu <johnnyhsu@realtek.com> + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.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/initval.h> +#include <sound/tlv.h> + +#include "rt5640.h" + +#define RT5640_DEVICE_ID 0x6231 + +#define RT5640_PR_RANGE_BASE (0xff + 1) +#define RT5640_PR_SPACING 0x100 + +#define RT5640_PR_BASE (RT5640_PR_RANGE_BASE + (0 * RT5640_PR_SPACING)) + +static const struct regmap_range_cfg rt5640_ranges[] = { + { .name = "PR", .range_min = RT5640_PR_BASE, + .range_max = RT5640_PR_BASE + 0xb4, + .selector_reg = RT5640_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5640_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + {RT5640_PR_BASE + 0x3d, 0x3600}, + {RT5640_PR_BASE + 0x12, 0x0aa8}, + {RT5640_PR_BASE + 0x14, 0x0aaa}, + {RT5640_PR_BASE + 0x20, 0x6110}, + {RT5640_PR_BASE + 0x21, 0xe0e0}, + {RT5640_PR_BASE + 0x23, 0x1804}, +}; +#define RT5640_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5640_reg[RT5640_VENDOR_ID2 + 1] = { + { 0x00, 0x000e }, + { 0x01, 0xc8c8 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x04, 0x8000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0000 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x7060 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5454 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaa00 }, + { 0x2d, 0x0000 }, + { 0x2e, 0xa000 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x47, 0x003e }, + { 0x48, 0xf800 }, + { 0x49, 0x3800 }, + { 0x4a, 0x0004 }, + { 0x4c, 0xfc00 }, + { 0x4d, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x53, 0xf000 }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6c, 0x0000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1114 }, + { 0x74, 0x0c00 }, + { 0x75, 0x1d00 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0008 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0600 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c00 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb8, 0x034b }, + { 0xb9, 0x0066 }, + { 0xba, 0x000b }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0400 }, + { 0xc2, 0x0000 }, + { 0xc4, 0x0000 }, + { 0xc5, 0x0000 }, + { 0xc6, 0x2000 }, + { 0xc8, 0x0000 }, + { 0xc9, 0x0000 }, + { 0xca, 0x0000 }, + { 0xcb, 0x0000 }, + { 0xcc, 0x0000 }, + { 0xcf, 0x0013 }, + { 0xd0, 0x0680 }, + { 0xd1, 0x1c17 }, + { 0xd2, 0x8c00 }, + { 0xd3, 0xaa20 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6231 }, +}; + +static int rt5640_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5640_RESET, 0); +} + +static bool rt5640_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5640_ranges); i++) + if ((reg >= rt5640_ranges[i].window_start && + reg <= rt5640_ranges[i].window_start + + rt5640_ranges[i].window_len) || + (reg >= rt5640_ranges[i].range_min && + reg <= rt5640_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5640_RESET: + case RT5640_ASRC_5: + case RT5640_EQ_CTRL1: + case RT5640_DRC_AGC_1: + case RT5640_ANC_CTRL1: + case RT5640_IRQ_CTRL2: + case RT5640_INT_IRQ_ST: + case RT5640_DSP_CTRL2: + case RT5640_DSP_CTRL3: + case RT5640_PRIV_INDEX: + case RT5640_PRIV_DATA: + case RT5640_PGM_REG_ARR1: + case RT5640_PGM_REG_ARR3: + case RT5640_VENDOR_ID: + case RT5640_VENDOR_ID1: + case RT5640_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5640_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5640_ranges); i++) + if ((reg >= rt5640_ranges[i].window_start && + reg <= rt5640_ranges[i].window_start + + rt5640_ranges[i].window_len) || + (reg >= rt5640_ranges[i].range_min && + reg <= rt5640_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5640_RESET: + case RT5640_SPK_VOL: + case RT5640_HP_VOL: + case RT5640_OUTPUT: + case RT5640_MONO_OUT: + case RT5640_IN1_IN2: + case RT5640_IN3_IN4: + case RT5640_INL_INR_VOL: + case RT5640_DAC1_DIG_VOL: + case RT5640_DAC2_DIG_VOL: + case RT5640_DAC2_CTRL: + case RT5640_ADC_DIG_VOL: + case RT5640_ADC_DATA: + case RT5640_ADC_BST_VOL: + case RT5640_STO_ADC_MIXER: + case RT5640_MONO_ADC_MIXER: + case RT5640_AD_DA_MIXER: + case RT5640_STO_DAC_MIXER: + case RT5640_MONO_DAC_MIXER: + case RT5640_DIG_MIXER: + case RT5640_DSP_PATH1: + case RT5640_DSP_PATH2: + case RT5640_DIG_INF_DATA: + case RT5640_REC_L1_MIXER: + case RT5640_REC_L2_MIXER: + case RT5640_REC_R1_MIXER: + case RT5640_REC_R2_MIXER: + case RT5640_HPO_MIXER: + case RT5640_SPK_L_MIXER: + case RT5640_SPK_R_MIXER: + case RT5640_SPO_L_MIXER: + case RT5640_SPO_R_MIXER: + case RT5640_SPO_CLSD_RATIO: + case RT5640_MONO_MIXER: + case RT5640_OUT_L1_MIXER: + case RT5640_OUT_L2_MIXER: + case RT5640_OUT_L3_MIXER: + case RT5640_OUT_R1_MIXER: + case RT5640_OUT_R2_MIXER: + case RT5640_OUT_R3_MIXER: + case RT5640_LOUT_MIXER: + case RT5640_PWR_DIG1: + case RT5640_PWR_DIG2: + case RT5640_PWR_ANLG1: + case RT5640_PWR_ANLG2: + case RT5640_PWR_MIXER: + case RT5640_PWR_VOL: + case RT5640_PRIV_INDEX: + case RT5640_PRIV_DATA: + case RT5640_I2S1_SDP: + case RT5640_I2S2_SDP: + case RT5640_ADDA_CLK1: + case RT5640_ADDA_CLK2: + case RT5640_DMIC: + case RT5640_GLB_CLK: + case RT5640_PLL_CTRL1: + case RT5640_PLL_CTRL2: + case RT5640_ASRC_1: + case RT5640_ASRC_2: + case RT5640_ASRC_3: + case RT5640_ASRC_4: + case RT5640_ASRC_5: + case RT5640_HP_OVCD: + case RT5640_CLS_D_OVCD: + case RT5640_CLS_D_OUT: + case RT5640_DEPOP_M1: + case RT5640_DEPOP_M2: + case RT5640_DEPOP_M3: + case RT5640_CHARGE_PUMP: + case RT5640_PV_DET_SPK_G: + case RT5640_MICBIAS: + case RT5640_EQ_CTRL1: + case RT5640_EQ_CTRL2: + case RT5640_WIND_FILTER: + case RT5640_DRC_AGC_1: + case RT5640_DRC_AGC_2: + case RT5640_DRC_AGC_3: + case RT5640_SVOL_ZC: + case RT5640_ANC_CTRL1: + case RT5640_ANC_CTRL2: + case RT5640_ANC_CTRL3: + case RT5640_JD_CTRL: + case RT5640_ANC_JD: + case RT5640_IRQ_CTRL1: + case RT5640_IRQ_CTRL2: + case RT5640_INT_IRQ_ST: + case RT5640_GPIO_CTRL1: + case RT5640_GPIO_CTRL2: + case RT5640_GPIO_CTRL3: + case RT5640_DSP_CTRL1: + case RT5640_DSP_CTRL2: + case RT5640_DSP_CTRL3: + case RT5640_DSP_CTRL4: + case RT5640_PGM_REG_ARR1: + case RT5640_PGM_REG_ARR2: + case RT5640_PGM_REG_ARR3: + case RT5640_PGM_REG_ARR4: + case RT5640_PGM_REG_ARR5: + case RT5640_SCB_FUNC: + case RT5640_SCB_CTRL: + case RT5640_BASE_BACK: + case RT5640_MP3_PLUS1: + case RT5640_MP3_PLUS2: + case RT5640_3D_HP: + case RT5640_ADJ_HPF: + case RT5640_HP_CALIB_AMP_DET: + case RT5640_HP_CALIB2: + case RT5640_SV_ZCD1: + case RT5640_SV_ZCD2: + case RT5640_DUMMY1: + case RT5640_DUMMY2: + case RT5640_DUMMY3: + case RT5640_VENDOR_ID: + case RT5640_VENDOR_ID1: + case RT5640_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5640_data_select[] = { + "Normal", "left copy to right", "right copy to left", "Swap"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); + +/* Class D speaker gain ratio */ +static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", + "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, + RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); + +static const struct snd_kcontrol_new rt5640_snd_controls[] = { + /* Speaker Output Volume */ + SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, + RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* Headphone Output Volume */ + SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, + RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5640_OUTPUT, + RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5640_OUTPUT, + RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, + RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, + RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, + RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5640_ADC_DIG_VOL, + RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5640_ADC_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 127, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5640_ADC_DATA, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 127, 0, adc_vol_tlv), + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("ADC Boost Gain", RT5640_ADC_BST_VOL, + RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + /* Class D speaker gain ratio */ + SOC_ENUM("Class D SPK Ratio Control", rt5640_clsd_spk_ratio_enum), + + SOC_ENUM("ADC IF1 Data Switch", rt5640_if1_adc_enum), + SOC_ENUM("DAC IF1 Data Switch", rt5640_if1_dac_enum), + SOC_ENUM("ADC IF2 Data Switch", rt5640_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5640->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5640_DMIC, RT5640_DMIC_CLK_MASK, + idx << RT5640_DMIC_CLK_SFT); + return idx; +} + +static int check_sysclk1_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5640_GLB_CLK); + val &= RT5640_SCLK_SRC_MASK; + if (val == RT5640_SCLK_SRC_PLL1 || val == RT5640_SCLK_SRC_PLL1T) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5640_sto_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_sto_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5640_AD_DA_MIXER, + RT5640_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5640_AD_DA_MIXER, + RT5640_M_IF1_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5640_AD_DA_MIXER, + RT5640_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5640_AD_DA_MIXER, + RT5640_M_IF1_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("ANC Switch", RT5640_STO_DAC_MIXER, + RT5640_M_ANC_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("ANC Switch", RT5640_STO_DAC_MIXER, + RT5640_M_ANC_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dig_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_DIG_MIXER, + RT5640_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_DIG_MIXER, + RT5640_M_DAC_L2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dig_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_DIG_MIXER, + RT5640_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_DIG_MIXER, + RT5640_M_DAC_R2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { + SOC_DAPM_SINGLE("HPOL Switch", RT5640_REC_L2_MIXER, + RT5640_M_HP_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, + RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST4_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5640_REC_L2_MIXER, + RT5640_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { + SOC_DAPM_SINGLE("HPOR Switch", RT5640_REC_R2_MIXER, + RT5640_M_HP_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, + RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST4_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5640_REC_R2_MIXER, + RT5640_M_OM_R_RM_R_SFT, 1, 1), +}; + +/* Analog Output Mixer */ +static const struct snd_kcontrol_new rt5640_spk_l_mix[] = { + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_SPK_L_MIXER, + RT5640_M_RM_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_SPK_L_MIXER, + RT5640_M_IN_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_SPK_L_MIXER, + RT5640_M_DAC_L1_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_SPK_L_MIXER, + RT5640_M_DAC_L2_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5640_SPK_L_MIXER, + RT5640_M_OM_L_SM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_spk_r_mix[] = { + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_SPK_R_MIXER, + RT5640_M_RM_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_SPK_R_MIXER, + RT5640_M_IN_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPK_R_MIXER, + RT5640_M_DAC_R1_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_SPK_R_MIXER, + RT5640_M_DAC_R2_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5640_SPK_R_MIXER, + RT5640_M_OM_R_SM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_out_l_mix[] = { + SOC_DAPM_SINGLE("SPK MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_SM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_R2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_out_r_mix[] = { + SOC_DAPM_SINGLE("SPK MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_SM_L_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_L2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, + RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_SPO_L_MIXER, + RT5640_M_DAC_L1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5640_SPO_L_MIXER, + RT5640_M_SV_R_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL L Switch", RT5640_SPO_L_MIXER, + RT5640_M_SV_L_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_SPO_L_MIXER, + RT5640_M_BST1_SPM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_spo_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_R_MIXER, + RT5640_M_DAC_R1_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5640_SPO_R_MIXER, + RT5640_M_SV_R_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_SPO_R_MIXER, + RT5640_M_BST1_SPM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC2 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC2_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, + RT5640_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_LOUT_MIXER, + RT5640_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5640_LOUT_MIXER, + RT5640_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5640_LOUT_MIXER, + RT5640_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_mix[] = { + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_MONO_MIXER, + RT5640_M_DAC_R2_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_MONO_MIXER, + RT5640_M_DAC_L2_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5640_MONO_MIXER, + RT5640_M_OV_R_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5640_MONO_MIXER, + RT5640_M_OV_L_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_MONO_MIXER, + RT5640_M_BST1_MM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +/* Stereo ADC source */ +static const char * const rt5640_stereo_adc1_src[] = { + "DIG MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5640_stereo_adc1_enum); + +static const char * const rt5640_stereo_adc2_src[] = { + "DMIC1", "DMIC2", "DIG MIX" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5640_stereo_adc2_enum); + +/* Mono ADC source */ +static const char * const rt5640_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADCL" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5640_mono_adc_l1_enum); + +static const char * const rt5640_mono_adc_l2_src[] = { + "DMIC L1", "DMIC L2", "Mono DAC MIXL" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5640_mono_adc_l2_enum); + +static const char * const rt5640_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADCR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5640_mono_adc_r1_enum); + +static const char * const rt5640_mono_adc_r2_src[] = { + "DMIC R1", "DMIC R2", "Mono DAC MIXR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5640_mono_adc_r2_enum); + +/* DAC2 channel source */ +static const char * const rt5640_dac_l2_src[] = { + "IF2", "Base L/R" +}; + +static int rt5640_dac_l2_values[] = { + 0, + 3, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt5640_dac_l2_enum, RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT, + 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values); + +static const struct snd_kcontrol_new rt5640_dac_l2_mux = + SOC_DAPM_VALUE_ENUM("DAC2 left channel source", rt5640_dac_l2_enum); + +static const char * const rt5640_dac_r2_src[] = { + "IF2", +}; + +static int rt5640_dac_r2_values[] = { + 0, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt5640_dac_r2_enum, RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT, + 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values); + +static const struct snd_kcontrol_new rt5640_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 right channel source", rt5640_dac_r2_enum); + +/* digital interface and iis interface map */ +static const char * const rt5640_dai_iis_map[] = { + "1:1|2:2", "1:2|2:1", "1:1|2:1", "1:2|2:2" +}; + +static int rt5640_dai_iis_map_values[] = { + 0, + 5, + 6, + 7, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt5640_dai_iis_map_enum, RT5640_I2S1_SDP, RT5640_I2S_IF_SFT, + 0x7, rt5640_dai_iis_map, rt5640_dai_iis_map_values); + +static const struct snd_kcontrol_new rt5640_dai_mux = + SOC_DAPM_VALUE_ENUM("DAI select", rt5640_dai_iis_map_enum); + +/* SDI select */ +static const char * const rt5640_sdi_sel[] = { + "IF1", "IF2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_sdi_sel_enum, RT5640_I2S2_SDP, + RT5640_I2S2_SDI_SFT, rt5640_sdi_sel); + +static const struct snd_kcontrol_new rt5640_sdi_mux = + SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); + +static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK | RT5640_GP3_PIN_MASK, + RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP3_PIN_DMIC1_SDA); + snd_soc_update_bits(codec, RT5640_DMIC, + RT5640_DMIC_1L_LH_MASK | RT5640_DMIC_1R_LH_MASK | + RT5640_DMIC_1_DP_MASK, + RT5640_DMIC_1L_LH_FALLING | RT5640_DMIC_1R_LH_RISING | + RT5640_DMIC_1_DP_IN1P); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK | RT5640_GP4_PIN_MASK, + RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP4_PIN_DMIC2_SDA); + snd_soc_update_bits(codec, RT5640_DMIC, + RT5640_DMIC_2L_LH_MASK | RT5640_DMIC_2R_LH_MASK | + RT5640_DMIC_2_DP_MASK, + RT5640_DMIC_2L_LH_FALLING | RT5640_DMIC_2R_LH_RISING | + RT5640_DMIC_2_DP_IN1N); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + rt5640->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, + RT5640_PWR_PLL_BIT, 0, NULL, 0), + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, + RT5640_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5640_PWR_ANLG2, + RT5640_PWR_MB1_BIT, 0, NULL, 0), + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5640_DMIC, + RT5640_DMIC_1_EN_SFT, 0, rt5640_set_dmic1_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5640_DMIC, + RT5640_DMIC_2_EN_SFT, 0, rt5640_set_dmic2_event, + SND_SOC_DAPM_PRE_PMU), + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5640_PWR_ANLG2, + RT5640_PWR_BST1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, + RT5640_PWR_BST4_BIT, 0, NULL, 0), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, + RT5640_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL, + RT5640_PWR_IN_R_BIT, 0, NULL, 0), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0, + rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5640_PWR_MIXER, RT5640_PWR_RM_R_BIT, 0, + rt5640_rec_r_mix, ARRAY_SIZE(rt5640_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, RT5640_PWR_DIG1, + RT5640_PWR_ADC_L_BIT, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, RT5640_PWR_DIG1, + RT5640_PWR_ADC_R_BIT, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("Stereo Filter", RT5640_PWR_DIG2, + RT5640_PWR_ADC_SF_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_adc_l_mix, ARRAY_SIZE(rt5640_sto_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_adc_r_mix, ARRAY_SIZE(rt5640_sto_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("Mono Left Filter", RT5640_PWR_DIG2, + RT5640_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_mono_adc_l_mix, ARRAY_SIZE(rt5640_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("Mono Right Filter", RT5640_PWR_DIG2, + RT5640_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_mono_adc_r_mix, ARRAY_SIZE(rt5640_mono_adc_r_mix)), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5640_PWR_DIG1, + RT5640_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5640_PWR_DIG1, + RT5640_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("DAI1 RX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("SDI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_sdi_mux), + SND_SOC_DAPM_MUX("DAI2 RX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("SDI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_sdi_mux), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_dac_r2_mux), + /* DAC Mixer */ + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_mono_dac_r_mix, ARRAY_SIZE(rt5640_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DIG MIXL", SND_SOC_NOPM, 0, 0, + rt5640_dig_l_mix, ARRAY_SIZE(rt5640_dig_l_mix)), + SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, + rt5640_dig_r_mix, ARRAY_SIZE(rt5640_dig_r_mix)), + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0), + /* SPK/OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, + 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), + SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, + 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, + RT5640_PWR_SV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKVOL R", RT5640_PWR_VOL, + RT5640_PWR_SV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL L", RT5640_PWR_VOL, + RT5640_PWR_OV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL R", RT5640_PWR_VOL, + RT5640_PWR_OV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL L", RT5640_PWR_VOL, + RT5640_PWR_HV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL R", RT5640_PWR_VOL, + RT5640_PWR_HV_R_BIT, 0, NULL, 0), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPOL MIX", SND_SOC_NOPM, 0, + 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), + SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, + 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, + rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, + RT5640_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, + RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), + SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("MONOP"), + SND_SOC_DAPM_OUTPUT("MONON"), +}; + +static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { + {"IN1P", NULL, "LDO2"}, + {"IN2P", NULL, "LDO2"}, + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + {"DMIC L2", NULL, "DMIC2"}, + {"DMIC R2", NULL, "DMIC2"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + + {"INL VOL", NULL, "IN2P"}, + {"INR VOL", NULL, "IN2N"}, + + {"RECMIXL", "HPOL Switch", "HPOL"}, + {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST2 Switch", "BST2"}, + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, + + {"RECMIXR", "HPOR Switch", "HPOR"}, + {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + {"RECMIXR", "BST1 Switch", "BST1"}, + {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC L2", NULL, "DMIC CLK"}, + {"DMIC L2", NULL, "DMIC2 Power"}, + {"DMIC R2", NULL, "DMIC CLK"}, + {"DMIC R2", NULL, "DMIC2 Power"}, + + {"Stereo ADC L2 Mux", "DMIC1", "DMIC L1"}, + {"Stereo ADC L2 Mux", "DMIC2", "DMIC L2"}, + {"Stereo ADC L2 Mux", "DIG MIX", "DIG MIXL"}, + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DIG MIX", "DIG MIXL"}, + + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DIG MIX", "DIG MIXR"}, + {"Stereo ADC R2 Mux", "DMIC1", "DMIC R1"}, + {"Stereo ADC R2 Mux", "DMIC2", "DMIC R2"}, + {"Stereo ADC R2 Mux", "DIG MIX", "DIG MIXR"}, + + {"Mono ADC L2 Mux", "DMIC L1", "DMIC L1"}, + {"Mono ADC L2 Mux", "DMIC L2", "DMIC L2"}, + {"Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "ADCL", "ADC L"}, + + {"Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + {"Mono ADC R1 Mux", "ADCR", "ADC R"}, + {"Mono ADC R2 Mux", "DMIC R1", "DMIC R1"}, + {"Mono ADC R2 Mux", "DMIC R2", "DMIC R2"}, + {"Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + + {"Stereo ADC MIXL", "ADC1 Switch", "Stereo ADC L1 Mux"}, + {"Stereo ADC MIXL", "ADC2 Switch", "Stereo ADC L2 Mux"}, + {"Stereo ADC MIXL", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", check_sysclk1_source}, + + {"Stereo ADC MIXR", "ADC1 Switch", "Stereo ADC R1 Mux"}, + {"Stereo ADC MIXR", "ADC2 Switch", "Stereo ADC R2 Mux"}, + {"Stereo ADC MIXR", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", check_sysclk1_source}, + + {"Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux"}, + {"Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux"}, + {"Mono ADC MIXL", NULL, "Mono Left Filter"}, + {"Mono Left Filter", NULL, "PLL1", check_sysclk1_source}, + + {"Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux"}, + {"Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux"}, + {"Mono ADC MIXR", NULL, "Mono Right Filter"}, + {"Mono Right Filter", NULL, "PLL1", check_sysclk1_source}, + + {"IF2 ADC L", NULL, "Mono ADC MIXL"}, + {"IF2 ADC R", NULL, "Mono ADC MIXR"}, + {"IF1 ADC L", NULL, "Stereo ADC MIXL"}, + {"IF1 ADC R", NULL, "Stereo ADC MIXR"}, + + {"IF1 ADC", NULL, "I2S1"}, + {"IF1 ADC", NULL, "IF1 ADC L"}, + {"IF1 ADC", NULL, "IF1 ADC R"}, + {"IF2 ADC", NULL, "I2S2"}, + {"IF2 ADC", NULL, "IF2 ADC L"}, + {"IF2 ADC", NULL, "IF2 ADC R"}, + + {"DAI1 TX Mux", "1:1|2:2", "IF1 ADC"}, + {"DAI1 TX Mux", "1:2|2:1", "IF2 ADC"}, + {"DAI1 IF1 Mux", "1:1|2:1", "IF1 ADC"}, + {"DAI1 IF2 Mux", "1:1|2:1", "IF2 ADC"}, + {"SDI1 TX Mux", "IF1", "DAI1 IF1 Mux"}, + {"SDI1 TX Mux", "IF2", "DAI1 IF2 Mux"}, + + {"DAI2 TX Mux", "1:2|2:1", "IF1 ADC"}, + {"DAI2 TX Mux", "1:1|2:2", "IF2 ADC"}, + {"DAI2 IF1 Mux", "1:2|2:2", "IF1 ADC"}, + {"DAI2 IF2 Mux", "1:2|2:2", "IF2 ADC"}, + {"SDI2 TX Mux", "IF1", "DAI2 IF1 Mux"}, + {"SDI2 TX Mux", "IF2", "DAI2 IF2 Mux"}, + + {"AIF1TX", NULL, "DAI1 TX Mux"}, + {"AIF1TX", NULL, "SDI1 TX Mux"}, + {"AIF2TX", NULL, "DAI2 TX Mux"}, + {"AIF2TX", NULL, "SDI2 TX Mux"}, + + {"DAI1 RX Mux", "1:1|2:2", "AIF1RX"}, + {"DAI1 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI1 RX Mux", "1:2|2:1", "AIF2RX"}, + {"DAI1 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"DAI2 RX Mux", "1:2|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:2", "AIF2RX"}, + {"DAI2 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"IF1 DAC", NULL, "I2S1"}, + {"IF1 DAC", NULL, "DAI1 RX Mux"}, + {"IF2 DAC", NULL, "I2S2"}, + {"IF2 DAC", NULL, "DAI2 RX Mux"}, + + {"IF1 DAC L", NULL, "IF1 DAC"}, + {"IF1 DAC R", NULL, "IF1 DAC"}, + {"IF2 DAC L", NULL, "IF2 DAC"}, + {"IF2 DAC R", NULL, "IF2 DAC"}, + + {"DAC MIXL", "Stereo ADC Switch", "Stereo ADC MIXL"}, + {"DAC MIXL", "INF1 Switch", "IF1 DAC L"}, + {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, + {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, + + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + + {"DAC L1", NULL, "Stereo DAC MIXL"}, + {"DAC L1", NULL, "PLL1", check_sysclk1_source}, + {"DAC R1", NULL, "Stereo DAC MIXR"}, + {"DAC R1", NULL, "PLL1", check_sysclk1_source}, + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", check_sysclk1_source}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", check_sysclk1_source}, + + {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, + {"SPK MIXL", "INL Switch", "INL VOL"}, + {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, + {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, + {"SPK MIXR", "INR Switch", "INR VOL"}, + {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXL", "BST1 Switch", "BST1"}, + {"OUT MIXL", "INL Switch", "INL VOL"}, + {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, + + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + {"OUT MIXR", "BST2 Switch", "BST2"}, + {"OUT MIXR", "BST1 Switch", "BST1"}, + {"OUT MIXR", "INR Switch", "INR VOL"}, + {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, + + {"SPKVOL L", NULL, "SPK MIXL"}, + {"SPKVOL R", NULL, "SPK MIXR"}, + {"HPOVOL L", NULL, "OUT MIXL"}, + {"HPOVOL R", NULL, "OUT MIXR"}, + {"OUTVOL L", NULL, "OUT MIXL"}, + {"OUTVOL R", NULL, "OUT MIXR"}, + + {"SPOL MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOL MIX", "DAC L1 Switch", "DAC L1"}, + {"SPOL MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOL MIX", "SPKVOL L Switch", "SPKVOL L"}, + {"SPOL MIX", "BST1 Switch", "BST1"}, + {"SPOR MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOR MIX", "BST1 Switch", "BST1"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, + {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, + {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, + + {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, + {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, + {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, + + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, + + {"SPOLP", NULL, "Improve SPK Amp Drv"}, + {"SPOLN", NULL, "Improve SPK Amp Drv"}, + {"SPORP", NULL, "Improve SPK Amp Drv"}, + {"SPORN", NULL, "Improve SPK Amp Drv"}, + + {"HPOL", NULL, "Improve HP Amp Drv"}, + {"HPOR", NULL, "Improve HP Amp Drv"}, + + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, + {"LOUTL", NULL, "LOUT MIX"}, + {"LOUTR", NULL, "LOUT MIX"}, + {"MONOP", NULL, "Mono MIX"}, + {"MONON", NULL, "Mono MIX"}, + {"MONOP", NULL, "Improve MONO Amp Drv"}, +}; + +static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) +{ + int ret = 0, val; + + if (codec == NULL) + return -EINVAL; + + val = snd_soc_read(codec, RT5640_I2S1_SDP); + val = (val & RT5640_I2S_IF_MASK) >> RT5640_I2S_IF_SFT; + switch (dai_id) { + case RT5640_AIF1: + switch (val) { + case RT5640_IF_123: + case RT5640_IF_132: + ret |= RT5640_U_IF1; + break; + case RT5640_IF_113: + ret |= RT5640_U_IF1; + case RT5640_IF_312: + case RT5640_IF_213: + ret |= RT5640_U_IF2; + break; + } + break; + + case RT5640_AIF2: + switch (val) { + case RT5640_IF_231: + case RT5640_IF_213: + ret |= RT5640_U_IF1; + break; + case RT5640_IF_223: + ret |= RT5640_U_IF1; + case RT5640_IF_123: + case RT5640_IF_321: + ret |= RT5640_U_IF2; + break; + } + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5640_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk, dai_sel; + int pre_div, bclk_ms, frame_size; + + rt5640->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + rt5640->bclk[dai->id] = rt5640->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5640->bclk[dai->id], rt5640->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5640_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5640_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5640_I2S_DL_8; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5640_U_IF1) { + mask_clk = RT5640_I2S_BCLK_MS1_MASK | RT5640_I2S_PD1_MASK; + val_clk = bclk_ms << RT5640_I2S_BCLK_MS1_SFT | + pre_div << RT5640_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5640_I2S1_SDP, + RT5640_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5640_ADDA_CLK1, mask_clk, val_clk); + } + if (dai_sel & RT5640_U_IF2) { + mask_clk = RT5640_I2S_BCLK_MS2_MASK | RT5640_I2S_PD2_MASK; + val_clk = bclk_ms << RT5640_I2S_BCLK_MS2_SFT | + pre_div << RT5640_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5640_I2S2_SDP, + RT5640_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5640_ADDA_CLK1, mask_clk, val_clk); + } + + return 0; +} + +static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0, dai_sel; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5640->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5640_I2S_MS_S; + rt5640->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5640_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5640_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5640_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5640_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5640_U_IF1) { + snd_soc_update_bits(codec, RT5640_I2S1_SDP, + RT5640_I2S_MS_MASK | RT5640_I2S_BP_MASK | + RT5640_I2S_DF_MASK, reg_val); + } + if (dai_sel & RT5640_U_IF2) { + snd_soc_update_bits(codec, RT5640_I2S2_SDP, + RT5640_I2S_MS_MASK | RT5640_I2S_BP_MASK | + RT5640_I2S_DF_MASK, reg_val); + } + + return 0; +} + +static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5640->sysclk && clk_id == rt5640->sysclk_src) + return 0; + + switch (clk_id) { + case RT5640_SCLK_S_MCLK: + reg_val |= RT5640_SCLK_SRC_MCLK; + break; + case RT5640_SCLK_S_PLL1: + reg_val |= RT5640_SCLK_SRC_PLL1; + break; + case RT5640_SCLK_S_PLL1_TK: + reg_val |= RT5640_SCLK_SRC_PLL1T; + break; + case RT5640_SCLK_S_RCCLK: + reg_val |= RT5640_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, reg_val); + rt5640->sysclk = freq; + rt5640->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + return 0; +} + +/** + * rt5640_pll_calc - Calculate PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calculate M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5640_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5640_pll_code *pll_code) +{ + int max_n = RT5640_PLL_N_MAX, max_m = RT5640_PLL_M_MAX; + int n = 0, m = 0, red, n_t, m_t, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5640_PLL_INP_MAX < freq_in || RT5640_PLL_INP_MIN > freq_in) + return -EINVAL; + + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = (freq_in >> 1) + (freq_in >> 2) * n_t; + if (in_t < 0) + continue; + if (in_t == freq_out) { + bypass = true; + n = n_t; + goto code_find; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - freq_out); + if (red < red_t) { + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = 2; + return 0; +} + +static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + struct rt5640_pll_code *pll_code = &rt5640->pll_code; + int ret, dai_sel; + + if (source == rt5640->pll_src && freq_in == rt5640->pll_in && + freq_out == rt5640->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5640->pll_in = 0; + rt5640->pll_out = 0; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, RT5640_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5640_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_MCLK); + break; + case RT5640_PLL1_S_BCLK1: + case RT5640_PLL1_S_BCLK2: + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, + "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5640_U_IF1) { + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); + } + if (dai_sel & RT5640_U_IF2) { + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5640_pll_calc(freq_in, freq_out, pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=2\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code); + + snd_soc_write(codec, RT5640_PLL_CTRL1, + pll_code->n_code << RT5640_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5640_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5640_PLL_M_SFT | + pll_code->m_bp << RT5640_PLL_M_BP_SFT); + + rt5640->pll_in = freq_in; + rt5640->pll_out = freq_out; + rt5640->pll_src = source; + + return 0; +} + +static int rt5640_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + regcache_cache_only(rt5640->regmap, false); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_VREF1 | RT5640_PWR_MB | + RT5640_PWR_BG | RT5640_PWR_VREF2, + RT5640_PWR_VREF1 | RT5640_PWR_MB | + RT5640_PWR_BG | RT5640_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + regcache_sync(rt5640->regmap); + snd_soc_update_bits(codec, RT5640_DUMMY1, + 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5640_MICBIAS, + 0x0030, 0x0030); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5640_DEPOP_M1, 0x0004); + snd_soc_write(codec, RT5640_DEPOP_M2, 0x1100); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0); + snd_soc_write(codec, RT5640_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5640_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5640_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5640_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5640_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5640_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5640_probe(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5640->codec = codec; + codec->control_data = rt5640->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + codec->dapm.idle_bias_off = 1; + rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); + snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + + return 0; +} + +static int rt5640_remove(struct snd_soc_codec *codec) +{ + rt5640_reset(codec); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5640_suspend(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); + rt5640_reset(codec); + regcache_cache_only(rt5640->regmap, true); + regcache_mark_dirty(rt5640->regmap); + + return 0; +} + +static int rt5640_resume(struct snd_soc_codec *codec) +{ + rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define rt5640_suspend NULL +#define rt5640_resume NULL +#endif + +#define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5640_aif_dai_ops = { + .hw_params = rt5640_hw_params, + .set_fmt = rt5640_set_dai_fmt, + .set_sysclk = rt5640_set_dai_sysclk, + .set_pll = rt5640_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5640_dai[] = { + { + .name = "rt5640-aif1", + .id = RT5640_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, + { + .name = "rt5640-aif2", + .id = RT5640_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { + .probe = rt5640_probe, + .remove = rt5640_remove, + .suspend = rt5640_suspend, + .resume = rt5640_resume, + .set_bias_level = rt5640_set_bias_level, + .controls = rt5640_snd_controls, + .num_controls = ARRAY_SIZE(rt5640_snd_controls), + .dapm_widgets = rt5640_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5640_dapm_widgets), + .dapm_routes = rt5640_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5640_dapm_routes), +}; + +static const struct regmap_config rt5640_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * + RT5640_PR_SPACING), + .volatile_reg = rt5640_volatile_register, + .readable_reg = rt5640_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5640_reg, + .num_reg_defaults = ARRAY_SIZE(rt5640_reg), + .ranges = rt5640_ranges, + .num_ranges = ARRAY_SIZE(rt5640_ranges), +}; + +static const struct i2c_device_id rt5640_i2c_id[] = { + { "rt5640", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); + +static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) +{ + rt5640->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5640->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + + rt5640->pdata.ldo1_en = of_get_named_gpio(np, + "realtek,ldo1-en-gpios", 0); + /* + * LDO1_EN is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5640->pdata.ldo1_en) && + (rt5640->pdata.ldo1_en != -ENOENT)) + return rt5640->pdata.ldo1_en; + + return 0; +} + +static int rt5640_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5640_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5640_priv *rt5640; + int ret; + unsigned int val; + + rt5640 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5640_priv), + GFP_KERNEL); + if (NULL == rt5640) + return -ENOMEM; + i2c_set_clientdata(i2c, rt5640); + + if (pdata) { + rt5640->pdata = *pdata; + /* + * Translate zero'd out (default) pdata value to an invalid + * GPIO ID. This makes the pdata and DT paths consistent in + * terms of the value left in this field when no GPIO is + * specified, but means we can't actually use GPIO 0. + */ + if (!rt5640->pdata.ldo1_en) + rt5640->pdata.ldo1_en = -EINVAL; + } else if (i2c->dev.of_node) { + ret = rt5640_parse_dt(rt5640, i2c->dev.of_node); + if (ret) + return ret; + } else + rt5640->pdata.ldo1_en = -EINVAL; + + rt5640->regmap = devm_regmap_init_i2c(i2c, &rt5640_regmap); + if (IS_ERR(rt5640->regmap)) { + ret = PTR_ERR(rt5640->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { + ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, + "RT5640 LDO1_EN"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDO1_EN %d: %d\n", + rt5640->pdata.ldo1_en, ret); + return ret; + } + msleep(400); + } + + regmap_read(rt5640->regmap, RT5640_VENDOR_ID2, &val); + if ((val != RT5640_DEVICE_ID)) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5640/39\n", val); + return -ENODEV; + } + + regmap_write(rt5640->regmap, RT5640_RESET, 0); + + ret = regmap_register_patch(rt5640->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5640->pdata.in1_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF1, RT5640_IN_DF1); + + if (rt5640->pdata.in2_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, + RT5640_IN_DF2, RT5640_IN_DF2); + + rt5640->hp_mute = 1; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5640_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5640_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + }, + .probe = rt5640_i2c_probe, + .remove = rt5640_i2c_remove, + .id_table = rt5640_i2c_id, +}; +module_i2c_driver(rt5640_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5640 driver"); +MODULE_AUTHOR("Johnny Hsu <johnnyhsu@realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h new file mode 100644 index 0000000..5e8df25a --- /dev/null +++ b/sound/soc/codecs/rt5640.h @@ -0,0 +1,2104 @@ +/* + * rt5640.h -- RT5640 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu@realtek.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. + */ + +#ifndef _RT5640_H +#define _RT5640_H + +#include <sound/rt5640.h> + +/* Info */ +#define RT5640_RESET 0x00 +#define RT5640_VENDOR_ID 0xfd +#define RT5640_VENDOR_ID1 0xfe +#define RT5640_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5640_SPK_VOL 0x01 +#define RT5640_HP_VOL 0x02 +#define RT5640_OUTPUT 0x03 +#define RT5640_MONO_OUT 0x04 +/* I/O - Input */ +#define RT5640_IN1_IN2 0x0d +#define RT5640_IN3_IN4 0x0e +#define RT5640_INL_INR_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5640_DAC1_DIG_VOL 0x19 +#define RT5640_DAC2_DIG_VOL 0x1a +#define RT5640_DAC2_CTRL 0x1b +#define RT5640_ADC_DIG_VOL 0x1c +#define RT5640_ADC_DATA 0x1d +#define RT5640_ADC_BST_VOL 0x1e +/* Mixer - D-D */ +#define RT5640_STO_ADC_MIXER 0x27 +#define RT5640_MONO_ADC_MIXER 0x28 +#define RT5640_AD_DA_MIXER 0x29 +#define RT5640_STO_DAC_MIXER 0x2a +#define RT5640_MONO_DAC_MIXER 0x2b +#define RT5640_DIG_MIXER 0x2c +#define RT5640_DSP_PATH1 0x2d +#define RT5640_DSP_PATH2 0x2e +#define RT5640_DIG_INF_DATA 0x2f +/* Mixer - ADC */ +#define RT5640_REC_L1_MIXER 0x3b +#define RT5640_REC_L2_MIXER 0x3c +#define RT5640_REC_R1_MIXER 0x3d +#define RT5640_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5640_HPO_MIXER 0x45 +#define RT5640_SPK_L_MIXER 0x46 +#define RT5640_SPK_R_MIXER 0x47 +#define RT5640_SPO_L_MIXER 0x48 +#define RT5640_SPO_R_MIXER 0x49 +#define RT5640_SPO_CLSD_RATIO 0x4a +#define RT5640_MONO_MIXER 0x4c +#define RT5640_OUT_L1_MIXER 0x4d +#define RT5640_OUT_L2_MIXER 0x4e +#define RT5640_OUT_L3_MIXER 0x4f +#define RT5640_OUT_R1_MIXER 0x50 +#define RT5640_OUT_R2_MIXER 0x51 +#define RT5640_OUT_R3_MIXER 0x52 +#define RT5640_LOUT_MIXER 0x53 +/* Power */ +#define RT5640_PWR_DIG1 0x61 +#define RT5640_PWR_DIG2 0x62 +#define RT5640_PWR_ANLG1 0x63 +#define RT5640_PWR_ANLG2 0x64 +#define RT5640_PWR_MIXER 0x65 +#define RT5640_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5640_PRIV_INDEX 0x6a +#define RT5640_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5640_I2S1_SDP 0x70 +#define RT5640_I2S2_SDP 0x71 +#define RT5640_ADDA_CLK1 0x73 +#define RT5640_ADDA_CLK2 0x74 +#define RT5640_DMIC 0x75 +/* Function - Analog */ +#define RT5640_GLB_CLK 0x80 +#define RT5640_PLL_CTRL1 0x81 +#define RT5640_PLL_CTRL2 0x82 +#define RT5640_ASRC_1 0x83 +#define RT5640_ASRC_2 0x84 +#define RT5640_ASRC_3 0x85 +#define RT5640_ASRC_4 0x89 +#define RT5640_ASRC_5 0x8a +#define RT5640_HP_OVCD 0x8b +#define RT5640_CLS_D_OVCD 0x8c +#define RT5640_CLS_D_OUT 0x8d +#define RT5640_DEPOP_M1 0x8e +#define RT5640_DEPOP_M2 0x8f +#define RT5640_DEPOP_M3 0x90 +#define RT5640_CHARGE_PUMP 0x91 +#define RT5640_PV_DET_SPK_G 0x92 +#define RT5640_MICBIAS 0x93 +/* Function - Digital */ +#define RT5640_EQ_CTRL1 0xb0 +#define RT5640_EQ_CTRL2 0xb1 +#define RT5640_WIND_FILTER 0xb2 +#define RT5640_DRC_AGC_1 0xb4 +#define RT5640_DRC_AGC_2 0xb5 +#define RT5640_DRC_AGC_3 0xb6 +#define RT5640_SVOL_ZC 0xb7 +#define RT5640_ANC_CTRL1 0xb8 +#define RT5640_ANC_CTRL2 0xb9 +#define RT5640_ANC_CTRL3 0xba +#define RT5640_JD_CTRL 0xbb +#define RT5640_ANC_JD 0xbc +#define RT5640_IRQ_CTRL1 0xbd +#define RT5640_IRQ_CTRL2 0xbe +#define RT5640_INT_IRQ_ST 0xbf +#define RT5640_GPIO_CTRL1 0xc0 +#define RT5640_GPIO_CTRL2 0xc1 +#define RT5640_GPIO_CTRL3 0xc2 +#define RT5640_DSP_CTRL1 0xc4 +#define RT5640_DSP_CTRL2 0xc5 +#define RT5640_DSP_CTRL3 0xc6 +#define RT5640_DSP_CTRL4 0xc7 +#define RT5640_PGM_REG_ARR1 0xc8 +#define RT5640_PGM_REG_ARR2 0xc9 +#define RT5640_PGM_REG_ARR3 0xca +#define RT5640_PGM_REG_ARR4 0xcb +#define RT5640_PGM_REG_ARR5 0xcc +#define RT5640_SCB_FUNC 0xcd +#define RT5640_SCB_CTRL 0xce +#define RT5640_BASE_BACK 0xcf +#define RT5640_MP3_PLUS1 0xd0 +#define RT5640_MP3_PLUS2 0xd1 +#define RT5640_3D_HP 0xd2 +#define RT5640_ADJ_HPF 0xd3 +#define RT5640_HP_CALIB_AMP_DET 0xd6 +#define RT5640_HP_CALIB2 0xd7 +#define RT5640_SV_ZCD1 0xd9 +#define RT5640_SV_ZCD2 0xda +/* Dummy Register */ +#define RT5640_DUMMY1 0xfa +#define RT5640_DUMMY2 0xfb +#define RT5640_DUMMY3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 +#define RT5640_3D_SPK 0x63 +#define RT5640_WND_1 0x6c +#define RT5640_WND_2 0x6d +#define RT5640_WND_3 0x6e +#define RT5640_WND_4 0x6f +#define RT5640_WND_5 0x70 +#define RT5640_WND_8 0x73 +#define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 +#define RT5640_EQ_BW_LOP 0xa0 +#define RT5640_EQ_GN_LOP 0xa1 +#define RT5640_EQ_FC_BP1 0xa2 +#define RT5640_EQ_BW_BP1 0xa3 +#define RT5640_EQ_GN_BP1 0xa4 +#define RT5640_EQ_FC_BP2 0xa5 +#define RT5640_EQ_BW_BP2 0xa6 +#define RT5640_EQ_GN_BP2 0xa7 +#define RT5640_EQ_FC_BP3 0xa8 +#define RT5640_EQ_BW_BP3 0xa9 +#define RT5640_EQ_GN_BP3 0xaa +#define RT5640_EQ_FC_BP4 0xab +#define RT5640_EQ_BW_BP4 0xac +#define RT5640_EQ_GN_BP4 0xad +#define RT5640_EQ_FC_HIP1 0xae +#define RT5640_EQ_GN_HIP1 0xaf +#define RT5640_EQ_FC_HIP2 0xb0 +#define RT5640_EQ_BW_HIP2 0xb1 +#define RT5640_EQ_GN_HIP2 0xb2 +#define RT5640_EQ_PRE_VOL 0xb3 +#define RT5640_EQ_PST_VOL 0xb4 + +/* global definition */ +#define RT5640_L_MUTE (0x1 << 15) +#define RT5640_L_MUTE_SFT 15 +#define RT5640_VOL_L_MUTE (0x1 << 14) +#define RT5640_VOL_L_SFT 14 +#define RT5640_R_MUTE (0x1 << 7) +#define RT5640_R_MUTE_SFT 7 +#define RT5640_VOL_R_MUTE (0x1 << 6) +#define RT5640_VOL_R_SFT 6 +#define RT5640_L_VOL_MASK (0x3f << 8) +#define RT5640_L_VOL_SFT 8 +#define RT5640_R_VOL_MASK (0x3f) +#define RT5640_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +/* IN3 and IN4 Control (0x0e) */ +#define RT5640_BST_SFT1 12 +#define RT5640_BST_SFT2 8 +#define RT5640_IN_DF1 (0x1 << 7) +#define RT5640_IN_SFT1 7 +#define RT5640_IN_DF2 (0x1 << 6) +#define RT5640_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5640_INL_SEL_MASK (0x1 << 15) +#define RT5640_INL_SEL_SFT 15 +#define RT5640_INL_SEL_IN4P (0x0 << 15) +#define RT5640_INL_SEL_MONOP (0x1 << 15) +#define RT5640_INL_VOL_MASK (0x1f << 8) +#define RT5640_INL_VOL_SFT 8 +#define RT5640_INR_SEL_MASK (0x1 << 7) +#define RT5640_INR_SEL_SFT 7 +#define RT5640_INR_SEL_IN4N (0x0 << 7) +#define RT5640_INR_SEL_MONON (0x1 << 7) +#define RT5640_INR_VOL_MASK (0x1f) +#define RT5640_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5640_DAC_L1_VOL_MASK (0xff << 8) +#define RT5640_DAC_L1_VOL_SFT 8 +#define RT5640_DAC_R1_VOL_MASK (0xff) +#define RT5640_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5640_DAC_L2_VOL_MASK (0xff << 8) +#define RT5640_DAC_L2_VOL_SFT 8 +#define RT5640_DAC_R2_VOL_MASK (0xff) +#define RT5640_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5640_M_DAC_L2_VOL (0x1 << 13) +#define RT5640_M_DAC_L2_VOL_SFT 13 +#define RT5640_M_DAC_R2_VOL (0x1 << 12) +#define RT5640_M_DAC_R2_VOL_SFT 12 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5640_ADC_L_VOL_MASK (0x7f << 8) +#define RT5640_ADC_L_VOL_SFT 8 +#define RT5640_ADC_R_VOL_MASK (0x7f) +#define RT5640_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5640_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5640_MONO_ADC_L_VOL_SFT 8 +#define RT5640_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5640_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5640_ADC_L_BST_MASK (0x3 << 14) +#define RT5640_ADC_L_BST_SFT 14 +#define RT5640_ADC_R_BST_MASK (0x3 << 12) +#define RT5640_ADC_R_BST_SFT 12 +#define RT5640_ADC_COMP_MASK (0x3 << 10) +#define RT5640_ADC_COMP_SFT 10 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5640_M_ADC_L1 (0x1 << 14) +#define RT5640_M_ADC_L1_SFT 14 +#define RT5640_M_ADC_L2 (0x1 << 13) +#define RT5640_M_ADC_L2_SFT 13 +#define RT5640_ADC_1_SRC_MASK (0x1 << 12) +#define RT5640_ADC_1_SRC_SFT 12 +#define RT5640_ADC_1_SRC_ADC (0x1 << 12) +#define RT5640_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5640_ADC_2_SRC_MASK (0x3 << 10) +#define RT5640_ADC_2_SRC_SFT 10 +#define RT5640_ADC_2_SRC_DMIC1 (0x0 << 10) +#define RT5640_ADC_2_SRC_DMIC2 (0x1 << 10) +#define RT5640_ADC_2_SRC_DACMIX (0x2 << 10) +#define RT5640_M_ADC_R1 (0x1 << 6) +#define RT5640_M_ADC_R1_SFT 6 +#define RT5640_M_ADC_R2 (0x1 << 5) +#define RT5640_M_ADC_R2_SFT 5 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5640_M_MONO_ADC_L1 (0x1 << 14) +#define RT5640_M_MONO_ADC_L1_SFT 14 +#define RT5640_M_MONO_ADC_L2 (0x1 << 13) +#define RT5640_M_MONO_ADC_L2_SFT 13 +#define RT5640_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5640_MONO_ADC_L1_SRC_SFT 12 +#define RT5640_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5640_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5640_MONO_ADC_L2_SRC_MASK (0x3 << 10) +#define RT5640_MONO_ADC_L2_SRC_SFT 10 +#define RT5640_MONO_ADC_L2_SRC_DMIC_L1 (0x0 << 10) +#define RT5640_MONO_ADC_L2_SRC_DMIC_L2 (0x1 << 10) +#define RT5640_MONO_ADC_L2_SRC_DACMIXL (0x2 << 10) +#define RT5640_M_MONO_ADC_R1 (0x1 << 6) +#define RT5640_M_MONO_ADC_R1_SFT 6 +#define RT5640_M_MONO_ADC_R2 (0x1 << 5) +#define RT5640_M_MONO_ADC_R2_SFT 5 +#define RT5640_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5640_MONO_ADC_R1_SRC_SFT 4 +#define RT5640_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5640_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5640_MONO_ADC_R2_SRC_MASK (0x3 << 2) +#define RT5640_MONO_ADC_R2_SRC_SFT 2 +#define RT5640_MONO_ADC_R2_SRC_DMIC_R1 (0x0 << 2) +#define RT5640_MONO_ADC_R2_SRC_DMIC_R2 (0x1 << 2) +#define RT5640_MONO_ADC_R2_SRC_DACMIXR (0x2 << 2) + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5640_M_ADCMIX_L (0x1 << 15) +#define RT5640_M_ADCMIX_L_SFT 15 +#define RT5640_M_IF1_DAC_L (0x1 << 14) +#define RT5640_M_IF1_DAC_L_SFT 14 +#define RT5640_M_ADCMIX_R (0x1 << 7) +#define RT5640_M_ADCMIX_R_SFT 7 +#define RT5640_M_IF1_DAC_R (0x1 << 6) +#define RT5640_M_IF1_DAC_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5640_M_DAC_L1 (0x1 << 14) +#define RT5640_M_DAC_L1_SFT 14 +#define RT5640_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5640_DAC_L1_STO_L_VOL_SFT 13 +#define RT5640_M_DAC_L2 (0x1 << 12) +#define RT5640_M_DAC_L2_SFT 12 +#define RT5640_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5640_DAC_L2_STO_L_VOL_SFT 11 +#define RT5640_M_ANC_DAC_L (0x1 << 10) +#define RT5640_M_ANC_DAC_L_SFT 10 +#define RT5640_M_DAC_R1 (0x1 << 6) +#define RT5640_M_DAC_R1_SFT 6 +#define RT5640_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5640_DAC_R1_STO_R_VOL_SFT 5 +#define RT5640_M_DAC_R2 (0x1 << 4) +#define RT5640_M_DAC_R2_SFT 4 +#define RT5640_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5640_DAC_R2_STO_R_VOL_SFT 3 +#define RT5640_M_ANC_DAC_R (0x1 << 2) +#define RT5640_M_ANC_DAC_R_SFT 2 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5640_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5640_M_DAC_L1_MONO_L_SFT 14 +#define RT5640_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5640_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5640_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5640_M_DAC_L2_MONO_L_SFT 12 +#define RT5640_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5640_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5640_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5640_M_DAC_R2_MONO_L_SFT 10 +#define RT5640_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5640_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5640_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5640_M_DAC_R1_MONO_R_SFT 6 +#define RT5640_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5640_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5640_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5640_M_DAC_R2_MONO_R_SFT 4 +#define RT5640_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5640_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5640_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5640_M_DAC_L2_MONO_R_SFT 2 +#define RT5640_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5640_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5640_M_STO_L_DAC_L (0x1 << 15) +#define RT5640_M_STO_L_DAC_L_SFT 15 +#define RT5640_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5640_STO_L_DAC_L_VOL_SFT 14 +#define RT5640_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5640_M_DAC_L2_DAC_L_SFT 13 +#define RT5640_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5640_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5640_M_STO_R_DAC_R (0x1 << 11) +#define RT5640_M_STO_R_DAC_R_SFT 11 +#define RT5640_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5640_STO_R_DAC_R_VOL_SFT 10 +#define RT5640_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5640_M_DAC_R2_DAC_R_SFT 9 +#define RT5640_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5640_DAC_R2_DAC_R_VOL_SFT 8 + +/* DSP Path Control 1 (0x2d) */ +#define RT5640_RXDP_SRC_MASK (0x1 << 15) +#define RT5640_RXDP_SRC_SFT 15 +#define RT5640_RXDP_SRC_NOR (0x0 << 15) +#define RT5640_RXDP_SRC_DIV3 (0x1 << 15) +#define RT5640_TXDP_SRC_MASK (0x1 << 14) +#define RT5640_TXDP_SRC_SFT 14 +#define RT5640_TXDP_SRC_NOR (0x0 << 14) +#define RT5640_TXDP_SRC_DIV3 (0x1 << 14) + +/* DSP Path Control 2 (0x2e) */ +#define RT5640_DAC_L2_SEL_MASK (0x3 << 14) +#define RT5640_DAC_L2_SEL_SFT 14 +#define RT5640_DAC_L2_SEL_IF2 (0x0 << 14) +#define RT5640_DAC_L2_SEL_IF3 (0x1 << 14) +#define RT5640_DAC_L2_SEL_TXDC (0x2 << 14) +#define RT5640_DAC_L2_SEL_BASS (0x3 << 14) +#define RT5640_DAC_R2_SEL_MASK (0x3 << 12) +#define RT5640_DAC_R2_SEL_SFT 12 +#define RT5640_DAC_R2_SEL_IF2 (0x0 << 12) +#define RT5640_DAC_R2_SEL_IF3 (0x1 << 12) +#define RT5640_DAC_R2_SEL_TXDC (0x2 << 12) +#define RT5640_IF2_ADC_L_SEL_MASK (0x1 << 11) +#define RT5640_IF2_ADC_L_SEL_SFT 11 +#define RT5640_IF2_ADC_L_SEL_TXDP (0x0 << 11) +#define RT5640_IF2_ADC_L_SEL_PASS (0x1 << 11) +#define RT5640_IF2_ADC_R_SEL_MASK (0x1 << 10) +#define RT5640_IF2_ADC_R_SEL_SFT 10 +#define RT5640_IF2_ADC_R_SEL_TXDP (0x0 << 10) +#define RT5640_IF2_ADC_R_SEL_PASS (0x1 << 10) +#define RT5640_RXDC_SEL_MASK (0x3 << 8) +#define RT5640_RXDC_SEL_SFT 8 +#define RT5640_RXDC_SEL_NOR (0x0 << 8) +#define RT5640_RXDC_SEL_L2R (0x1 << 8) +#define RT5640_RXDC_SEL_R2L (0x2 << 8) +#define RT5640_RXDC_SEL_SWAP (0x3 << 8) +#define RT5640_RXDP_SEL_MASK (0x3 << 6) +#define RT5640_RXDP_SEL_SFT 6 +#define RT5640_RXDP_SEL_NOR (0x0 << 6) +#define RT5640_RXDP_SEL_L2R (0x1 << 6) +#define RT5640_RXDP_SEL_R2L (0x2 << 6) +#define RT5640_RXDP_SEL_SWAP (0x3 << 6) +#define RT5640_TXDC_SEL_MASK (0x3 << 4) +#define RT5640_TXDC_SEL_SFT 4 +#define RT5640_TXDC_SEL_NOR (0x0 << 4) +#define RT5640_TXDC_SEL_L2R (0x1 << 4) +#define RT5640_TXDC_SEL_R2L (0x2 << 4) +#define RT5640_TXDC_SEL_SWAP (0x3 << 4) +#define RT5640_TXDP_SEL_MASK (0x3 << 2) +#define RT5640_TXDP_SEL_SFT 2 +#define RT5640_TXDP_SEL_NOR (0x0 << 2) +#define RT5640_TXDP_SEL_L2R (0x1 << 2) +#define RT5640_TXDP_SEL_R2L (0x2 << 2) +#define RT5640_TRXDP_SEL_SWAP (0x3 << 2) + +/* Digital Interface Data Control (0x2f) */ +#define RT5640_IF1_DAC_SEL_MASK (0x3 << 14) +#define RT5640_IF1_DAC_SEL_SFT 14 +#define RT5640_IF1_DAC_SEL_NOR (0x0 << 14) +#define RT5640_IF1_DAC_SEL_L2R (0x1 << 14) +#define RT5640_IF1_DAC_SEL_R2L (0x2 << 14) +#define RT5640_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5640_IF1_ADC_SEL_MASK (0x3 << 12) +#define RT5640_IF1_ADC_SEL_SFT 12 +#define RT5640_IF1_ADC_SEL_NOR (0x0 << 12) +#define RT5640_IF1_ADC_SEL_L2R (0x1 << 12) +#define RT5640_IF1_ADC_SEL_R2L (0x2 << 12) +#define RT5640_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5640_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5640_IF2_DAC_SEL_SFT 10 +#define RT5640_IF2_DAC_SEL_NOR (0x0 << 10) +#define RT5640_IF2_DAC_SEL_L2R (0x1 << 10) +#define RT5640_IF2_DAC_SEL_R2L (0x2 << 10) +#define RT5640_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5640_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5640_IF2_ADC_SEL_SFT 8 +#define RT5640_IF2_ADC_SEL_NOR (0x0 << 8) +#define RT5640_IF2_ADC_SEL_L2R (0x1 << 8) +#define RT5640_IF2_ADC_SEL_R2L (0x2 << 8) +#define RT5640_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5640_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5640_IF3_DAC_SEL_SFT 6 +#define RT5640_IF3_DAC_SEL_NOR (0x0 << 6) +#define RT5640_IF3_DAC_SEL_L2R (0x1 << 6) +#define RT5640_IF3_DAC_SEL_R2L (0x2 << 6) +#define RT5640_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5640_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5640_IF3_ADC_SEL_SFT 4 +#define RT5640_IF3_ADC_SEL_NOR (0x0 << 4) +#define RT5640_IF3_ADC_SEL_L2R (0x1 << 4) +#define RT5640_IF3_ADC_SEL_R2L (0x2 << 4) +#define RT5640_IF3_ADC_SEL_SWAP (0x3 << 4) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5640_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5640_G_HP_L_RM_L_SFT 13 +#define RT5640_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5640_G_IN_L_RM_L_SFT 10 +#define RT5640_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5640_G_BST4_RM_L_SFT 7 +#define RT5640_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5640_G_BST3_RM_L_SFT 4 +#define RT5640_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5640_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5640_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5640_G_BST1_RM_L_SFT 13 +#define RT5640_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5640_G_OM_L_RM_L_SFT 10 +#define RT5640_M_HP_L_RM_L (0x1 << 6) +#define RT5640_M_HP_L_RM_L_SFT 6 +#define RT5640_M_IN_L_RM_L (0x1 << 5) +#define RT5640_M_IN_L_RM_L_SFT 5 +#define RT5640_M_BST4_RM_L (0x1 << 4) +#define RT5640_M_BST4_RM_L_SFT 4 +#define RT5640_M_BST3_RM_L (0x1 << 3) +#define RT5640_M_BST3_RM_L_SFT 3 +#define RT5640_M_BST2_RM_L (0x1 << 2) +#define RT5640_M_BST2_RM_L_SFT 2 +#define RT5640_M_BST1_RM_L (0x1 << 1) +#define RT5640_M_BST1_RM_L_SFT 1 +#define RT5640_M_OM_L_RM_L (0x1) +#define RT5640_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5640_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5640_G_HP_R_RM_R_SFT 13 +#define RT5640_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5640_G_IN_R_RM_R_SFT 10 +#define RT5640_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5640_G_BST4_RM_R_SFT 7 +#define RT5640_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5640_G_BST3_RM_R_SFT 4 +#define RT5640_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5640_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5640_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5640_G_BST1_RM_R_SFT 13 +#define RT5640_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5640_G_OM_R_RM_R_SFT 10 +#define RT5640_M_HP_R_RM_R (0x1 << 6) +#define RT5640_M_HP_R_RM_R_SFT 6 +#define RT5640_M_IN_R_RM_R (0x1 << 5) +#define RT5640_M_IN_R_RM_R_SFT 5 +#define RT5640_M_BST4_RM_R (0x1 << 4) +#define RT5640_M_BST4_RM_R_SFT 4 +#define RT5640_M_BST3_RM_R (0x1 << 3) +#define RT5640_M_BST3_RM_R_SFT 3 +#define RT5640_M_BST2_RM_R (0x1 << 2) +#define RT5640_M_BST2_RM_R_SFT 2 +#define RT5640_M_BST1_RM_R (0x1 << 1) +#define RT5640_M_BST1_RM_R_SFT 1 +#define RT5640_M_OM_R_RM_R (0x1) +#define RT5640_M_OM_R_RM_R_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5640_M_DAC2_HM (0x1 << 15) +#define RT5640_M_DAC2_HM_SFT 15 +#define RT5640_M_DAC1_HM (0x1 << 14) +#define RT5640_M_DAC1_HM_SFT 14 +#define RT5640_M_HPVOL_HM (0x1 << 13) +#define RT5640_M_HPVOL_HM_SFT 13 +#define RT5640_G_HPOMIX_MASK (0x1 << 12) +#define RT5640_G_HPOMIX_SFT 12 + +/* SPK Left Mixer Control (0x46) */ +#define RT5640_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5640_G_RM_L_SM_L_SFT 14 +#define RT5640_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5640_G_IN_L_SM_L_SFT 12 +#define RT5640_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5640_G_DAC_L1_SM_L_SFT 10 +#define RT5640_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5640_G_DAC_L2_SM_L_SFT 8 +#define RT5640_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5640_G_OM_L_SM_L_SFT 6 +#define RT5640_M_RM_L_SM_L (0x1 << 5) +#define RT5640_M_RM_L_SM_L_SFT 5 +#define RT5640_M_IN_L_SM_L (0x1 << 4) +#define RT5640_M_IN_L_SM_L_SFT 4 +#define RT5640_M_DAC_L1_SM_L (0x1 << 3) +#define RT5640_M_DAC_L1_SM_L_SFT 3 +#define RT5640_M_DAC_L2_SM_L (0x1 << 2) +#define RT5640_M_DAC_L2_SM_L_SFT 2 +#define RT5640_M_OM_L_SM_L (0x1 << 1) +#define RT5640_M_OM_L_SM_L_SFT 1 + +/* SPK Right Mixer Control (0x47) */ +#define RT5640_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5640_G_RM_R_SM_R_SFT 14 +#define RT5640_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5640_G_IN_R_SM_R_SFT 12 +#define RT5640_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5640_G_DAC_R1_SM_R_SFT 10 +#define RT5640_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5640_G_DAC_R2_SM_R_SFT 8 +#define RT5640_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5640_G_OM_R_SM_R_SFT 6 +#define RT5640_M_RM_R_SM_R (0x1 << 5) +#define RT5640_M_RM_R_SM_R_SFT 5 +#define RT5640_M_IN_R_SM_R (0x1 << 4) +#define RT5640_M_IN_R_SM_R_SFT 4 +#define RT5640_M_DAC_R1_SM_R (0x1 << 3) +#define RT5640_M_DAC_R1_SM_R_SFT 3 +#define RT5640_M_DAC_R2_SM_R (0x1 << 2) +#define RT5640_M_DAC_R2_SM_R_SFT 2 +#define RT5640_M_OM_R_SM_R (0x1 << 1) +#define RT5640_M_OM_R_SM_R_SFT 1 + +/* SPOLMIX Control (0x48) */ +#define RT5640_M_DAC_R1_SPM_L (0x1 << 15) +#define RT5640_M_DAC_R1_SPM_L_SFT 15 +#define RT5640_M_DAC_L1_SPM_L (0x1 << 14) +#define RT5640_M_DAC_L1_SPM_L_SFT 14 +#define RT5640_M_SV_R_SPM_L (0x1 << 13) +#define RT5640_M_SV_R_SPM_L_SFT 13 +#define RT5640_M_SV_L_SPM_L (0x1 << 12) +#define RT5640_M_SV_L_SPM_L_SFT 12 +#define RT5640_M_BST1_SPM_L (0x1 << 11) +#define RT5640_M_BST1_SPM_L_SFT 11 + +/* SPORMIX Control (0x49) */ +#define RT5640_M_DAC_R1_SPM_R (0x1 << 13) +#define RT5640_M_DAC_R1_SPM_R_SFT 13 +#define RT5640_M_SV_R_SPM_R (0x1 << 12) +#define RT5640_M_SV_R_SPM_R_SFT 12 +#define RT5640_M_BST1_SPM_R (0x1 << 11) +#define RT5640_M_BST1_SPM_R_SFT 11 + +/* SPOLMIX / SPORMIX Ratio Control (0x4a) */ +#define RT5640_SPO_CLSD_RATIO_MASK (0x7) +#define RT5640_SPO_CLSD_RATIO_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5640_M_DAC_R2_MM (0x1 << 15) +#define RT5640_M_DAC_R2_MM_SFT 15 +#define RT5640_M_DAC_L2_MM (0x1 << 14) +#define RT5640_M_DAC_L2_MM_SFT 14 +#define RT5640_M_OV_R_MM (0x1 << 13) +#define RT5640_M_OV_R_MM_SFT 13 +#define RT5640_M_OV_L_MM (0x1 << 12) +#define RT5640_M_OV_L_MM_SFT 12 +#define RT5640_M_BST1_MM (0x1 << 11) +#define RT5640_M_BST1_MM_SFT 11 +#define RT5640_G_MONOMIX_MASK (0x1 << 10) +#define RT5640_G_MONOMIX_SFT 10 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5640_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5640_G_BST3_OM_L_SFT 13 +#define RT5640_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5640_G_BST2_OM_L_SFT 10 +#define RT5640_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5640_G_BST1_OM_L_SFT 7 +#define RT5640_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5640_G_IN_L_OM_L_SFT 4 +#define RT5640_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5640_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5640_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5640_G_DAC_R2_OM_L_SFT 13 +#define RT5640_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5640_G_DAC_L2_OM_L_SFT 10 +#define RT5640_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5640_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5640_M_SM_L_OM_L (0x1 << 8) +#define RT5640_M_SM_L_OM_L_SFT 8 +#define RT5640_M_BST3_OM_L (0x1 << 7) +#define RT5640_M_BST3_OM_L_SFT 7 +#define RT5640_M_BST2_OM_L (0x1 << 6) +#define RT5640_M_BST2_OM_L_SFT 6 +#define RT5640_M_BST1_OM_L (0x1 << 5) +#define RT5640_M_BST1_OM_L_SFT 5 +#define RT5640_M_IN_L_OM_L (0x1 << 4) +#define RT5640_M_IN_L_OM_L_SFT 4 +#define RT5640_M_RM_L_OM_L (0x1 << 3) +#define RT5640_M_RM_L_OM_L_SFT 3 +#define RT5640_M_DAC_R2_OM_L (0x1 << 2) +#define RT5640_M_DAC_R2_OM_L_SFT 2 +#define RT5640_M_DAC_L2_OM_L (0x1 << 1) +#define RT5640_M_DAC_L2_OM_L_SFT 1 +#define RT5640_M_DAC_L1_OM_L (0x1) +#define RT5640_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5640_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5640_G_BST4_OM_R_SFT 13 +#define RT5640_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5640_G_BST2_OM_R_SFT 10 +#define RT5640_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5640_G_BST1_OM_R_SFT 7 +#define RT5640_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5640_G_IN_R_OM_R_SFT 4 +#define RT5640_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5640_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5640_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5640_G_DAC_L2_OM_R_SFT 13 +#define RT5640_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5640_G_DAC_R2_OM_R_SFT 10 +#define RT5640_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5640_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5640_M_SM_L_OM_R (0x1 << 8) +#define RT5640_M_SM_L_OM_R_SFT 8 +#define RT5640_M_BST4_OM_R (0x1 << 7) +#define RT5640_M_BST4_OM_R_SFT 7 +#define RT5640_M_BST2_OM_R (0x1 << 6) +#define RT5640_M_BST2_OM_R_SFT 6 +#define RT5640_M_BST1_OM_R (0x1 << 5) +#define RT5640_M_BST1_OM_R_SFT 5 +#define RT5640_M_IN_R_OM_R (0x1 << 4) +#define RT5640_M_IN_R_OM_R_SFT 4 +#define RT5640_M_RM_R_OM_R (0x1 << 3) +#define RT5640_M_RM_R_OM_R_SFT 3 +#define RT5640_M_DAC_L2_OM_R (0x1 << 2) +#define RT5640_M_DAC_L2_OM_R_SFT 2 +#define RT5640_M_DAC_R2_OM_R (0x1 << 1) +#define RT5640_M_DAC_R2_OM_R_SFT 1 +#define RT5640_M_DAC_R1_OM_R (0x1) +#define RT5640_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5640_M_DAC_L1_LM (0x1 << 15) +#define RT5640_M_DAC_L1_LM_SFT 15 +#define RT5640_M_DAC_R1_LM (0x1 << 14) +#define RT5640_M_DAC_R1_LM_SFT 14 +#define RT5640_M_OV_L_LM (0x1 << 13) +#define RT5640_M_OV_L_LM_SFT 13 +#define RT5640_M_OV_R_LM (0x1 << 12) +#define RT5640_M_OV_R_LM_SFT 12 +#define RT5640_G_LOUTMIX_MASK (0x1 << 11) +#define RT5640_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5640_PWR_I2S1 (0x1 << 15) +#define RT5640_PWR_I2S1_BIT 15 +#define RT5640_PWR_I2S2 (0x1 << 14) +#define RT5640_PWR_I2S2_BIT 14 +#define RT5640_PWR_DAC_L1 (0x1 << 12) +#define RT5640_PWR_DAC_L1_BIT 12 +#define RT5640_PWR_DAC_R1 (0x1 << 11) +#define RT5640_PWR_DAC_R1_BIT 11 +#define RT5640_PWR_DAC_L2 (0x1 << 7) +#define RT5640_PWR_DAC_L2_BIT 7 +#define RT5640_PWR_DAC_R2 (0x1 << 6) +#define RT5640_PWR_DAC_R2_BIT 6 +#define RT5640_PWR_ADC_L (0x1 << 2) +#define RT5640_PWR_ADC_L_BIT 2 +#define RT5640_PWR_ADC_R (0x1 << 1) +#define RT5640_PWR_ADC_R_BIT 1 +#define RT5640_PWR_CLS_D (0x1) +#define RT5640_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5640_PWR_ADC_SF (0x1 << 15) +#define RT5640_PWR_ADC_SF_BIT 15 +#define RT5640_PWR_ADC_MF_L (0x1 << 14) +#define RT5640_PWR_ADC_MF_L_BIT 14 +#define RT5640_PWR_ADC_MF_R (0x1 << 13) +#define RT5640_PWR_ADC_MF_R_BIT 13 +#define RT5640_PWR_I2S_DSP (0x1 << 12) +#define RT5640_PWR_I2S_DSP_BIT 12 + +/* Power Management for Analog 1 (0x63) */ +#define RT5640_PWR_VREF1 (0x1 << 15) +#define RT5640_PWR_VREF1_BIT 15 +#define RT5640_PWR_FV1 (0x1 << 14) +#define RT5640_PWR_FV1_BIT 14 +#define RT5640_PWR_MB (0x1 << 13) +#define RT5640_PWR_MB_BIT 13 +#define RT5640_PWR_LM (0x1 << 12) +#define RT5640_PWR_LM_BIT 12 +#define RT5640_PWR_BG (0x1 << 11) +#define RT5640_PWR_BG_BIT 11 +#define RT5640_PWR_MM (0x1 << 10) +#define RT5640_PWR_MM_BIT 10 +#define RT5640_PWR_MA (0x1 << 8) +#define RT5640_PWR_MA_BIT 8 +#define RT5640_PWR_HP_L (0x1 << 7) +#define RT5640_PWR_HP_L_BIT 7 +#define RT5640_PWR_HP_R (0x1 << 6) +#define RT5640_PWR_HP_R_BIT 6 +#define RT5640_PWR_HA (0x1 << 5) +#define RT5640_PWR_HA_BIT 5 +#define RT5640_PWR_VREF2 (0x1 << 4) +#define RT5640_PWR_VREF2_BIT 4 +#define RT5640_PWR_FV2 (0x1 << 3) +#define RT5640_PWR_FV2_BIT 3 +#define RT5640_PWR_LDO2 (0x1 << 2) +#define RT5640_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5640_PWR_BST1 (0x1 << 15) +#define RT5640_PWR_BST1_BIT 15 +#define RT5640_PWR_BST2 (0x1 << 14) +#define RT5640_PWR_BST2_BIT 14 +#define RT5640_PWR_BST3 (0x1 << 13) +#define RT5640_PWR_BST3_BIT 13 +#define RT5640_PWR_BST4 (0x1 << 12) +#define RT5640_PWR_BST4_BIT 12 +#define RT5640_PWR_MB1 (0x1 << 11) +#define RT5640_PWR_MB1_BIT 11 +#define RT5640_PWR_PLL (0x1 << 9) +#define RT5640_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5640_PWR_OM_L (0x1 << 15) +#define RT5640_PWR_OM_L_BIT 15 +#define RT5640_PWR_OM_R (0x1 << 14) +#define RT5640_PWR_OM_R_BIT 14 +#define RT5640_PWR_SM_L (0x1 << 13) +#define RT5640_PWR_SM_L_BIT 13 +#define RT5640_PWR_SM_R (0x1 << 12) +#define RT5640_PWR_SM_R_BIT 12 +#define RT5640_PWR_RM_L (0x1 << 11) +#define RT5640_PWR_RM_L_BIT 11 +#define RT5640_PWR_RM_R (0x1 << 10) +#define RT5640_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5640_PWR_SV_L (0x1 << 15) +#define RT5640_PWR_SV_L_BIT 15 +#define RT5640_PWR_SV_R (0x1 << 14) +#define RT5640_PWR_SV_R_BIT 14 +#define RT5640_PWR_OV_L (0x1 << 13) +#define RT5640_PWR_OV_L_BIT 13 +#define RT5640_PWR_OV_R (0x1 << 12) +#define RT5640_PWR_OV_R_BIT 12 +#define RT5640_PWR_HV_L (0x1 << 11) +#define RT5640_PWR_HV_L_BIT 11 +#define RT5640_PWR_HV_R (0x1 << 10) +#define RT5640_PWR_HV_R_BIT 10 +#define RT5640_PWR_IN_L (0x1 << 9) +#define RT5640_PWR_IN_L_BIT 9 +#define RT5640_PWR_IN_R (0x1 << 8) +#define RT5640_PWR_IN_R_BIT 8 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5640_I2S_MS_MASK (0x1 << 15) +#define RT5640_I2S_MS_SFT 15 +#define RT5640_I2S_MS_M (0x0 << 15) +#define RT5640_I2S_MS_S (0x1 << 15) +#define RT5640_I2S_IF_MASK (0x7 << 12) +#define RT5640_I2S_IF_SFT 12 +#define RT5640_I2S_O_CP_MASK (0x3 << 10) +#define RT5640_I2S_O_CP_SFT 10 +#define RT5640_I2S_O_CP_OFF (0x0 << 10) +#define RT5640_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5640_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5640_I2S_I_CP_MASK (0x3 << 8) +#define RT5640_I2S_I_CP_SFT 8 +#define RT5640_I2S_I_CP_OFF (0x0 << 8) +#define RT5640_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5640_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5640_I2S_BP_MASK (0x1 << 7) +#define RT5640_I2S_BP_SFT 7 +#define RT5640_I2S_BP_NOR (0x0 << 7) +#define RT5640_I2S_BP_INV (0x1 << 7) +#define RT5640_I2S_DL_MASK (0x3 << 2) +#define RT5640_I2S_DL_SFT 2 +#define RT5640_I2S_DL_16 (0x0 << 2) +#define RT5640_I2S_DL_20 (0x1 << 2) +#define RT5640_I2S_DL_24 (0x2 << 2) +#define RT5640_I2S_DL_8 (0x3 << 2) +#define RT5640_I2S_DF_MASK (0x3) +#define RT5640_I2S_DF_SFT 0 +#define RT5640_I2S_DF_I2S (0x0) +#define RT5640_I2S_DF_LEFT (0x1) +#define RT5640_I2S_DF_PCM_A (0x2) +#define RT5640_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5640_I2S2_SDI_MASK (0x1 << 6) +#define RT5640_I2S2_SDI_SFT 6 +#define RT5640_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5640_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5640_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5640_I2S_BCLK_MS1_SFT 15 +#define RT5640_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5640_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5640_I2S_PD1_MASK (0x7 << 12) +#define RT5640_I2S_PD1_SFT 12 +#define RT5640_I2S_PD1_1 (0x0 << 12) +#define RT5640_I2S_PD1_2 (0x1 << 12) +#define RT5640_I2S_PD1_3 (0x2 << 12) +#define RT5640_I2S_PD1_4 (0x3 << 12) +#define RT5640_I2S_PD1_6 (0x4 << 12) +#define RT5640_I2S_PD1_8 (0x5 << 12) +#define RT5640_I2S_PD1_12 (0x6 << 12) +#define RT5640_I2S_PD1_16 (0x7 << 12) +#define RT5640_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5640_I2S_BCLK_MS2_SFT 11 +#define RT5640_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5640_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5640_I2S_PD2_MASK (0x7 << 8) +#define RT5640_I2S_PD2_SFT 8 +#define RT5640_I2S_PD2_1 (0x0 << 8) +#define RT5640_I2S_PD2_2 (0x1 << 8) +#define RT5640_I2S_PD2_3 (0x2 << 8) +#define RT5640_I2S_PD2_4 (0x3 << 8) +#define RT5640_I2S_PD2_6 (0x4 << 8) +#define RT5640_I2S_PD2_8 (0x5 << 8) +#define RT5640_I2S_PD2_12 (0x6 << 8) +#define RT5640_I2S_PD2_16 (0x7 << 8) +#define RT5640_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5640_I2S_BCLK_MS3_SFT 7 +#define RT5640_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5640_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5640_I2S_PD3_MASK (0x7 << 4) +#define RT5640_I2S_PD3_SFT 4 +#define RT5640_I2S_PD3_1 (0x0 << 4) +#define RT5640_I2S_PD3_2 (0x1 << 4) +#define RT5640_I2S_PD3_3 (0x2 << 4) +#define RT5640_I2S_PD3_4 (0x3 << 4) +#define RT5640_I2S_PD3_6 (0x4 << 4) +#define RT5640_I2S_PD3_8 (0x5 << 4) +#define RT5640_I2S_PD3_12 (0x6 << 4) +#define RT5640_I2S_PD3_16 (0x7 << 4) +#define RT5640_DAC_OSR_MASK (0x3 << 2) +#define RT5640_DAC_OSR_SFT 2 +#define RT5640_DAC_OSR_128 (0x0 << 2) +#define RT5640_DAC_OSR_64 (0x1 << 2) +#define RT5640_DAC_OSR_32 (0x2 << 2) +#define RT5640_DAC_OSR_16 (0x3 << 2) +#define RT5640_ADC_OSR_MASK (0x3) +#define RT5640_ADC_OSR_SFT 0 +#define RT5640_ADC_OSR_128 (0x0) +#define RT5640_ADC_OSR_64 (0x1) +#define RT5640_ADC_OSR_32 (0x2) +#define RT5640_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5640_DAC_L_OSR_MASK (0x3 << 14) +#define RT5640_DAC_L_OSR_SFT 14 +#define RT5640_DAC_L_OSR_128 (0x0 << 14) +#define RT5640_DAC_L_OSR_64 (0x1 << 14) +#define RT5640_DAC_L_OSR_32 (0x2 << 14) +#define RT5640_DAC_L_OSR_16 (0x3 << 14) +#define RT5640_ADC_R_OSR_MASK (0x3 << 12) +#define RT5640_ADC_R_OSR_SFT 12 +#define RT5640_ADC_R_OSR_128 (0x0 << 12) +#define RT5640_ADC_R_OSR_64 (0x1 << 12) +#define RT5640_ADC_R_OSR_32 (0x2 << 12) +#define RT5640_ADC_R_OSR_16 (0x3 << 12) +#define RT5640_DAHPF_EN (0x1 << 11) +#define RT5640_DAHPF_EN_SFT 11 +#define RT5640_ADHPF_EN (0x1 << 10) +#define RT5640_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5640_DMIC_1_EN_MASK (0x1 << 15) +#define RT5640_DMIC_1_EN_SFT 15 +#define RT5640_DMIC_1_DIS (0x0 << 15) +#define RT5640_DMIC_1_EN (0x1 << 15) +#define RT5640_DMIC_2_EN_MASK (0x1 << 14) +#define RT5640_DMIC_2_EN_SFT 14 +#define RT5640_DMIC_2_DIS (0x0 << 14) +#define RT5640_DMIC_2_EN (0x1 << 14) +#define RT5640_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5640_DMIC_1L_LH_SFT 13 +#define RT5640_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5640_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5640_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5640_DMIC_1R_LH_SFT 12 +#define RT5640_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5640_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5640_DMIC_1_DP_MASK (0x1 << 11) +#define RT5640_DMIC_1_DP_SFT 11 +#define RT5640_DMIC_1_DP_GPIO3 (0x0 << 11) +#define RT5640_DMIC_1_DP_IN1P (0x1 << 11) +#define RT5640_DMIC_2_DP_MASK (0x1 << 10) +#define RT5640_DMIC_2_DP_SFT 10 +#define RT5640_DMIC_2_DP_GPIO4 (0x0 << 10) +#define RT5640_DMIC_2_DP_IN1N (0x1 << 10) +#define RT5640_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5640_DMIC_2L_LH_SFT 9 +#define RT5640_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5640_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5640_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5640_DMIC_2R_LH_SFT 8 +#define RT5640_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5640_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5640_DMIC_CLK_MASK (0x7 << 5) +#define RT5640_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5640_SCLK_SRC_MASK (0x3 << 14) +#define RT5640_SCLK_SRC_SFT 14 +#define RT5640_SCLK_SRC_MCLK (0x0 << 14) +#define RT5640_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5640_SCLK_SRC_PLL1T (0x2 << 14) +#define RT5640_SCLK_SRC_RCCLK (0x3 << 14) /* 15MHz */ +#define RT5640_PLL1_SRC_MASK (0x3 << 12) +#define RT5640_PLL1_SRC_SFT 12 +#define RT5640_PLL1_SRC_MCLK (0x0 << 12) +#define RT5640_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5640_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5640_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5640_PLL1_PD_MASK (0x1 << 3) +#define RT5640_PLL1_PD_SFT 3 +#define RT5640_PLL1_PD_1 (0x0 << 3) +#define RT5640_PLL1_PD_2 (0x1 << 3) + +#define RT5640_PLL_INP_MAX 40000000 +#define RT5640_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5640_PLL_N_MAX 0x1ff +#define RT5640_PLL_N_MASK (RT5640_PLL_N_MAX << 7) +#define RT5640_PLL_N_SFT 7 +#define RT5640_PLL_K_MAX 0x1f +#define RT5640_PLL_K_MASK (RT5640_PLL_K_MAX) +#define RT5640_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5640_PLL_M_MAX 0xf +#define RT5640_PLL_M_MASK (RT5640_PLL_M_MAX << 12) +#define RT5640_PLL_M_SFT 12 +#define RT5640_PLL_M_BP (0x1 << 11) +#define RT5640_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5640_STO_T_MASK (0x1 << 15) +#define RT5640_STO_T_SFT 15 +#define RT5640_STO_T_SCLK (0x0 << 15) +#define RT5640_STO_T_LRCK1 (0x1 << 15) +#define RT5640_M1_T_MASK (0x1 << 14) +#define RT5640_M1_T_SFT 14 +#define RT5640_M1_T_I2S2 (0x0 << 14) +#define RT5640_M1_T_I2S2_D3 (0x1 << 14) +#define RT5640_I2S2_F_MASK (0x1 << 12) +#define RT5640_I2S2_F_SFT 12 +#define RT5640_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5640_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5640_DMIC_1_M_MASK (0x1 << 9) +#define RT5640_DMIC_1_M_SFT 9 +#define RT5640_DMIC_1_M_NOR (0x0 << 9) +#define RT5640_DMIC_1_M_ASYN (0x1 << 9) +#define RT5640_DMIC_2_M_MASK (0x1 << 8) +#define RT5640_DMIC_2_M_SFT 8 +#define RT5640_DMIC_2_M_NOR (0x0 << 8) +#define RT5640_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5640_MDA_L_M_MASK (0x1 << 15) +#define RT5640_MDA_L_M_SFT 15 +#define RT5640_MDA_L_M_NOR (0x0 << 15) +#define RT5640_MDA_L_M_ASYN (0x1 << 15) +#define RT5640_MDA_R_M_MASK (0x1 << 14) +#define RT5640_MDA_R_M_SFT 14 +#define RT5640_MDA_R_M_NOR (0x0 << 14) +#define RT5640_MDA_R_M_ASYN (0x1 << 14) +#define RT5640_MAD_L_M_MASK (0x1 << 13) +#define RT5640_MAD_L_M_SFT 13 +#define RT5640_MAD_L_M_NOR (0x0 << 13) +#define RT5640_MAD_L_M_ASYN (0x1 << 13) +#define RT5640_MAD_R_M_MASK (0x1 << 12) +#define RT5640_MAD_R_M_SFT 12 +#define RT5640_MAD_R_M_NOR (0x0 << 12) +#define RT5640_MAD_R_M_ASYN (0x1 << 12) +#define RT5640_ADC_M_MASK (0x1 << 11) +#define RT5640_ADC_M_SFT 11 +#define RT5640_ADC_M_NOR (0x0 << 11) +#define RT5640_ADC_M_ASYN (0x1 << 11) +#define RT5640_STO_DAC_M_MASK (0x1 << 5) +#define RT5640_STO_DAC_M_SFT 5 +#define RT5640_STO_DAC_M_NOR (0x0 << 5) +#define RT5640_STO_DAC_M_ASYN (0x1 << 5) +#define RT5640_I2S1_R_D_MASK (0x1 << 4) +#define RT5640_I2S1_R_D_SFT 4 +#define RT5640_I2S1_R_D_DIS (0x0 << 4) +#define RT5640_I2S1_R_D_EN (0x1 << 4) +#define RT5640_I2S2_R_D_MASK (0x1 << 3) +#define RT5640_I2S2_R_D_SFT 3 +#define RT5640_I2S2_R_D_DIS (0x0 << 3) +#define RT5640_I2S2_R_D_EN (0x1 << 3) +#define RT5640_PRE_SCLK_MASK (0x3) +#define RT5640_PRE_SCLK_SFT 0 +#define RT5640_PRE_SCLK_512 (0x0) +#define RT5640_PRE_SCLK_1024 (0x1) +#define RT5640_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5640_I2S1_RATE_MASK (0xf << 12) +#define RT5640_I2S1_RATE_SFT 12 +#define RT5640_I2S2_RATE_MASK (0xf << 8) +#define RT5640_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5640_I2S1_PD_MASK (0x7 << 12) +#define RT5640_I2S1_PD_SFT 12 +#define RT5640_I2S2_PD_MASK (0x7 << 8) +#define RT5640_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5640_HP_OVCD_MASK (0x1 << 10) +#define RT5640_HP_OVCD_SFT 10 +#define RT5640_HP_OVCD_DIS (0x0 << 10) +#define RT5640_HP_OVCD_EN (0x1 << 10) +#define RT5640_HP_OC_TH_MASK (0x3 << 8) +#define RT5640_HP_OC_TH_SFT 8 +#define RT5640_HP_OC_TH_90 (0x0 << 8) +#define RT5640_HP_OC_TH_105 (0x1 << 8) +#define RT5640_HP_OC_TH_120 (0x2 << 8) +#define RT5640_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5640_CLSD_OC_MASK (0x1 << 9) +#define RT5640_CLSD_OC_SFT 9 +#define RT5640_CLSD_OC_PU (0x0 << 9) +#define RT5640_CLSD_OC_PD (0x1 << 9) +#define RT5640_AUTO_PD_MASK (0x1 << 8) +#define RT5640_AUTO_PD_SFT 8 +#define RT5640_AUTO_PD_DIS (0x0 << 8) +#define RT5640_AUTO_PD_EN (0x1 << 8) +#define RT5640_CLSD_OC_TH_MASK (0x3f) +#define RT5640_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5640_CLSD_RATIO_MASK (0xf << 12) +#define RT5640_CLSD_RATIO_SFT 12 +#define RT5640_CLSD_OM_MASK (0x1 << 11) +#define RT5640_CLSD_OM_SFT 11 +#define RT5640_CLSD_OM_MONO (0x0 << 11) +#define RT5640_CLSD_OM_STO (0x1 << 11) +#define RT5640_CLSD_SCH_MASK (0x1 << 10) +#define RT5640_CLSD_SCH_SFT 10 +#define RT5640_CLSD_SCH_L (0x0 << 10) +#define RT5640_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5640_SMT_TRIG_MASK (0x1 << 15) +#define RT5640_SMT_TRIG_SFT 15 +#define RT5640_SMT_TRIG_DIS (0x0 << 15) +#define RT5640_SMT_TRIG_EN (0x1 << 15) +#define RT5640_HP_L_SMT_MASK (0x1 << 9) +#define RT5640_HP_L_SMT_SFT 9 +#define RT5640_HP_L_SMT_DIS (0x0 << 9) +#define RT5640_HP_L_SMT_EN (0x1 << 9) +#define RT5640_HP_R_SMT_MASK (0x1 << 8) +#define RT5640_HP_R_SMT_SFT 8 +#define RT5640_HP_R_SMT_DIS (0x0 << 8) +#define RT5640_HP_R_SMT_EN (0x1 << 8) +#define RT5640_HP_CD_PD_MASK (0x1 << 7) +#define RT5640_HP_CD_PD_SFT 7 +#define RT5640_HP_CD_PD_DIS (0x0 << 7) +#define RT5640_HP_CD_PD_EN (0x1 << 7) +#define RT5640_RSTN_MASK (0x1 << 6) +#define RT5640_RSTN_SFT 6 +#define RT5640_RSTN_DIS (0x0 << 6) +#define RT5640_RSTN_EN (0x1 << 6) +#define RT5640_RSTP_MASK (0x1 << 5) +#define RT5640_RSTP_SFT 5 +#define RT5640_RSTP_DIS (0x0 << 5) +#define RT5640_RSTP_EN (0x1 << 5) +#define RT5640_HP_CO_MASK (0x1 << 4) +#define RT5640_HP_CO_SFT 4 +#define RT5640_HP_CO_DIS (0x0 << 4) +#define RT5640_HP_CO_EN (0x1 << 4) +#define RT5640_HP_CP_MASK (0x1 << 3) +#define RT5640_HP_CP_SFT 3 +#define RT5640_HP_CP_PD (0x0 << 3) +#define RT5640_HP_CP_PU (0x1 << 3) +#define RT5640_HP_SG_MASK (0x1 << 2) +#define RT5640_HP_SG_SFT 2 +#define RT5640_HP_SG_DIS (0x0 << 2) +#define RT5640_HP_SG_EN (0x1 << 2) +#define RT5640_HP_DP_MASK (0x1 << 1) +#define RT5640_HP_DP_SFT 1 +#define RT5640_HP_DP_PD (0x0 << 1) +#define RT5640_HP_DP_PU (0x1 << 1) +#define RT5640_HP_CB_MASK (0x1) +#define RT5640_HP_CB_SFT 0 +#define RT5640_HP_CB_PD (0x0) +#define RT5640_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5640_DEPOP_MASK (0x1 << 13) +#define RT5640_DEPOP_SFT 13 +#define RT5640_DEPOP_AUTO (0x0 << 13) +#define RT5640_DEPOP_MAN (0x1 << 13) +#define RT5640_RAMP_MASK (0x1 << 12) +#define RT5640_RAMP_SFT 12 +#define RT5640_RAMP_DIS (0x0 << 12) +#define RT5640_RAMP_EN (0x1 << 12) +#define RT5640_BPS_MASK (0x1 << 11) +#define RT5640_BPS_SFT 11 +#define RT5640_BPS_DIS (0x0 << 11) +#define RT5640_BPS_EN (0x1 << 11) +#define RT5640_FAST_UPDN_MASK (0x1 << 10) +#define RT5640_FAST_UPDN_SFT 10 +#define RT5640_FAST_UPDN_DIS (0x0 << 10) +#define RT5640_FAST_UPDN_EN (0x1 << 10) +#define RT5640_MRES_MASK (0x3 << 8) +#define RT5640_MRES_SFT 8 +#define RT5640_MRES_15MO (0x0 << 8) +#define RT5640_MRES_25MO (0x1 << 8) +#define RT5640_MRES_35MO (0x2 << 8) +#define RT5640_MRES_45MO (0x3 << 8) +#define RT5640_VLO_MASK (0x1 << 7) +#define RT5640_VLO_SFT 7 +#define RT5640_VLO_3V (0x0 << 7) +#define RT5640_VLO_32V (0x1 << 7) +#define RT5640_DIG_DP_MASK (0x1 << 6) +#define RT5640_DIG_DP_SFT 6 +#define RT5640_DIG_DP_DIS (0x0 << 6) +#define RT5640_DIG_DP_EN (0x1 << 6) +#define RT5640_DP_TH_MASK (0x3 << 4) +#define RT5640_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5640_CP_SYS_MASK (0x7 << 12) +#define RT5640_CP_SYS_SFT 12 +#define RT5640_CP_FQ1_MASK (0x7 << 8) +#define RT5640_CP_FQ1_SFT 8 +#define RT5640_CP_FQ2_MASK (0x7 << 4) +#define RT5640_CP_FQ2_SFT 4 +#define RT5640_CP_FQ3_MASK (0x7) +#define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5640_OSW_L_MASK (0x1 << 11) +#define RT5640_OSW_L_SFT 11 +#define RT5640_OSW_L_DIS (0x0 << 11) +#define RT5640_OSW_L_EN (0x1 << 11) +#define RT5640_OSW_R_MASK (0x1 << 10) +#define RT5640_OSW_R_SFT 10 +#define RT5640_OSW_R_DIS (0x0 << 10) +#define RT5640_OSW_R_EN (0x1 << 10) +#define RT5640_PM_HP_MASK (0x3 << 8) +#define RT5640_PM_HP_SFT 8 +#define RT5640_PM_HP_LV (0x0 << 8) +#define RT5640_PM_HP_MV (0x1 << 8) +#define RT5640_PM_HP_HV (0x2 << 8) +#define RT5640_IB_HP_MASK (0x3 << 6) +#define RT5640_IB_HP_SFT 6 +#define RT5640_IB_HP_125IL (0x0 << 6) +#define RT5640_IB_HP_25IL (0x1 << 6) +#define RT5640_IB_HP_5IL (0x2 << 6) +#define RT5640_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5640_PVDD_DET_MASK (0x1 << 15) +#define RT5640_PVDD_DET_SFT 15 +#define RT5640_PVDD_DET_DIS (0x0 << 15) +#define RT5640_PVDD_DET_EN (0x1 << 15) +#define RT5640_SPK_AG_MASK (0x1 << 14) +#define RT5640_SPK_AG_SFT 14 +#define RT5640_SPK_AG_DIS (0x0 << 14) +#define RT5640_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5640_MIC1_BS_MASK (0x1 << 15) +#define RT5640_MIC1_BS_SFT 15 +#define RT5640_MIC1_BS_9AV (0x0 << 15) +#define RT5640_MIC1_BS_75AV (0x1 << 15) +#define RT5640_MIC2_BS_MASK (0x1 << 14) +#define RT5640_MIC2_BS_SFT 14 +#define RT5640_MIC2_BS_9AV (0x0 << 14) +#define RT5640_MIC2_BS_75AV (0x1 << 14) +#define RT5640_MIC1_CLK_MASK (0x1 << 13) +#define RT5640_MIC1_CLK_SFT 13 +#define RT5640_MIC1_CLK_DIS (0x0 << 13) +#define RT5640_MIC1_CLK_EN (0x1 << 13) +#define RT5640_MIC2_CLK_MASK (0x1 << 12) +#define RT5640_MIC2_CLK_SFT 12 +#define RT5640_MIC2_CLK_DIS (0x0 << 12) +#define RT5640_MIC2_CLK_EN (0x1 << 12) +#define RT5640_MIC1_OVCD_MASK (0x1 << 11) +#define RT5640_MIC1_OVCD_SFT 11 +#define RT5640_MIC1_OVCD_DIS (0x0 << 11) +#define RT5640_MIC1_OVCD_EN (0x1 << 11) +#define RT5640_MIC1_OVTH_MASK (0x3 << 9) +#define RT5640_MIC1_OVTH_SFT 9 +#define RT5640_MIC1_OVTH_600UA (0x0 << 9) +#define RT5640_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5640_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5640_MIC2_OVCD_MASK (0x1 << 8) +#define RT5640_MIC2_OVCD_SFT 8 +#define RT5640_MIC2_OVCD_DIS (0x0 << 8) +#define RT5640_MIC2_OVCD_EN (0x1 << 8) +#define RT5640_MIC2_OVTH_MASK (0x3 << 6) +#define RT5640_MIC2_OVTH_SFT 6 +#define RT5640_MIC2_OVTH_600UA (0x0 << 6) +#define RT5640_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5640_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5640_PWR_MB_MASK (0x1 << 5) +#define RT5640_PWR_MB_SFT 5 +#define RT5640_PWR_MB_PD (0x0 << 5) +#define RT5640_PWR_MB_PU (0x1 << 5) +#define RT5640_PWR_CLK25M_MASK (0x1 << 4) +#define RT5640_PWR_CLK25M_SFT 4 +#define RT5640_PWR_CLK25M_PD (0x0 << 4) +#define RT5640_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5640_EQ_SRC_MASK (0x1 << 15) +#define RT5640_EQ_SRC_SFT 15 +#define RT5640_EQ_SRC_DAC (0x0 << 15) +#define RT5640_EQ_SRC_ADC (0x1 << 15) +#define RT5640_EQ_UPD (0x1 << 14) +#define RT5640_EQ_UPD_BIT 14 +#define RT5640_EQ_CD_MASK (0x1 << 13) +#define RT5640_EQ_CD_SFT 13 +#define RT5640_EQ_CD_DIS (0x0 << 13) +#define RT5640_EQ_CD_EN (0x1 << 13) +#define RT5640_EQ_DITH_MASK (0x3 << 8) +#define RT5640_EQ_DITH_SFT 8 +#define RT5640_EQ_DITH_NOR (0x0 << 8) +#define RT5640_EQ_DITH_LSB (0x1 << 8) +#define RT5640_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5640_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5640_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5640_EQ_HPF1_M_SFT 8 +#define RT5640_EQ_HPF1_M_HI (0x0 << 8) +#define RT5640_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5640_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5640_EQ_LPF1_M_SFT 7 +#define RT5640_EQ_LPF1_M_LO (0x0 << 7) +#define RT5640_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5640_EQ_HPF2_MASK (0x1 << 6) +#define RT5640_EQ_HPF2_SFT 6 +#define RT5640_EQ_HPF2_DIS (0x0 << 6) +#define RT5640_EQ_HPF2_EN (0x1 << 6) +#define RT5640_EQ_HPF1_MASK (0x1 << 5) +#define RT5640_EQ_HPF1_SFT 5 +#define RT5640_EQ_HPF1_DIS (0x0 << 5) +#define RT5640_EQ_HPF1_EN (0x1 << 5) +#define RT5640_EQ_BPF4_MASK (0x1 << 4) +#define RT5640_EQ_BPF4_SFT 4 +#define RT5640_EQ_BPF4_DIS (0x0 << 4) +#define RT5640_EQ_BPF4_EN (0x1 << 4) +#define RT5640_EQ_BPF3_MASK (0x1 << 3) +#define RT5640_EQ_BPF3_SFT 3 +#define RT5640_EQ_BPF3_DIS (0x0 << 3) +#define RT5640_EQ_BPF3_EN (0x1 << 3) +#define RT5640_EQ_BPF2_MASK (0x1 << 2) +#define RT5640_EQ_BPF2_SFT 2 +#define RT5640_EQ_BPF2_DIS (0x0 << 2) +#define RT5640_EQ_BPF2_EN (0x1 << 2) +#define RT5640_EQ_BPF1_MASK (0x1 << 1) +#define RT5640_EQ_BPF1_SFT 1 +#define RT5640_EQ_BPF1_DIS (0x0 << 1) +#define RT5640_EQ_BPF1_EN (0x1 << 1) +#define RT5640_EQ_LPF_MASK (0x1) +#define RT5640_EQ_LPF_SFT 0 +#define RT5640_EQ_LPF_DIS (0x0) +#define RT5640_EQ_LPF_EN (0x1) + +/* Memory Test (0xb2) */ +#define RT5640_MT_MASK (0x1 << 15) +#define RT5640_MT_SFT 15 +#define RT5640_MT_DIS (0x0 << 15) +#define RT5640_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5640_DRC_AGC_P_MASK (0x1 << 15) +#define RT5640_DRC_AGC_P_SFT 15 +#define RT5640_DRC_AGC_P_DAC (0x0 << 15) +#define RT5640_DRC_AGC_P_ADC (0x1 << 15) +#define RT5640_DRC_AGC_MASK (0x1 << 14) +#define RT5640_DRC_AGC_SFT 14 +#define RT5640_DRC_AGC_DIS (0x0 << 14) +#define RT5640_DRC_AGC_EN (0x1 << 14) +#define RT5640_DRC_AGC_UPD (0x1 << 13) +#define RT5640_DRC_AGC_UPD_BIT 13 +#define RT5640_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5640_DRC_AGC_AR_SFT 8 +#define RT5640_DRC_AGC_R_MASK (0x7 << 5) +#define RT5640_DRC_AGC_R_SFT 5 +#define RT5640_DRC_AGC_R_48K (0x1 << 5) +#define RT5640_DRC_AGC_R_96K (0x2 << 5) +#define RT5640_DRC_AGC_R_192K (0x3 << 5) +#define RT5640_DRC_AGC_R_441K (0x5 << 5) +#define RT5640_DRC_AGC_R_882K (0x6 << 5) +#define RT5640_DRC_AGC_R_1764K (0x7 << 5) +#define RT5640_DRC_AGC_RC_MASK (0x1f) +#define RT5640_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5640_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5640_DRC_AGC_POB_SFT 8 +#define RT5640_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5640_DRC_AGC_CP_SFT 7 +#define RT5640_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5640_DRC_AGC_CP_EN (0x1 << 7) +#define RT5640_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5640_DRC_AGC_CPR_SFT 5 +#define RT5640_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5640_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5640_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5640_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5640_DRC_AGC_PRB_MASK (0x1f) +#define RT5640_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5640_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5640_DRC_AGC_NGB_SFT 12 +#define RT5640_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5640_DRC_AGC_TAR_SFT 7 +#define RT5640_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5640_DRC_AGC_NG_SFT 6 +#define RT5640_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5640_DRC_AGC_NG_EN (0x1 << 6) +#define RT5640_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5640_DRC_AGC_NGH_SFT 5 +#define RT5640_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5640_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5640_DRC_AGC_NGT_MASK (0x1f) +#define RT5640_DRC_AGC_NGT_SFT 0 + +/* ANC Control 1 (0xb8) */ +#define RT5640_ANC_M_MASK (0x1 << 15) +#define RT5640_ANC_M_SFT 15 +#define RT5640_ANC_M_NOR (0x0 << 15) +#define RT5640_ANC_M_REV (0x1 << 15) +#define RT5640_ANC_MASK (0x1 << 14) +#define RT5640_ANC_SFT 14 +#define RT5640_ANC_DIS (0x0 << 14) +#define RT5640_ANC_EN (0x1 << 14) +#define RT5640_ANC_MD_MASK (0x3 << 12) +#define RT5640_ANC_MD_SFT 12 +#define RT5640_ANC_MD_DIS (0x0 << 12) +#define RT5640_ANC_MD_67MS (0x1 << 12) +#define RT5640_ANC_MD_267MS (0x2 << 12) +#define RT5640_ANC_MD_1067MS (0x3 << 12) +#define RT5640_ANC_SN_MASK (0x1 << 11) +#define RT5640_ANC_SN_SFT 11 +#define RT5640_ANC_SN_DIS (0x0 << 11) +#define RT5640_ANC_SN_EN (0x1 << 11) +#define RT5640_ANC_CLK_MASK (0x1 << 10) +#define RT5640_ANC_CLK_SFT 10 +#define RT5640_ANC_CLK_ANC (0x0 << 10) +#define RT5640_ANC_CLK_REG (0x1 << 10) +#define RT5640_ANC_ZCD_MASK (0x3 << 8) +#define RT5640_ANC_ZCD_SFT 8 +#define RT5640_ANC_ZCD_DIS (0x0 << 8) +#define RT5640_ANC_ZCD_T1 (0x1 << 8) +#define RT5640_ANC_ZCD_T2 (0x2 << 8) +#define RT5640_ANC_ZCD_WT (0x3 << 8) +#define RT5640_ANC_CS_MASK (0x1 << 7) +#define RT5640_ANC_CS_SFT 7 +#define RT5640_ANC_CS_DIS (0x0 << 7) +#define RT5640_ANC_CS_EN (0x1 << 7) +#define RT5640_ANC_SW_MASK (0x1 << 6) +#define RT5640_ANC_SW_SFT 6 +#define RT5640_ANC_SW_NOR (0x0 << 6) +#define RT5640_ANC_SW_AUTO (0x1 << 6) +#define RT5640_ANC_CO_L_MASK (0x3f) +#define RT5640_ANC_CO_L_SFT 0 + +/* ANC Control 2 (0xb6) */ +#define RT5640_ANC_FG_R_MASK (0xf << 12) +#define RT5640_ANC_FG_R_SFT 12 +#define RT5640_ANC_FG_L_MASK (0xf << 8) +#define RT5640_ANC_FG_L_SFT 8 +#define RT5640_ANC_CG_R_MASK (0xf << 4) +#define RT5640_ANC_CG_R_SFT 4 +#define RT5640_ANC_CG_L_MASK (0xf) +#define RT5640_ANC_CG_L_SFT 0 + +/* ANC Control 3 (0xb6) */ +#define RT5640_ANC_CD_MASK (0x1 << 6) +#define RT5640_ANC_CD_SFT 6 +#define RT5640_ANC_CD_BOTH (0x0 << 6) +#define RT5640_ANC_CD_IND (0x1 << 6) +#define RT5640_ANC_CO_R_MASK (0x3f) +#define RT5640_ANC_CO_R_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5640_JD_MASK (0x7 << 13) +#define RT5640_JD_SFT 13 +#define RT5640_JD_DIS (0x0 << 13) +#define RT5640_JD_GPIO1 (0x1 << 13) +#define RT5640_JD_JD1_IN4P (0x2 << 13) +#define RT5640_JD_JD2_IN4N (0x3 << 13) +#define RT5640_JD_GPIO2 (0x4 << 13) +#define RT5640_JD_GPIO3 (0x5 << 13) +#define RT5640_JD_GPIO4 (0x6 << 13) +#define RT5640_JD_HP_MASK (0x1 << 11) +#define RT5640_JD_HP_SFT 11 +#define RT5640_JD_HP_DIS (0x0 << 11) +#define RT5640_JD_HP_EN (0x1 << 11) +#define RT5640_JD_HP_TRG_MASK (0x1 << 10) +#define RT5640_JD_HP_TRG_SFT 10 +#define RT5640_JD_HP_TRG_LO (0x0 << 10) +#define RT5640_JD_HP_TRG_HI (0x1 << 10) +#define RT5640_JD_SPL_MASK (0x1 << 9) +#define RT5640_JD_SPL_SFT 9 +#define RT5640_JD_SPL_DIS (0x0 << 9) +#define RT5640_JD_SPL_EN (0x1 << 9) +#define RT5640_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5640_JD_SPL_TRG_SFT 8 +#define RT5640_JD_SPL_TRG_LO (0x0 << 8) +#define RT5640_JD_SPL_TRG_HI (0x1 << 8) +#define RT5640_JD_SPR_MASK (0x1 << 7) +#define RT5640_JD_SPR_SFT 7 +#define RT5640_JD_SPR_DIS (0x0 << 7) +#define RT5640_JD_SPR_EN (0x1 << 7) +#define RT5640_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5640_JD_SPR_TRG_SFT 6 +#define RT5640_JD_SPR_TRG_LO (0x0 << 6) +#define RT5640_JD_SPR_TRG_HI (0x1 << 6) +#define RT5640_JD_MO_MASK (0x1 << 5) +#define RT5640_JD_MO_SFT 5 +#define RT5640_JD_MO_DIS (0x0 << 5) +#define RT5640_JD_MO_EN (0x1 << 5) +#define RT5640_JD_MO_TRG_MASK (0x1 << 4) +#define RT5640_JD_MO_TRG_SFT 4 +#define RT5640_JD_MO_TRG_LO (0x0 << 4) +#define RT5640_JD_MO_TRG_HI (0x1 << 4) +#define RT5640_JD_LO_MASK (0x1 << 3) +#define RT5640_JD_LO_SFT 3 +#define RT5640_JD_LO_DIS (0x0 << 3) +#define RT5640_JD_LO_EN (0x1 << 3) +#define RT5640_JD_LO_TRG_MASK (0x1 << 2) +#define RT5640_JD_LO_TRG_SFT 2 +#define RT5640_JD_LO_TRG_LO (0x0 << 2) +#define RT5640_JD_LO_TRG_HI (0x1 << 2) +#define RT5640_JD1_IN4P_MASK (0x1 << 1) +#define RT5640_JD1_IN4P_SFT 1 +#define RT5640_JD1_IN4P_DIS (0x0 << 1) +#define RT5640_JD1_IN4P_EN (0x1 << 1) +#define RT5640_JD2_IN4N_MASK (0x1) +#define RT5640_JD2_IN4N_SFT 0 +#define RT5640_JD2_IN4N_DIS (0x0) +#define RT5640_JD2_IN4N_EN (0x1) + +/* Jack detect for ANC (0xbc) */ +#define RT5640_ANC_DET_MASK (0x3 << 4) +#define RT5640_ANC_DET_SFT 4 +#define RT5640_ANC_DET_DIS (0x0 << 4) +#define RT5640_ANC_DET_MB1 (0x1 << 4) +#define RT5640_ANC_DET_MB2 (0x2 << 4) +#define RT5640_ANC_DET_JD (0x3 << 4) +#define RT5640_AD_TRG_MASK (0x1 << 3) +#define RT5640_AD_TRG_SFT 3 +#define RT5640_AD_TRG_LO (0x0 << 3) +#define RT5640_AD_TRG_HI (0x1 << 3) +#define RT5640_ANCM_DET_MASK (0x3 << 4) +#define RT5640_ANCM_DET_SFT 4 +#define RT5640_ANCM_DET_DIS (0x0 << 4) +#define RT5640_ANCM_DET_MB1 (0x1 << 4) +#define RT5640_ANCM_DET_MB2 (0x2 << 4) +#define RT5640_ANCM_DET_JD (0x3 << 4) +#define RT5640_AMD_TRG_MASK (0x1 << 3) +#define RT5640_AMD_TRG_SFT 3 +#define RT5640_AMD_TRG_LO (0x0 << 3) +#define RT5640_AMD_TRG_HI (0x1 << 3) + +/* IRQ Control 1 (0xbd) */ +#define RT5640_IRQ_JD_MASK (0x1 << 15) +#define RT5640_IRQ_JD_SFT 15 +#define RT5640_IRQ_JD_BP (0x0 << 15) +#define RT5640_IRQ_JD_NOR (0x1 << 15) +#define RT5640_IRQ_OT_MASK (0x1 << 14) +#define RT5640_IRQ_OT_SFT 14 +#define RT5640_IRQ_OT_BP (0x0 << 14) +#define RT5640_IRQ_OT_NOR (0x1 << 14) +#define RT5640_JD_STKY_MASK (0x1 << 13) +#define RT5640_JD_STKY_SFT 13 +#define RT5640_JD_STKY_DIS (0x0 << 13) +#define RT5640_JD_STKY_EN (0x1 << 13) +#define RT5640_OT_STKY_MASK (0x1 << 12) +#define RT5640_OT_STKY_SFT 12 +#define RT5640_OT_STKY_DIS (0x0 << 12) +#define RT5640_OT_STKY_EN (0x1 << 12) +#define RT5640_JD_P_MASK (0x1 << 11) +#define RT5640_JD_P_SFT 11 +#define RT5640_JD_P_NOR (0x0 << 11) +#define RT5640_JD_P_INV (0x1 << 11) +#define RT5640_OT_P_MASK (0x1 << 10) +#define RT5640_OT_P_SFT 10 +#define RT5640_OT_P_NOR (0x0 << 10) +#define RT5640_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5640_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5640_IRQ_MB1_OC_SFT 15 +#define RT5640_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5640_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5640_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5640_IRQ_MB2_OC_SFT 14 +#define RT5640_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5640_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5640_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5640_MB1_OC_STKY_SFT 11 +#define RT5640_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5640_MB1_OC_STKY_EN (0x1 << 11) +#define RT5640_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5640_MB2_OC_STKY_SFT 10 +#define RT5640_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5640_MB2_OC_STKY_EN (0x1 << 10) +#define RT5640_MB1_OC_P_MASK (0x1 << 7) +#define RT5640_MB1_OC_P_SFT 7 +#define RT5640_MB1_OC_P_NOR (0x0 << 7) +#define RT5640_MB1_OC_P_INV (0x1 << 7) +#define RT5640_MB2_OC_P_MASK (0x1 << 6) +#define RT5640_MB2_OC_P_SFT 6 +#define RT5640_MB2_OC_P_NOR (0x0 << 6) +#define RT5640_MB2_OC_P_INV (0x1 << 6) +#define RT5640_MB1_OC_CLR (0x1 << 3) +#define RT5640_MB1_OC_CLR_SFT 3 +#define RT5640_MB2_OC_CLR (0x1 << 2) +#define RT5640_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5640_GP1_PIN_MASK (0x1 << 15) +#define RT5640_GP1_PIN_SFT 15 +#define RT5640_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5640_GP1_PIN_IRQ (0x1 << 15) +#define RT5640_GP2_PIN_MASK (0x1 << 14) +#define RT5640_GP2_PIN_SFT 14 +#define RT5640_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5640_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5640_GP3_PIN_MASK (0x3 << 12) +#define RT5640_GP3_PIN_SFT 12 +#define RT5640_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5640_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5640_GP3_PIN_IRQ (0x2 << 12) +#define RT5640_GP4_PIN_MASK (0x1 << 11) +#define RT5640_GP4_PIN_SFT 11 +#define RT5640_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5640_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5640_DP_SIG_MASK (0x1 << 10) +#define RT5640_DP_SIG_SFT 10 +#define RT5640_DP_SIG_TEST (0x0 << 10) +#define RT5640_DP_SIG_AP (0x1 << 10) +#define RT5640_GPIO_M_MASK (0x1 << 9) +#define RT5640_GPIO_M_SFT 9 +#define RT5640_GPIO_M_FLT (0x0 << 9) +#define RT5640_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5640_GP4_PF_MASK (0x1 << 11) +#define RT5640_GP4_PF_SFT 11 +#define RT5640_GP4_PF_IN (0x0 << 11) +#define RT5640_GP4_PF_OUT (0x1 << 11) +#define RT5640_GP4_OUT_MASK (0x1 << 10) +#define RT5640_GP4_OUT_SFT 10 +#define RT5640_GP4_OUT_LO (0x0 << 10) +#define RT5640_GP4_OUT_HI (0x1 << 10) +#define RT5640_GP4_P_MASK (0x1 << 9) +#define RT5640_GP4_P_SFT 9 +#define RT5640_GP4_P_NOR (0x0 << 9) +#define RT5640_GP4_P_INV (0x1 << 9) +#define RT5640_GP3_PF_MASK (0x1 << 8) +#define RT5640_GP3_PF_SFT 8 +#define RT5640_GP3_PF_IN (0x0 << 8) +#define RT5640_GP3_PF_OUT (0x1 << 8) +#define RT5640_GP3_OUT_MASK (0x1 << 7) +#define RT5640_GP3_OUT_SFT 7 +#define RT5640_GP3_OUT_LO (0x0 << 7) +#define RT5640_GP3_OUT_HI (0x1 << 7) +#define RT5640_GP3_P_MASK (0x1 << 6) +#define RT5640_GP3_P_SFT 6 +#define RT5640_GP3_P_NOR (0x0 << 6) +#define RT5640_GP3_P_INV (0x1 << 6) +#define RT5640_GP2_PF_MASK (0x1 << 5) +#define RT5640_GP2_PF_SFT 5 +#define RT5640_GP2_PF_IN (0x0 << 5) +#define RT5640_GP2_PF_OUT (0x1 << 5) +#define RT5640_GP2_OUT_MASK (0x1 << 4) +#define RT5640_GP2_OUT_SFT 4 +#define RT5640_GP2_OUT_LO (0x0 << 4) +#define RT5640_GP2_OUT_HI (0x1 << 4) +#define RT5640_GP2_P_MASK (0x1 << 3) +#define RT5640_GP2_P_SFT 3 +#define RT5640_GP2_P_NOR (0x0 << 3) +#define RT5640_GP2_P_INV (0x1 << 3) +#define RT5640_GP1_PF_MASK (0x1 << 2) +#define RT5640_GP1_PF_SFT 2 +#define RT5640_GP1_PF_IN (0x0 << 2) +#define RT5640_GP1_PF_OUT (0x1 << 2) +#define RT5640_GP1_OUT_MASK (0x1 << 1) +#define RT5640_GP1_OUT_SFT 1 +#define RT5640_GP1_OUT_LO (0x0 << 1) +#define RT5640_GP1_OUT_HI (0x1 << 1) +#define RT5640_GP1_P_MASK (0x1) +#define RT5640_GP1_P_SFT 0 +#define RT5640_GP1_P_NOR (0x0) +#define RT5640_GP1_P_INV (0x1) + +/* FM34-500 Register Control 1 (0xc4) */ +#define RT5640_DSP_ADD_SFT 0 + +/* FM34-500 Register Control 2 (0xc5) */ +#define RT5640_DSP_DAT_SFT 0 + +/* FM34-500 Register Control 3 (0xc6) */ +#define RT5640_DSP_BUSY_MASK (0x1 << 15) +#define RT5640_DSP_BUSY_BIT 15 +#define RT5640_DSP_DS_MASK (0x1 << 14) +#define RT5640_DSP_DS_SFT 14 +#define RT5640_DSP_DS_FM3010 (0x1 << 14) +#define RT5640_DSP_DS_TEMP (0x1 << 14) +#define RT5640_DSP_CLK_MASK (0x3 << 12) +#define RT5640_DSP_CLK_SFT 12 +#define RT5640_DSP_CLK_384K (0x0 << 12) +#define RT5640_DSP_CLK_192K (0x1 << 12) +#define RT5640_DSP_CLK_96K (0x2 << 12) +#define RT5640_DSP_CLK_64K (0x3 << 12) +#define RT5640_DSP_PD_PIN_MASK (0x1 << 11) +#define RT5640_DSP_PD_PIN_SFT 11 +#define RT5640_DSP_PD_PIN_LO (0x0 << 11) +#define RT5640_DSP_PD_PIN_HI (0x1 << 11) +#define RT5640_DSP_RST_PIN_MASK (0x1 << 10) +#define RT5640_DSP_RST_PIN_SFT 10 +#define RT5640_DSP_RST_PIN_LO (0x0 << 10) +#define RT5640_DSP_RST_PIN_HI (0x1 << 10) +#define RT5640_DSP_R_EN (0x1 << 9) +#define RT5640_DSP_R_EN_BIT 9 +#define RT5640_DSP_W_EN (0x1 << 8) +#define RT5640_DSP_W_EN_BIT 8 +#define RT5640_DSP_CMD_MASK (0xff) +#define RT5640_DSP_CMD_SFT 0 +#define RT5640_DSP_CMD_MW (0x3B) /* Memory Write */ +#define RT5640_DSP_CMD_MR (0x37) /* Memory Read */ +#define RT5640_DSP_CMD_RR (0x60) /* Register Read */ +#define RT5640_DSP_CMD_RW (0x68) /* Register Write */ + +/* Programmable Register Array Control 1 (0xc8) */ +#define RT5640_REG_SEQ_MASK (0xf << 12) +#define RT5640_REG_SEQ_SFT 12 +#define RT5640_SEQ1_ST_MASK (0x1 << 11) /*RO*/ +#define RT5640_SEQ1_ST_SFT 11 +#define RT5640_SEQ1_ST_RUN (0x0 << 11) +#define RT5640_SEQ1_ST_FIN (0x1 << 11) +#define RT5640_SEQ2_ST_MASK (0x1 << 10) /*RO*/ +#define RT5640_SEQ2_ST_SFT 10 +#define RT5640_SEQ2_ST_RUN (0x0 << 10) +#define RT5640_SEQ2_ST_FIN (0x1 << 10) +#define RT5640_REG_LV_MASK (0x1 << 9) +#define RT5640_REG_LV_SFT 9 +#define RT5640_REG_LV_MX (0x0 << 9) +#define RT5640_REG_LV_PR (0x1 << 9) +#define RT5640_SEQ_2_PT_MASK (0x1 << 8) +#define RT5640_SEQ_2_PT_BIT 8 +#define RT5640_REG_IDX_MASK (0xff) +#define RT5640_REG_IDX_SFT 0 + +/* Programmable Register Array Control 2 (0xc9) */ +#define RT5640_REG_DAT_MASK (0xffff) +#define RT5640_REG_DAT_SFT 0 + +/* Programmable Register Array Control 3 (0xca) */ +#define RT5640_SEQ_DLY_MASK (0xff << 8) +#define RT5640_SEQ_DLY_SFT 8 +#define RT5640_PROG_MASK (0x1 << 7) +#define RT5640_PROG_SFT 7 +#define RT5640_PROG_DIS (0x0 << 7) +#define RT5640_PROG_EN (0x1 << 7) +#define RT5640_SEQ1_PT_RUN (0x1 << 6) +#define RT5640_SEQ1_PT_RUN_BIT 6 +#define RT5640_SEQ2_PT_RUN (0x1 << 5) +#define RT5640_SEQ2_PT_RUN_BIT 5 + +/* Programmable Register Array Control 4 (0xcb) */ +#define RT5640_SEQ1_START_MASK (0xf << 8) +#define RT5640_SEQ1_START_SFT 8 +#define RT5640_SEQ1_END_MASK (0xf) +#define RT5640_SEQ1_END_SFT 0 + +/* Programmable Register Array Control 5 (0xcc) */ +#define RT5640_SEQ2_START_MASK (0xf << 8) +#define RT5640_SEQ2_START_SFT 8 +#define RT5640_SEQ2_END_MASK (0xf) +#define RT5640_SEQ2_END_SFT 0 + +/* Scramble Function (0xcd) */ +#define RT5640_SCB_KEY_MASK (0xff) +#define RT5640_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5640_SCB_SWAP_MASK (0x1 << 15) +#define RT5640_SCB_SWAP_SFT 15 +#define RT5640_SCB_SWAP_DIS (0x0 << 15) +#define RT5640_SCB_SWAP_EN (0x1 << 15) +#define RT5640_SCB_MASK (0x1 << 14) +#define RT5640_SCB_SFT 14 +#define RT5640_SCB_DIS (0x0 << 14) +#define RT5640_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5640_BB_MASK (0x1 << 15) +#define RT5640_BB_SFT 15 +#define RT5640_BB_DIS (0x0 << 15) +#define RT5640_BB_EN (0x1 << 15) +#define RT5640_BB_CT_MASK (0x7 << 12) +#define RT5640_BB_CT_SFT 12 +#define RT5640_BB_CT_A (0x0 << 12) +#define RT5640_BB_CT_B (0x1 << 12) +#define RT5640_BB_CT_C (0x2 << 12) +#define RT5640_BB_CT_D (0x3 << 12) +#define RT5640_M_BB_L_MASK (0x1 << 9) +#define RT5640_M_BB_L_SFT 9 +#define RT5640_M_BB_R_MASK (0x1 << 8) +#define RT5640_M_BB_R_SFT 8 +#define RT5640_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5640_M_BB_HPF_L_SFT 7 +#define RT5640_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5640_M_BB_HPF_R_SFT 6 +#define RT5640_G_BB_BST_MASK (0x3f) +#define RT5640_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5640_M_MP3_L_MASK (0x1 << 15) +#define RT5640_M_MP3_L_SFT 15 +#define RT5640_M_MP3_R_MASK (0x1 << 14) +#define RT5640_M_MP3_R_SFT 14 +#define RT5640_M_MP3_MASK (0x1 << 13) +#define RT5640_M_MP3_SFT 13 +#define RT5640_M_MP3_DIS (0x0 << 13) +#define RT5640_M_MP3_EN (0x1 << 13) +#define RT5640_EG_MP3_MASK (0x1f << 8) +#define RT5640_EG_MP3_SFT 8 +#define RT5640_MP3_HLP_MASK (0x1 << 7) +#define RT5640_MP3_HLP_SFT 7 +#define RT5640_MP3_HLP_DIS (0x0 << 7) +#define RT5640_MP3_HLP_EN (0x1 << 7) +#define RT5640_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5640_M_MP3_ORG_L_SFT 6 +#define RT5640_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5640_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5640_MP3_WT_MASK (0x1 << 13) +#define RT5640_MP3_WT_SFT 13 +#define RT5640_MP3_WT_1_4 (0x0 << 13) +#define RT5640_MP3_WT_1_2 (0x1 << 13) +#define RT5640_OG_MP3_MASK (0x1f << 8) +#define RT5640_OG_MP3_SFT 8 +#define RT5640_HG_MP3_MASK (0x3f) +#define RT5640_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5640_3D_CF_MASK (0x1 << 15) +#define RT5640_3D_CF_SFT 15 +#define RT5640_3D_CF_DIS (0x0 << 15) +#define RT5640_3D_CF_EN (0x1 << 15) +#define RT5640_3D_HP_MASK (0x1 << 14) +#define RT5640_3D_HP_SFT 14 +#define RT5640_3D_HP_DIS (0x0 << 14) +#define RT5640_3D_HP_EN (0x1 << 14) +#define RT5640_3D_BT_MASK (0x1 << 13) +#define RT5640_3D_BT_SFT 13 +#define RT5640_3D_BT_DIS (0x0 << 13) +#define RT5640_3D_BT_EN (0x1 << 13) +#define RT5640_3D_1F_MIX_MASK (0x3 << 11) +#define RT5640_3D_1F_MIX_SFT 11 +#define RT5640_3D_HP_M_MASK (0x1 << 10) +#define RT5640_3D_HP_M_SFT 10 +#define RT5640_3D_HP_M_SUR (0x0 << 10) +#define RT5640_3D_HP_M_FRO (0x1 << 10) +#define RT5640_M_3D_HRTF_MASK (0x1 << 9) +#define RT5640_M_3D_HRTF_SFT 9 +#define RT5640_M_3D_D2H_MASK (0x1 << 8) +#define RT5640_M_3D_D2H_SFT 8 +#define RT5640_M_3D_D2R_MASK (0x1 << 7) +#define RT5640_M_3D_D2R_SFT 7 +#define RT5640_M_3D_REVB_MASK (0x1 << 6) +#define RT5640_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5640_2ND_HPF_MASK (0x1 << 15) +#define RT5640_2ND_HPF_SFT 15 +#define RT5640_2ND_HPF_DIS (0x0 << 15) +#define RT5640_2ND_HPF_EN (0x1 << 15) +#define RT5640_HPF_CF_L_MASK (0x7 << 12) +#define RT5640_HPF_CF_L_SFT 12 +#define RT5640_1ST_HPF_MASK (0x1 << 11) +#define RT5640_1ST_HPF_SFT 11 +#define RT5640_1ST_HPF_DIS (0x0 << 11) +#define RT5640_1ST_HPF_EN (0x1 << 11) +#define RT5640_HPF_CF_R_MASK (0x7 << 8) +#define RT5640_HPF_CF_R_SFT 8 +#define RT5640_ZD_T_MASK (0x3 << 6) +#define RT5640_ZD_T_SFT 6 +#define RT5640_ZD_F_MASK (0x3 << 4) +#define RT5640_ZD_F_SFT 4 +#define RT5640_ZD_F_IM (0x0 << 4) +#define RT5640_ZD_F_ZC_IM (0x1 << 4) +#define RT5640_ZD_F_ZC_IOD (0x2 << 4) +#define RT5640_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5640_SI_DAC_MASK (0x1 << 11) +#define RT5640_SI_DAC_SFT 11 +#define RT5640_SI_DAC_AUTO (0x0 << 11) +#define RT5640_SI_DAC_TEST (0x1 << 11) +#define RT5640_DC_CAL_M_MASK (0x1 << 10) +#define RT5640_DC_CAL_M_SFT 10 +#define RT5640_DC_CAL_M_CAL (0x0 << 10) +#define RT5640_DC_CAL_M_NOR (0x1 << 10) +#define RT5640_DC_CAL_MASK (0x1 << 9) +#define RT5640_DC_CAL_SFT 9 +#define RT5640_DC_CAL_DIS (0x0 << 9) +#define RT5640_DC_CAL_EN (0x1 << 9) +#define RT5640_HPD_RCV_MASK (0x7 << 6) +#define RT5640_HPD_RCV_SFT 6 +#define RT5640_HPD_PS_MASK (0x1 << 5) +#define RT5640_HPD_PS_SFT 5 +#define RT5640_HPD_PS_DIS (0x0 << 5) +#define RT5640_HPD_PS_EN (0x1 << 5) +#define RT5640_CAL_M_MASK (0x1 << 4) +#define RT5640_CAL_M_SFT 4 +#define RT5640_CAL_M_DEP (0x0 << 4) +#define RT5640_CAL_M_CAL (0x1 << 4) +#define RT5640_CAL_MASK (0x1 << 3) +#define RT5640_CAL_SFT 3 +#define RT5640_CAL_DIS (0x0 << 3) +#define RT5640_CAL_EN (0x1 << 3) +#define RT5640_CAL_TEST_MASK (0x1 << 2) +#define RT5640_CAL_TEST_SFT 2 +#define RT5640_CAL_TEST_DIS (0x0 << 2) +#define RT5640_CAL_TEST_EN (0x1 << 2) +#define RT5640_CAL_P_MASK (0x3) +#define RT5640_CAL_P_SFT 0 +#define RT5640_CAL_P_NONE (0x0) +#define RT5640_CAL_P_CAL (0x1) +#define RT5640_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5640_SV_MASK (0x1 << 15) +#define RT5640_SV_SFT 15 +#define RT5640_SV_DIS (0x0 << 15) +#define RT5640_SV_EN (0x1 << 15) +#define RT5640_SPO_SV_MASK (0x1 << 14) +#define RT5640_SPO_SV_SFT 14 +#define RT5640_SPO_SV_DIS (0x0 << 14) +#define RT5640_SPO_SV_EN (0x1 << 14) +#define RT5640_OUT_SV_MASK (0x1 << 13) +#define RT5640_OUT_SV_SFT 13 +#define RT5640_OUT_SV_DIS (0x0 << 13) +#define RT5640_OUT_SV_EN (0x1 << 13) +#define RT5640_HP_SV_MASK (0x1 << 12) +#define RT5640_HP_SV_SFT 12 +#define RT5640_HP_SV_DIS (0x0 << 12) +#define RT5640_HP_SV_EN (0x1 << 12) +#define RT5640_ZCD_DIG_MASK (0x1 << 11) +#define RT5640_ZCD_DIG_SFT 11 +#define RT5640_ZCD_DIG_DIS (0x0 << 11) +#define RT5640_ZCD_DIG_EN (0x1 << 11) +#define RT5640_ZCD_MASK (0x1 << 10) +#define RT5640_ZCD_SFT 10 +#define RT5640_ZCD_PD (0x0 << 10) +#define RT5640_ZCD_PU (0x1 << 10) +#define RT5640_M_ZCD_MASK (0x3f << 4) +#define RT5640_M_ZCD_SFT 4 +#define RT5640_M_ZCD_RM_L (0x1 << 9) +#define RT5640_M_ZCD_RM_R (0x1 << 8) +#define RT5640_M_ZCD_SM_L (0x1 << 7) +#define RT5640_M_ZCD_SM_R (0x1 << 6) +#define RT5640_M_ZCD_OM_L (0x1 << 5) +#define RT5640_M_ZCD_OM_R (0x1 << 4) +#define RT5640_SV_DLY_MASK (0xf) +#define RT5640_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5640_ZCD_HP_MASK (0x1 << 15) +#define RT5640_ZCD_HP_SFT 15 +#define RT5640_ZCD_HP_DIS (0x0 << 15) +#define RT5640_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5640_3D_SPK_MASK (0x1 << 15) +#define RT5640_3D_SPK_SFT 15 +#define RT5640_3D_SPK_DIS (0x0 << 15) +#define RT5640_3D_SPK_EN (0x1 << 15) +#define RT5640_3D_SPK_M_MASK (0x3 << 13) +#define RT5640_3D_SPK_M_SFT 13 +#define RT5640_3D_SPK_CG_MASK (0x1f << 8) +#define RT5640_3D_SPK_CG_SFT 8 +#define RT5640_3D_SPK_SG_MASK (0x1f) +#define RT5640_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5640_WND_MASK (0x1 << 15) +#define RT5640_WND_SFT 15 +#define RT5640_WND_DIS (0x0 << 15) +#define RT5640_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5640_WND_FC_NW_MASK (0x3f << 10) +#define RT5640_WND_FC_NW_SFT 10 +#define RT5640_WND_FC_WK_MASK (0x3f << 4) +#define RT5640_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5640_HPF_FC_MASK (0x3f << 6) +#define RT5640_HPF_FC_SFT 6 +#define RT5640_WND_FC_ST_MASK (0x3f) +#define RT5640_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5640_WND_TH_LO_MASK (0x3ff) +#define RT5640_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5640_WND_TH_HI_MASK (0x3ff) +#define RT5640_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5640_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5640_WND_WIND_SFT 13 +#define RT5640_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5640_WND_STRONG_SFT 12 +enum { + RT5640_NO_WIND, + RT5640_BREEZE, + RT5640_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5640_DP_ATT_MASK (0x3 << 14) +#define RT5640_DP_ATT_SFT 14 +#define RT5640_DP_SPK_MASK (0x1 << 10) +#define RT5640_DP_SPK_SFT 10 +#define RT5640_DP_SPK_DIS (0x0 << 10) +#define RT5640_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5640_EQ_PRE_VOL_MASK (0xffff) +#define RT5640_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5640_EQ_PST_VOL_MASK (0xffff) +#define RT5640_EQ_PST_VOL_SFT 0 + +#define RT5640_NO_JACK BIT(0) +#define RT5640_HEADSET_DET BIT(1) +#define RT5640_HEADPHO_DET BIT(2) + +/* System Clock Source */ +#define RT5640_SCLK_S_MCLK 0 +#define RT5640_SCLK_S_PLL1 1 +#define RT5640_SCLK_S_PLL1_TK 2 +#define RT5640_SCLK_S_RCCLK 3 + +/* PLL1 Source */ +#define RT5640_PLL1_S_MCLK 0 +#define RT5640_PLL1_S_BCLK1 1 +#define RT5640_PLL1_S_BCLK2 2 +#define RT5640_PLL1_S_BCLK3 3 + + +enum { + RT5640_AIF1, + RT5640_AIF2, + RT5640_AIF3, + RT5640_AIFS, +}; + +enum { + RT5640_U_IF1 = 0x1, + RT5640_U_IF2 = 0x2, + RT5640_U_IF3 = 0x4, +}; + +enum { + RT5640_IF_123, + RT5640_IF_132, + RT5640_IF_312, + RT5640_IF_321, + RT5640_IF_231, + RT5640_IF_213, + RT5640_IF_113, + RT5640_IF_223, + RT5640_IF_ALL, +}; + +enum { + RT5640_DMIC_DIS, + RT5640_DMIC1, + RT5640_DMIC2, +}; + +struct rt5640_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5640_priv { + struct snd_soc_codec *codec; + struct rt5640_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5640_AIFS]; + int bclk[RT5640_AIFS]; + int master[RT5640_AIFS]; + + struct rt5640_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + int dmic_en; + bool hp_mute; +}; + +#endif diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..1f4093f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/clk.h> +#include <linux/regmap.h> #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> @@ -34,30 +35,30 @@ #define SGTL5000_MAX_REG_OFFSET 0x013A /* default value of sgtl5000 registers */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { - [SGTL5000_CHIP_CLK_CTRL] = 0x0008, - [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, - [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, - [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, - [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, - [SGTL5000_CHIP_ANA_CTRL] = 0x0111, - [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, - [SGTL5000_CHIP_ANA_POWER] = 0x7060, - [SGTL5000_CHIP_PLL_CTRL] = 0x5000, - [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, - [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, - [SGTL5000_DAP_SURROUND] = 0x0040, - [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, - [SGTL5000_DAP_MAIN_CHAN] = 0x8000, - [SGTL5000_DAP_AVC_CTRL] = 0x0510, - [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, - [SGTL5000_DAP_AVC_ATTACK] = 0x0028, - [SGTL5000_DAP_AVC_DECAY] = 0x0050, +static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, + { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, + { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, + { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, + { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, + { SGTL5000_CHIP_ANA_POWER, 0x7060 }, + { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, + { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_SURROUND, 0x0040 }, + { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, + { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, + { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, + { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -112,6 +113,8 @@ struct sgtl5000_priv { int fmt; /* i2s data format */ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; + struct regmap *regmap; + struct clk *mclk; }; /* @@ -150,16 +153,26 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + case SND_SOC_DAPM_PRE_PMD: + /* + * Don't clear VAG_POWERUP, when both DAC and ADC are + * operational to prevent inadvertently starving the + * other one of them. + */ + if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + mask) != mask) { + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + } break; default: break; @@ -217,12 +230,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +242,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ @@ -389,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", SGTL5000_CHIP_ANA_ADC_CTRL, - 8, 2, 0, capture_6db_attenuate), + 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), SOC_DOUBLE_TLV("Headphone Playback Volume", @@ -645,16 +654,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); } else { + /* otherwise, clk_ctrl must be set before pll power down */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + /* power down pll */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, 0); } - /* if using pll, clk_ctrl must be set after pll power up */ - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); - return 0; } @@ -909,10 +921,25 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); + + regcache_cache_only(sgtl5000->regmap, false); + + ret = regcache_sync(sgtl5000->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + + regcache_cache_only(sgtl5000->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + return ret; + } } break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); break; @@ -958,17 +985,76 @@ static struct snd_soc_dai_driver sgtl5000_dai = { .symmetric_rates = 1, }; -static int sgtl5000_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool sgtl5000_volatile(struct device *dev, unsigned int reg) { switch (reg) { case SGTL5000_CHIP_ID: case SGTL5000_CHIP_ADCDAC_CTRL: case SGTL5000_CHIP_ANA_STATUS: - return 1; + return true; } - return 0; + return false; +} + +static bool sgtl5000_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_DIG_POWER: + case SGTL5000_CHIP_CLK_CTRL: + case SGTL5000_CHIP_I2S_CTRL: + case SGTL5000_CHIP_SSS_CTRL: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_DAC_VOL: + case SGTL5000_CHIP_PAD_STRENGTH: + case SGTL5000_CHIP_ANA_ADC_CTRL: + case SGTL5000_CHIP_ANA_HP_CTRL: + case SGTL5000_CHIP_ANA_CTRL: + case SGTL5000_CHIP_LINREG_CTRL: + case SGTL5000_CHIP_REF_CTRL: + case SGTL5000_CHIP_MIC_CTRL: + case SGTL5000_CHIP_LINE_OUT_CTRL: + case SGTL5000_CHIP_LINE_OUT_VOL: + case SGTL5000_CHIP_ANA_POWER: + case SGTL5000_CHIP_PLL_CTRL: + case SGTL5000_CHIP_CLK_TOP_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + case SGTL5000_CHIP_SHORT_CTRL: + case SGTL5000_CHIP_ANA_TEST2: + case SGTL5000_DAP_CTRL: + case SGTL5000_DAP_PEQ: + case SGTL5000_DAP_BASS_ENHANCE: + case SGTL5000_DAP_BASS_ENHANCE_CTRL: + case SGTL5000_DAP_AUDIO_EQ: + case SGTL5000_DAP_SURROUND: + case SGTL5000_DAP_FLT_COEF_ACCESS: + case SGTL5000_DAP_COEF_WR_B0_MSB: + case SGTL5000_DAP_COEF_WR_B0_LSB: + case SGTL5000_DAP_EQ_BASS_BAND0: + case SGTL5000_DAP_EQ_BASS_BAND1: + case SGTL5000_DAP_EQ_BASS_BAND2: + case SGTL5000_DAP_EQ_BASS_BAND3: + case SGTL5000_DAP_EQ_BASS_BAND4: + case SGTL5000_DAP_MAIN_CHAN: + case SGTL5000_DAP_MIX_CHAN: + case SGTL5000_DAP_AVC_CTRL: + case SGTL5000_DAP_AVC_THRESHOLD: + case SGTL5000_DAP_AVC_ATTACK: + case SGTL5000_DAP_AVC_DECAY: + case SGTL5000_DAP_COEF_WR_B1_MSB: + case SGTL5000_DAP_COEF_WR_B1_LSB: + case SGTL5000_DAP_COEF_WR_B2_MSB: + case SGTL5000_DAP_COEF_WR_B2_LSB: + case SGTL5000_DAP_COEF_WR_A1_MSB: + case SGTL5000_DAP_COEF_WR_A1_LSB: + case SGTL5000_DAP_COEF_WR_A2_MSB: + case SGTL5000_DAP_COEF_WR_A2_LSB: + return true; + + default: + return false; + } } #ifdef CONFIG_SUSPEND @@ -1214,7 +1300,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - u16 reg; + int reg; int ret; int rev; int i; @@ -1242,23 +1328,17 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* read chip information */ - reg = snd_soc_read(codec, SGTL5000_CHIP_ID); - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(codec->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - ret = -ENODEV; - goto err_regulator_disable; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); - /* * workaround for revision 0x11 and later, * roll back to use internal LDO */ + + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); + if (ret) + goto err_regulator_disable; + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1300,7 +1380,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* setup i2c data ops */ - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + codec->control_data = sgtl5000->regmap; + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1391,11 +1472,6 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .suspend = sgtl5000_suspend, .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = sgtl5000_regs, - .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, @@ -1404,28 +1480,118 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), }; +static const struct regmap_config sgtl5000_regmap = { + .reg_bits = 16, + .val_bits = 16, + .reg_stride = 2, + + .max_register = SGTL5000_MAX_REG_OFFSET, + .volatile_reg = sgtl5000_volatile, + .readable_reg = sgtl5000_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sgtl5000_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), +}; + +/* + * Write all the default values from sgtl5000_reg_defaults[] array into the + * sgtl5000 registers, to make sure we always start with the sane registers + * values as stated in the datasheet. + * + * Since sgtl5000 does not have a reset line, nor a reset command in software, + * we follow this approach to guarantee we always start from the default values + * and avoid problems like, not being able to probe after an audio playback + * followed by a system reset or a 'reboot' command in Linux + */ +static int sgtl5000_fill_defaults(struct sgtl5000_priv *sgtl5000) +{ + int i, ret, val, index; + + for (i = 0; i < ARRAY_SIZE(sgtl5000_reg_defaults); i++) { + val = sgtl5000_reg_defaults[i].def; + index = sgtl5000_reg_defaults[i].reg; + ret = regmap_write(sgtl5000->regmap, index, val); + if (ret) + return ret; + } + + return 0; +} + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret; + int ret, reg, rev; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); if (!sgtl5000) return -ENOMEM; + sgtl5000->regmap = devm_regmap_init_i2c(client, &sgtl5000_regmap); + if (IS_ERR(sgtl5000->regmap)) { + ret = PTR_ERR(sgtl5000->regmap); + dev_err(&client->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + + sgtl5000->mclk = devm_clk_get(&client->dev, NULL); + if (IS_ERR(sgtl5000->mclk)) { + ret = PTR_ERR(sgtl5000->mclk); + dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + /* Defer the probe to see if the clk will be provided later */ + if (ret == -ENOENT) + return -EPROBE_DEFER; + return ret; + } + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + return ret; + + /* read chip information */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); + if (ret) + goto disable_clk; + + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(&client->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto disable_clk; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + i2c_set_clientdata(client, sgtl5000); + /* Ensure sgtl5000 will start with sane register values */ + ret = sgtl5000_fill_defaults(sgtl5000); + if (ret) + goto disable_clk; + ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto disable_clk; + + return 0; + +disable_clk: + clk_disable_unprepare(sgtl5000->mclk); return ret; } static int sgtl5000_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + clk_disable_unprepare(sgtl5000->mclk); return 0; } diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 8a9f435..2f8c889 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -12,7 +12,7 @@ #define _SGTL5000_H /* - * Register values. + * Registers addresses */ #define SGTL5000_CHIP_ID 0x0000 #define SGTL5000_CHIP_DIG_POWER 0x0002 @@ -347,7 +347,7 @@ #define SGTL5000_PLL_INT_DIV_MASK 0xf800 #define SGTL5000_PLL_INT_DIV_SHIFT 11 #define SGTL5000_PLL_INT_DIV_WIDTH 5 -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff #define SGTL5000_PLL_FRAC_DIV_SHIFT 0 #define SGTL5000_PLL_FRAC_DIV_WIDTH 11 diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 721587c..38f3b10 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -38,9 +38,9 @@ enum si476x_digital_io_output_format { SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT = 8, }; -#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0b111 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \ - (0b111 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)) -#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0b1111110) +#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0x7 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \ + (0x7 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)) +#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0x7e) enum si476x_daudio_formats { SI476X_DAUDIO_MODE_I2S = (0x0 << 1), @@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec, return err; } +static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +}; + +static const struct snd_soc_dapm_route si476x_dapm_routes[] = { + { "Capture", NULL, "LOUT" }, + { "Capture", NULL, "ROUT" }, +}; + static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, .read = si476x_codec_read, .write = si476x_codec_write, + .dapm_widgets = si476x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), + .dapm_routes = si476x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes), }; static int si476x_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index d1ae869d..dba26e63 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -883,7 +883,7 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sn95031_codec = { +static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, .read = sn95031_read, diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index dd8d856..e3501f4 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -21,12 +21,28 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> + +static const struct snd_soc_dapm_widget dir_widgets[] = { + SND_SOC_DAPM_INPUT("spdif-in"), +}; + +static const struct snd_soc_dapm_route dir_routes[] = { + { "Capture", NULL, "spdif-in" }, +}; #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dir; +static struct snd_soc_codec_driver soc_codec_spdif_dir = { + .dapm_widgets = dir_widgets, + .num_dapm_widgets = ARRAY_SIZE(dir_widgets), + .dapm_routes = dir_routes, + .num_dapm_routes = ARRAY_SIZE(dir_routes), +}; static struct snd_soc_dai_driver dir_stub_dai = { .name = "dir-hifi", @@ -51,12 +67,21 @@ static int spdif_dir_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dir_dt_ids[] = { + { .compatible = "linux,spdif-dir", }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dir_dt_ids); +#endif + static struct platform_driver spdif_dir_driver = { .probe = spdif_dir_probe, .remove = spdif_dir_remove, .driver = { .name = "spdif-dir", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dir_dt_ids), }, }; diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transmitter.c index 112a49d..a078aa3 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -20,14 +20,29 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> #define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 -#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) +static const struct snd_soc_dapm_widget dit_widgets[] = { + SND_SOC_DAPM_OUTPUT("spdif-out"), +}; + +static const struct snd_soc_dapm_route dit_routes[] = { + { "spdif-out", NULL, "Playback" }, +}; -static struct snd_soc_codec_driver soc_codec_spdif_dit; +static struct snd_soc_codec_driver soc_codec_spdif_dit = { + .dapm_widgets = dit_widgets, + .num_dapm_widgets = ARRAY_SIZE(dit_widgets), + .dapm_routes = dit_routes, + .num_dapm_routes = ARRAY_SIZE(dit_routes), +}; static struct snd_soc_dai_driver dit_stub_dai = { .name = "dit-hifi", @@ -52,12 +67,21 @@ static int spdif_dit_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = "linux,spdif-dit", }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dit_dt_ids); +#endif + static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 0000000..95aed55 --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,856 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_data/ssm2518.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + + return -EINVAL; +} + +static int ssm2518_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + + if (tx_mask == 0 || rx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 0: + ssm2518->constraints = NULL; + break; + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio < 0 && ssm2518->enable_gpio != -ENOENT) + return ssm2518->enable_gpio; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + regcache_cache_bypass(ssm2518->regmap, false); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + ret = ssm2518_set_power(ssm2518, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 0000000..62511d8 --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f8d30e5..492644e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec) static int ssm2602_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); + regcache_sync(ssm2602->regmap); ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index cfb55fe..06edb39 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -363,16 +363,18 @@ static void sta32x_watchdog(struct work_struct *work) } if (!sta32x->shutdown) - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } static void sta32x_watchdog_start(struct sta32x_priv *sta32x) { if (sta32x->pdata->needs_esd_watchdog) { sta32x->shutdown = 0; - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } } diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2eda85ba..a5455c1 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -28,8 +28,6 @@ #include "stac9766.h" -#define STAC9766_VERSION "0.10" - /* * STAC9766 register cache */ @@ -145,14 +143,14 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } @@ -164,7 +162,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } @@ -175,7 +173,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { - val = soc_ac97_ops.read(codec->ac97, reg); + val = soc_ac97_ops->read(codec->ac97, reg); return val; } return cache[reg / 2]; @@ -242,15 +240,15 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; @@ -274,7 +272,7 @@ reset: return -EIO; } codec->ac97->bus->ops->warm_reset(codec->ac97); - id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { stac9766_reset(codec, 0); reset++; @@ -338,9 +336,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); - - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d447c4a..6d31d88 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -83,6 +83,14 @@ #define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ #define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ #define TAS5086_BKNDERR 0x1c +#define TAS5086_INPUT_MUX 0x20 +#define TAS5086_PWM_OUTPUT_MUX 0x25 + +#define TAS5086_MAX_REGISTER TAS5086_PWM_OUTPUT_MUX + +#define TAS5086_PWM_START_MIDZ_FOR_START_1 (1 << 7) +#define TAS5086_PWM_START_MIDZ_FOR_START_2 (1 << 6) +#define TAS5086_PWM_START_CHANNEL_MASK (0x3f) /* * Default TAS5086 power-up configuration @@ -119,9 +127,30 @@ static const struct reg_default tas5086_reg_defaults[] = { { 0x1c, 0x05 }, }; +static int tas5086_register_size(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5086_CLOCK_CONTROL ... TAS5086_BKNDERR: + return 1; + case TAS5086_INPUT_MUX: + case TAS5086_PWM_OUTPUT_MUX: + return 4; + } + + dev_err(dev, "Unsupported register address: %d\n", reg); + return 0; +} + static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) { - return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); + switch (reg) { + case 0x0f: + case 0x11 ... 0x17: + case 0x1d ... 0x1f: + return false; + default: + return true; + } } static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) @@ -140,6 +169,76 @@ static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); } +static int tas5086_reg_write(void *context, unsigned int reg, + unsigned int value) +{ + struct i2c_client *client = context; + unsigned int i, size; + uint8_t buf[5]; + int ret; + + size = tas5086_register_size(&client->dev, reg); + if (size == 0) + return -EINVAL; + + buf[0] = reg; + + for (i = size; i >= 1; --i) { + buf[i] = value; + value >>= 8; + } + + ret = i2c_master_send(client, buf, size + 1); + if (ret == size + 1) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static int tas5086_reg_read(void *context, unsigned int reg, + unsigned int *value) +{ + struct i2c_client *client = context; + uint8_t send_buf, recv_buf[4]; + struct i2c_msg msgs[2]; + unsigned int size; + unsigned int i; + int ret; + + size = tas5086_register_size(&client->dev, reg); + if (size == 0) + return -EINVAL; + + send_buf = reg; + + msgs[0].addr = client->addr; + msgs[0].len = sizeof(send_buf); + msgs[0].buf = &send_buf; + msgs[0].flags = 0; + + msgs[1].addr = client->addr; + msgs[1].len = size; + msgs[1].buf = recv_buf; + msgs[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); + if (ret < 0) + return ret; + else if (ret != ARRAY_SIZE(msgs)) + return -EIO; + + *value = 0; + + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } + + return 0; +} + struct tas5086_private { struct regmap *regmap; unsigned int mclk, sclk; @@ -376,6 +475,202 @@ static const struct snd_kcontrol_new tas5086_controls[] = { tas5086_get_deemph, tas5086_put_deemph), }; +/* Input mux controls */ +static const char *tas5086_dapm_sdin_texts[] = +{ + "SDIN1-L", "SDIN1-R", "SDIN2-L", "SDIN2-R", + "SDIN3-L", "SDIN3-R", "Ground (0)", "nc" +}; + +static const struct soc_enum tas5086_dapm_input_mux_enum[] = { + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 20, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 16, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 12, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 8, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 4, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 0, 8, tas5086_dapm_sdin_texts), +}; + +static const struct snd_kcontrol_new tas5086_dapm_input_mux_controls[] = { + SOC_DAPM_ENUM("Channel 1 input", tas5086_dapm_input_mux_enum[0]), + SOC_DAPM_ENUM("Channel 2 input", tas5086_dapm_input_mux_enum[1]), + SOC_DAPM_ENUM("Channel 3 input", tas5086_dapm_input_mux_enum[2]), + SOC_DAPM_ENUM("Channel 4 input", tas5086_dapm_input_mux_enum[3]), + SOC_DAPM_ENUM("Channel 5 input", tas5086_dapm_input_mux_enum[4]), + SOC_DAPM_ENUM("Channel 6 input", tas5086_dapm_input_mux_enum[5]), +}; + +/* Output mux controls */ +static const char *tas5086_dapm_channel_texts[] = + { "Channel 1 Mux", "Channel 2 Mux", "Channel 3 Mux", + "Channel 4 Mux", "Channel 5 Mux", "Channel 6 Mux" }; + +static const struct soc_enum tas5086_dapm_output_mux_enum[] = { + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 20, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 16, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 12, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 8, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 4, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 0, 6, tas5086_dapm_channel_texts), +}; + +static const struct snd_kcontrol_new tas5086_dapm_output_mux_controls[] = { + SOC_DAPM_ENUM("PWM1 Output", tas5086_dapm_output_mux_enum[0]), + SOC_DAPM_ENUM("PWM2 Output", tas5086_dapm_output_mux_enum[1]), + SOC_DAPM_ENUM("PWM3 Output", tas5086_dapm_output_mux_enum[2]), + SOC_DAPM_ENUM("PWM4 Output", tas5086_dapm_output_mux_enum[3]), + SOC_DAPM_ENUM("PWM5 Output", tas5086_dapm_output_mux_enum[4]), + SOC_DAPM_ENUM("PWM6 Output", tas5086_dapm_output_mux_enum[5]), +}; + +static const struct snd_soc_dapm_widget tas5086_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("SDIN1-L"), + SND_SOC_DAPM_INPUT("SDIN1-R"), + SND_SOC_DAPM_INPUT("SDIN2-L"), + SND_SOC_DAPM_INPUT("SDIN2-R"), + SND_SOC_DAPM_INPUT("SDIN3-L"), + SND_SOC_DAPM_INPUT("SDIN3-R"), + SND_SOC_DAPM_INPUT("SDIN4-L"), + SND_SOC_DAPM_INPUT("SDIN4-R"), + + SND_SOC_DAPM_OUTPUT("PWM1"), + SND_SOC_DAPM_OUTPUT("PWM2"), + SND_SOC_DAPM_OUTPUT("PWM3"), + SND_SOC_DAPM_OUTPUT("PWM4"), + SND_SOC_DAPM_OUTPUT("PWM5"), + SND_SOC_DAPM_OUTPUT("PWM6"), + + SND_SOC_DAPM_MUX("Channel 1 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[0]), + SND_SOC_DAPM_MUX("Channel 2 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[1]), + SND_SOC_DAPM_MUX("Channel 3 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[2]), + SND_SOC_DAPM_MUX("Channel 4 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[3]), + SND_SOC_DAPM_MUX("Channel 5 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[4]), + SND_SOC_DAPM_MUX("Channel 6 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[5]), + + SND_SOC_DAPM_MUX("PWM1 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[0]), + SND_SOC_DAPM_MUX("PWM2 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[1]), + SND_SOC_DAPM_MUX("PWM3 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[2]), + SND_SOC_DAPM_MUX("PWM4 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[3]), + SND_SOC_DAPM_MUX("PWM5 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[4]), + SND_SOC_DAPM_MUX("PWM6 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[5]), +}; + +static const struct snd_soc_dapm_route tas5086_dapm_routes[] = { + /* SDIN inputs -> channel muxes */ + { "Channel 1 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 1 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 1 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 1 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 1 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 1 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 2 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 2 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 2 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 2 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 2 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 2 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 2 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 2 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 2 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 2 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 2 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 2 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 3 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 3 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 3 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 3 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 3 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 3 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 4 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 4 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 4 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 4 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 4 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 4 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 5 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 5 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 5 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 5 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 5 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 5 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 6 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 6 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 6 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 6 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 6 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 6 Mux", "SDIN3-R", "SDIN3-R" }, + + /* Channel muxes -> PWM muxes */ + { "PWM1 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM2 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM3 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM4 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM5 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM6 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + + { "PWM1 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM2 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM3 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM4 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM5 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM6 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + + { "PWM1 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM2 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM3 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM4 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM5 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM6 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + + { "PWM1 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM2 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM3 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM4 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM5 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM6 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + + { "PWM1 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM2 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM3 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM4 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM5 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM6 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + + { "PWM1 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM2 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM3 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM4 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM5 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM6 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + + /* The PWM muxes are directly connected to the PWM outputs */ + { "PWM1", NULL, "PWM1 Mux" }, + { "PWM2", NULL, "PWM2 Mux" }, + { "PWM3", NULL, "PWM3 Mux" }, + { "PWM4", NULL, "PWM4 Mux" }, + { "PWM5", NULL, "PWM5 Mux" }, + { "PWM6", NULL, "PWM6 Mux" }, + +}; + static const struct snd_soc_dai_ops tas5086_dai_ops = { .hw_params = tas5086_hw_params, .set_sysclk = tas5086_set_dai_sysclk, @@ -426,13 +721,34 @@ static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int charge_period = 1300000; /* hardware default is 1300 ms */ + u8 pwm_start_mid_z = 0; int i, ret; if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; of_property_read_u32(of_node, "ti,charge-period", &charge_period); + + for (i = 0; i < 6; i++) { + char name[25]; + + snprintf(name, sizeof(name), + "ti,mid-z-channel-%d", i + 1); + + if (of_get_property(of_node, name, NULL) != NULL) + pwm_start_mid_z |= 1 << i; + } } + /* + * If any of the channels is configured to start in Mid-Z mode, + * configure 'part 1' of the PWM starts to use Mid-Z, and tell + * all configured mid-z channels to start start under 'part 1'. + */ + if (pwm_start_mid_z) + regmap_write(priv->regmap, TAS5086_PWM_START, + TAS5086_PWM_START_MIDZ_FOR_START_1 | + pwm_start_mid_z); + /* lookup and set split-capacitor charge period */ if (charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); @@ -490,6 +806,10 @@ static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { .resume = tas5086_soc_resume, .controls = tas5086_controls, .num_controls = ARRAY_SIZE(tas5086_controls), + .dapm_widgets = tas5086_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5086_dapm_widgets), + .dapm_routes = tas5086_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tas5086_dapm_routes), }; static const struct i2c_device_id tas5086_i2c_id[] = { @@ -500,14 +820,16 @@ MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); static const struct regmap_config tas5086_regmap = { .reg_bits = 8, - .val_bits = 8, - .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .val_bits = 32, + .max_register = TAS5086_MAX_REGISTER, .reg_defaults = tas5086_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), .cache_type = REGCACHE_RBTREE, .volatile_reg = tas5086_volatile_reg, .writeable_reg = tas5086_writeable_reg, .readable_reg = tas5086_accessible_reg, + .reg_read = tas5086_reg_read, + .reg_write = tas5086_reg_write, }; static int tas5086_i2c_probe(struct i2c_client *i2c, @@ -522,7 +844,7 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; - priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b1f6982..7b8f3d9 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,7 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; - struct snd_soc_codec codec; + struct snd_soc_codec *codec; int master; int datfm; int mclk; @@ -119,6 +119,22 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } +static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_route tlv320aic26_dapm_routes[] = { + { "Capture", NULL, "MICIN" }, + { "Capture", NULL, "AUX" }, + + { "HPL", NULL, "Playback" }, + { "HPR", NULL, "Playback" }, +}; + /* --------------------------------------------------------------------- * Digital Audio Interface Operations */ @@ -174,9 +190,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); @@ -185,13 +201,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg |= 0x0800; if (fsref == 48000) reg |= 0x2000; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Audio Control 1 (FSref divisor) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); reg &= ~0x0fff; reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); return 0; } @@ -212,7 +228,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg |= 0x8080; else reg &= ~0x8080; - aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); return 0; } @@ -330,7 +346,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -346,9 +362,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -360,25 +376,26 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); */ static int aic26_probe(struct snd_soc_codec *codec) { + struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, err, i, reg; - dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); + aic26->codec = codec; /* Reset the codec to power on defaults */ - aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Fill register cache */ for (i = 0; i < codec->driver->reg_cache_size; i++) - aic26_reg_read(codec, i); + snd_soc_read(codec, i); /* Register the sysfs files for debugging */ /* Create SysFS files */ @@ -401,6 +418,10 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .dapm_widgets = tlv320aic26_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), + .dapm_routes = tlv320aic26_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), }; /* --------------------------------------------------------------------- diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 17df4e3..2ed57d4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate) return -EINVAL; } -static int aic32x4_add_widgets(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets, - ARRAY_SIZE(aic32x4_dapm_widgets)); - - snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes, - ARRAY_SIZE(aic32x4_dapm_routes)); - - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; -} - static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, aic32x4_snd_controls, - ARRAY_SIZE(aic32x4_snd_controls)); - aic32x4_add_widgets(codec); /* * Workaround: for an unknown reason, the ADC needs to be powered up @@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .suspend = aic32x4_suspend, .resume = aic32x4_resume, .set_bias_level = aic32x4_set_bias_level, + + .controls = aic32x4_snd_controls, + .num_controls = ARRAY_SIZE(aic32x4_snd_controls), + .dapm_widgets = aic32x4_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), + .dapm_routes = aic32x4_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; static int aic32x4_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 1514bf8..6e3f269 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -128,10 +128,8 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } + SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \ + snd_soc_dapm_get_volsw, snd_soc_dapm_put_volsw_aic3x) /* * All input lines are connected when !0xf and disconnected with 0xf bit field, @@ -140,8 +138,7 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -149,10 +146,9 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val_mask; - int ret; - struct snd_soc_dapm_path *path; - int found = 0; + unsigned short val; + struct snd_soc_dapm_update update; + int connect, change; val = (ucontrol->value.integer.value[0] & mask); @@ -160,42 +156,26 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, if (val) val = mask; + connect = !!val; + if (invert) val = mask - val; - val_mask = mask << shift; - val = val << shift; - - mutex_lock(&widget->codec->mutex); - if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { - if (path->kcontrol != kcontrol) - continue; + mask <<= shift; + val <<= shift; - /* found, now check type */ - found = 1; - if (val) - /* new connection */ - path->connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - path->connect = invert ? 1 : 0; + change = snd_soc_test_bits(codec, val, mask, reg); + if (change) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - dapm_mark_dirty(path->source, "tlv320aic3x source"); - dapm_mark_dirty(path->sink, "tlv320aic3x sink"); - - break; - } + snd_soc_dapm_mixer_update_power(&codec->dapm, kcontrol, connect, + &update); } - mutex_unlock(&widget->codec->mutex); - - if (found) - snd_soc_dapm_sync(widget->dapm); - - ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); - return ret; + return change; } /* @@ -1494,6 +1474,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", AIC3X_MODEL_3X }, { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, + { "tlv320aic3106", AIC3X_MODEL_3X }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1584,6 +1565,9 @@ static int aic3x_i2c_remove(struct i2c_client *client) #if defined(CONFIG_OF) static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, + { .compatible = "ti,tlv320aic33" }, + { .compatible = "ti,tlv320aic3007" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8e6e5b0..1e3884d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -137,8 +137,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { - struct snd_soc_codec codec; - unsigned int codec_powered; /* reference counts of AIF/APLL users */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9b9a6e5..3c79dbb 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -38,6 +38,14 @@ #include "twl6040.h" +enum twl6040_dai_id { + TWL6040_DAI_LEGACY = 0, + TWL6040_DAI_UL, + TWL6040_DAI_DL1, + TWL6040_DAI_DL2, + TWL6040_DAI_VIB, +}; + #define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) @@ -67,6 +75,8 @@ struct twl6040_data { int pll_power_mode; int hs_power_mode; int hs_power_mode_locked; + bool dl1_unmuted; + bool dl2_unmuted; unsigned int clk_in; unsigned int sysclk; struct twl6040_jack_data hs_jack; @@ -220,6 +230,25 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, return value; } +static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + + switch (reg) { + case TWL6040_REG_HSLCTL: + case TWL6040_REG_HSRCTL: + case TWL6040_REG_EARCTL: + /* DL1 path */ + return priv->dl1_unmuted; + case TWL6040_REG_HFLCTL: + case TWL6040_REG_HFRCTL: + return priv->dl2_unmuted; + default: + return 1; + }; +} + /* * write to the twl6040 register space */ @@ -232,7 +261,8 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (likely(reg < TWL6040_REG_SW_SHADOW)) + if (likely(reg < TWL6040_REG_SW_SHADOW) && + twl6040_is_path_unmuted(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; @@ -399,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -407,9 +438,7 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; @@ -1026,16 +1055,84 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } +static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id id, + int mute) +{ + struct twl6040 *twl6040 = codec->control_data; + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int hslctl, hsrctl, earctl; + int hflctl, hfrctl; + + switch (id) { + case TWL6040_DAI_DL1: + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + earctl = twl6040_read_reg_cache(codec, TWL6040_REG_EARCTL); + + if (mute) { + /* Power down drivers and DACs */ + earctl &= ~0x01; + hslctl &= ~(TWL6040_HSDRVENA | TWL6040_HSDACENA); + hsrctl &= ~(TWL6040_HSDRVENA | TWL6040_HSDACENA); + + } + + twl6040_reg_write(twl6040, TWL6040_REG_EARCTL, earctl); + twl6040_reg_write(twl6040, TWL6040_REG_HSLCTL, hslctl); + twl6040_reg_write(twl6040, TWL6040_REG_HSRCTL, hsrctl); + priv->dl1_unmuted = !mute; + break; + case TWL6040_DAI_DL2: + hflctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFLCTL); + hfrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFRCTL); + + if (mute) { + /* Power down drivers and DACs */ + hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | + TWL6040_HFDRVENA); + hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | + TWL6040_HFDRVENA); + } + + twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); + twl6040_reg_write(twl6040, TWL6040_REG_HFRCTL, hfrctl); + priv->dl2_unmuted = !mute; + break; + default: + break; + }; +} + +static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) +{ + switch (dai->id) { + case TWL6040_DAI_LEGACY: + twl6040_mute_path(dai->codec, TWL6040_DAI_DL1, mute); + twl6040_mute_path(dai->codec, TWL6040_DAI_DL2, mute); + break; + case TWL6040_DAI_DL1: + case TWL6040_DAI_DL2: + twl6040_mute_path(dai->codec, dai->id, mute); + break; + default: + break; + } + + return 0; +} + static const struct snd_soc_dai_ops twl6040_dai_ops = { .startup = twl6040_startup, .hw_params = twl6040_hw_params, .prepare = twl6040_prepare, .set_sysclk = twl6040_set_dai_sysclk, + .digital_mute = twl6040_digital_mute, }; static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-legacy", + .id = TWL6040_DAI_LEGACY, .playback = { .stream_name = "Legacy Playback", .channels_min = 1, @@ -1054,6 +1151,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-ul", + .id = TWL6040_DAI_UL, .capture = { .stream_name = "Capture", .channels_min = 1, @@ -1065,6 +1163,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-dl1", + .id = TWL6040_DAI_DL1, .playback = { .stream_name = "Headset Playback", .channels_min = 1, @@ -1076,6 +1175,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-dl2", + .id = TWL6040_DAI_DL2, .playback = { .stream_name = "Handsfree Playback", .channels_min = 1, @@ -1087,6 +1187,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-vib", + .id = TWL6040_DAI_VIB, .playback = { .stream_name = "Vibra Playback", .channels_min = 1, @@ -1143,7 +1244,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) mutex_init(&priv->mutex); - ret = devm_request_threaded_irq(codec->dev, priv->plug_irq, NULL, + ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, IRQF_NO_SUSPEND, "twl6040_irq_plug", codec); if (ret) { @@ -1159,6 +1260,9 @@ static int twl6040_probe(struct snd_soc_codec *codec) static int twl6040_remove(struct snd_soc_codec *codec) { + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + + free_irq(priv->plug_irq, codec); twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44..c94d4c1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; struct uda134x_platform_data *pd = codec->control_data; int i; u8 *cache = codec->reg_cache; @@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* ADC, DAC on */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg | 0x03); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - /* ADC, DAC power off */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +/* UDA1341 has the DAC/ADC power down in STATUS1 */ +static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), +}; + +/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ +static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->card->dev->platform_data; + const struct snd_soc_dapm_widget *widgets; + unsigned num_widgets; int ret; @@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) else uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pd->model == UDA134X_UDA1341) { + widgets = uda1341_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); + } else { + widgets = uda1340_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); + } + + ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + if (ret) { + printk(KERN_ERR "%s failed to register dapm controls: %d", + __func__, ret); + kfree(uda134x); + return ret; + } + switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: @@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .read = uda134x_read_reg_cache, .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .dapm_widgets = uda134x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), + .dapm_routes = uda134x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), }; static int uda134x_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 54cd3da..b7ab2ef 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -290,6 +290,18 @@ static const struct snd_kcontrol_new wl1273_controls[] = { snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), }; +static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route wl1273_dapm_routes[] = { + { "Capture", NULL, "RX" }, + + { "TX", NULL, "Playback" }, +}; + static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -483,6 +495,11 @@ static int wl1273_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, + + .dapm_widgets = wl1273_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets), + .dapm_routes = wl1273_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes), }; static int wl1273_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 370af0c..d5ebcb0 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/interrupt.h> #include <linux/irqreturn.h> #include <linux/init.h> #include <linux/spi/spi.h> @@ -409,39 +410,39 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; - out = kzalloc(len, GFP_KERNEL); + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; - img = kzalloc(len, GFP_KERNEL); + img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - - xfer->codec = codec; - list_add_tail(&xfer->list, &xfer_list); - spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; @@ -522,14 +523,14 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL); + img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } - out = kzalloc(fw->size, GFP_KERNEL); + out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; @@ -669,14 +670,14 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = -ENOMEM; len = pll_rec.length + 8; - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); goto abort; } - img_swap = kzalloc(len, GFP_KERNEL); + img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); @@ -972,6 +973,13 @@ static int wm0010_spi_probe(struct spi_device *spi) } wm0010->irq = irq; + ret = irq_set_irq_wake(irq, 1); + if (ret) { + dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", + irq, ret); + return ret; + } + if (spi->max_speed_hz) wm0010->board_max_spi_speed = spi->max_speed_hz; else @@ -995,6 +1003,8 @@ static int wm0010_spi_remove(struct spi_device *spi) gpio_set_value_cansleep(wm0010->gpio_reset, wm0010->gpio_reset_value); + irq_set_irq_wake(wm0010->irq, 0); + if (wm0010->irq) free_irq(wm0010->irq, wm0010); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 100fdad..8bbddc1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -814,7 +814,20 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), -SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), +SOC_VALUE_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), + +SOC_DOUBLE("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0), +SOC_DOUBLE("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0), +SOC_SINGLE("EPOUT DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE3L_ENA_SHIFT, 1, 0), + +SOC_SINGLE("DRE Threshold", ARIZONA_DRE_CONTROL_2, + ARIZONA_DRE_T_LOW_SHIFT, 63, 0), + +SOC_SINGLE("DRE Low Level ABS", ARIZONA_DRE_CONTROL_3, + ARIZONA_DRE_LOW_LEVEL_ABS_SHIFT, 15, 0), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), @@ -852,6 +865,15 @@ ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -898,6 +920,15 @@ ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); @@ -967,6 +998,8 @@ SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), + SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | @@ -1117,6 +1150,56 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, @@ -1189,6 +1272,15 @@ ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"), ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), @@ -1249,6 +1341,14 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF2RX2", "AIF2RX2" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ { name, "EQ1", "EQ1" }, \ { name, "EQ2", "EQ2" }, \ { name, "EQ3", "EQ3" }, \ @@ -1304,17 +1404,25 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "OUT5L", NULL, "SYSCLK" }, { "OUT5R", NULL, "SYSCLK" }, + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, { "MICBIAS3", NULL, "MICVDD" }, + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + { "Noise Generator", NULL, "NOISE" }, { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -1345,13 +1453,41 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF3RX1", NULL, "AIF3 Playback" }, { "AIF3RX2", NULL, "AIF3 Playback" }, + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + { "AIF1 Playback", NULL, "SYSCLK" }, { "AIF2 Playback", NULL, "SYSCLK" }, { "AIF3 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, { "AIF1 Capture", NULL, "SYSCLK" }, { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, { "IN1L PGA", NULL, "IN1L" }, { "IN1R PGA", NULL, "IN1R" }, @@ -1362,23 +1498,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, - { "ASRC1L", NULL, "ASRC1L Input" }, - { "ASRC1R", NULL, "ASRC1R Input" }, - { "ASRC2L", NULL, "ASRC2L Input" }, - { "ASRC2R", NULL, "ASRC2R Input" }, - - { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, - { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, - - { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, - { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, - - { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, - { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, - - { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, - { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, - ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1408,6 +1527,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), @@ -1421,22 +1549,25 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), - ARIZONA_MUX_ROUTES("ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), - ARIZONA_MUX_ROUTES("ISRC1DEC1"), - ARIZONA_MUX_ROUTES("ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), - ARIZONA_MUX_ROUTES("ISRC2INT1"), - ARIZONA_MUX_ROUTES("ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), - ARIZONA_MUX_ROUTES("ISRC2DEC1"), - ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), ARIZONA_DSP_ROUTES("DSP1"), @@ -1468,6 +1599,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "SPKDAT1R", NULL, "OUT5R" }, { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, }; static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, @@ -1560,6 +1694,63 @@ static struct snd_soc_dai_driver wm5102_dai[] = { .ops = &arizona_dai_ops, .symmetric_rates = 1, }, + { + .name = "wm5102-slim1", + .id = 4, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5102-slim2", + .id = 5, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5102-slim3", + .id = 6, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, }; static int wm5102_codec_probe(struct snd_soc_codec *codec) @@ -1578,6 +1769,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_spk(codec); + arizona_init_gpio(codec); snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 88ad7db..bbd6438 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -58,14 +58,10 @@ static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0) static const struct snd_kcontrol_new wm5110_snd_controls[] = { -SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, - ARIZONA_IN4_OSR_SHIFT, 1, 0), +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), +SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), +SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), @@ -309,6 +305,15 @@ ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -360,6 +365,15 @@ ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); @@ -414,6 +428,9 @@ SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_INPUT("IN4L"), SND_SOC_DAPM_INPUT("IN4R"), +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), +SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), + SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | @@ -550,6 +567,56 @@ SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, @@ -640,6 +707,15 @@ ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"), ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), @@ -690,6 +766,14 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF2RX2", "AIF2RX2" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ { name, "EQ1", "EQ1" }, \ { name, "EQ2", "EQ2" }, \ { name, "EQ3", "EQ3" }, \ @@ -736,17 +820,27 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "OUT6L", NULL, "SYSCLK" }, { "OUT6R", NULL, "SYSCLK" }, + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "IN4L", NULL, "SYSCLK" }, + { "IN4R", NULL, "SYSCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, { "MICBIAS3", NULL, "MICVDD" }, + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + { "Noise Generator", NULL, "NOISE" }, { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -777,13 +871,41 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF3RX1", NULL, "AIF3 Playback" }, { "AIF3RX2", NULL, "AIF3 Playback" }, + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + { "AIF1 Playback", NULL, "SYSCLK" }, { "AIF2 Playback", NULL, "SYSCLK" }, { "AIF3 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, { "AIF1 Capture", NULL, "SYSCLK" }, { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, { "IN1L PGA", NULL, "IN1L" }, { "IN1R PGA", NULL, "IN1R" }, @@ -829,6 +951,15 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), @@ -844,10 +975,13 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, @@ -871,6 +1005,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "SPKDAT2R", NULL, "OUT6R" }, { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, + { "DRC2 Signal Activity", NULL, "DRC2L" }, + { "DRC2 Signal Activity", NULL, "DRC2R" }, }; static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, @@ -963,6 +1102,63 @@ static struct snd_soc_dai_driver wm5110_dai[] = { .ops = &arizona_dai_ops, .symmetric_rates = 1, }, + { + .name = "wm5110-slim1", + .id = 4, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5110-slim2", + .id = 5, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5110-slim3", + .id = 6, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, }; static int wm5110_codec_probe(struct snd_soc_codec *codec) @@ -978,6 +1174,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_spk(codec); + arizona_init_gpio(codec); snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0e8b3aa..af1318d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,8 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1319,8 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index af6d227..d2a0928 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -143,13 +143,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, } #define WM8400_OUTPGA_SINGLE_R_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_get_volsw, .put = wm8400_outpga_put_volsw_vu, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT_TLV(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm8400_outpga_put_volsw_vu, tlv_array) static const char *wm8400_digital_sidetone[] = diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 462f5e4..7b1a6d5 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route wm8727_dapm_routes[] = { + { "VOUTL", NULL, "Playback" }, + { "VOUTR", NULL, "Playback" }, +}; + /* * Note this is a simple chip with no configuration interface, sample rate is * determined automatically by examining the Master clock and Bit clock ratios @@ -43,7 +53,12 @@ static struct snd_soc_dai_driver wm8727_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8727; +static struct snd_soc_codec_driver soc_codec_dev_wm8727 = { + .dapm_widgets = wm8727_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets), + .dapm_routes = wm8727_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes), +}; static int wm8727_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5276062..456bb8c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -45,6 +45,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { struct wm8731_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; + const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; int sysclk_type; int playback_fs; @@ -290,6 +291,36 @@ static const struct _coeff_div coeff_div[] = { {12000000, 88200, 136, 0xf, 0x1, 0x1}, }; +/* rates constraints */ +static const unsigned int wm8731_rates_12000000[] = { + 8000, 32000, 44100, 48000, 96000, 88200, +}; + +static const unsigned int wm8731_rates_12288000_18432000[] = { + 8000, 32000, 48000, 96000, +}; + +static const unsigned int wm8731_rates_11289600_16934400[] = { + 8000, 44100, 88200, +}; + +static const struct snd_pcm_hw_constraint_list wm8731_constraints_12000000 = { + .list = wm8731_rates_12000000, + .count = ARRAY_SIZE(wm8731_rates_12000000), +}; + +static const +struct snd_pcm_hw_constraint_list wm8731_constraints_12288000_18432000 = { + .list = wm8731_rates_12288000_18432000, + .count = ARRAY_SIZE(wm8731_rates_12288000_18432000), +}; + +static const +struct snd_pcm_hw_constraint_list wm8731_constraints_11289600_16934400 = { + .list = wm8731_rates_11289600_16934400, + .count = ARRAY_SIZE(wm8731_rates_11289600_16934400), +}; + static inline int get_coeff(int mclk, int rate) { int i; @@ -362,17 +393,26 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, } switch (freq) { - case 11289600: + case 0: + wm8731->constraints = NULL; + break; case 12000000: + wm8731->constraints = &wm8731_constraints_12000000; + break; case 12288000: - case 16934400: case 18432000: - wm8731->sysclk = freq; + wm8731->constraints = &wm8731_constraints_12288000_18432000; + break; + case 16934400: + case 11289600: + wm8731->constraints = &wm8731_constraints_11289600_16934400; break; default: return -EINVAL; } + wm8731->sysclk = freq; + snd_soc_dapm_sync(&codec->dapm); return 0; @@ -475,12 +515,26 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int wm8731_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(dai->codec); + + if (wm8731->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8731->constraints); + + return 0; +} + #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { + .startup = wm8731_startup, .hw_params = wm8731_hw_params, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0a4ab4c..d96ebf5 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1456,8 +1456,9 @@ static int wm8753_resume(struct snd_soc_codec *codec) if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); codec->dapm.bias_level = SND_SOC_BIAS_ON; - schedule_delayed_work(&codec->dapm.delayed_work, - msecs_to_jiffies(caps_charge)); + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, + msecs_to_jiffies(caps_charge)); } return 0; diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f1fdbf6..8092495 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -26,6 +26,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782", .capture = { @@ -40,7 +50,12 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8782; +static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { + .dapm_widgets = wm8782_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), + .dapm_routes = wm8782_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), +}; static int wm8782_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 9d88437..eebcb1d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -364,9 +364,7 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; int ret; @@ -403,10 +401,8 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, } #define SOC_DAPM_SINGLE_W(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = wm8903_class_w_put, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, wm8903_class_w_put) static int wm8903_deemph[] = { 0, 32000, 44100, 48000 }; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c..4dfa8dc 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -603,13 +603,8 @@ SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("High Pass Filter Mode", hpf_mode), - -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC 128x OSR Switch", - .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, - .put = wm8904_adc_osr_put, - .private_value = SOC_SINGLE_VALUE(WM8904_ANALOGUE_ADC_0, 0, 1, 0), -}, +SOC_SINGLE_EXT("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0, + snd_soc_get_volsw, wm8904_adc_osr_put), }; static const char *drc_path_text[] = { @@ -1017,7 +1012,7 @@ static const struct soc_enum liner_enum = SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); static const struct snd_kcontrol_new liner_mux = - SOC_DAPM_ENUM("LINEL Mux", liner_enum); + SOC_DAPM_ENUM("LINER Mux", liner_enum); static const char *sidetone_text[] = { "None", "Left", "Right" @@ -1207,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) break; } - snd_soc_dapm_new_widgets(dapm); return 0; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd..f156010 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), -SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, - 0, 127, 0), +SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC, + 0, 255, 0, adc_tlv), SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", WM8960_BYPASS1, 4, 7, 1, bypass_tlv), @@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (pll_div.k) { reg |= 0x20; - snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff); } snd_soc_write(codec, WM8960_PLL1, reg); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e971028..11d80f3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -51,6 +51,7 @@ static const char *wm8962_supply_names[WM8962_NUM_SUPPLIES] = { /* codec private data */ struct wm8962_priv { + struct wm8962_pdata pdata; struct regmap *regmap; struct snd_soc_codec *codec; @@ -1600,7 +1601,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -1609,16 +1609,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, - reg_cache[WM8962_HPOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_HPOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_HPOUTL_VOLUME, + snd_soc_read(codec, WM8962_HPOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, - reg_cache[WM8962_HPOUTR_VOLUME]); + if (ret & WM8962_HPOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_HPOUTR_VOLUME, + snd_soc_read(codec, WM8962_HPOUTR_VOLUME)); - return 0; + return 1; } /* The VU bits for the speakers are in a different register to the mute @@ -2345,12 +2348,13 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { static int wm8962_add_widgets(struct snd_soc_codec *codec) { - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + struct wm8962_pdata *pdata = &wm8962->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_add_codec_controls(codec, wm8962_snd_controls, ARRAY_SIZE(wm8962_snd_controls)); - if (pdata && pdata->spk_mono) + if (pdata->spk_mono) snd_soc_add_codec_controls(codec, wm8962_spk_mono_controls, ARRAY_SIZE(wm8962_spk_mono_controls)); else @@ -2360,7 +2364,7 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8962_dapm_widgets, ARRAY_SIZE(wm8962_dapm_widgets)); - if (pdata && pdata->spk_mono) + if (pdata->spk_mono) snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_mono_widgets, ARRAY_SIZE(wm8962_dapm_spk_mono_widgets)); else @@ -2369,7 +2373,7 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8962_intercon, ARRAY_SIZE(wm8962_intercon)); - if (pdata && pdata->spk_mono) + if (pdata->spk_mono) snd_soc_dapm_add_routes(dapm, wm8962_spk_mono_intercon, ARRAY_SIZE(wm8962_spk_mono_intercon)); else @@ -2617,8 +2621,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } @@ -3042,8 +3044,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) pm_wakeup_event(dev, 300); - schedule_delayed_work(&wm8962->mic_work, - msecs_to_jiffies(250)); + queue_delayed_work(system_power_efficient_wq, + &wm8962->mic_work, + msecs_to_jiffies(250)); } return IRQ_HANDLED; @@ -3171,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev, long int time; int ret; - ret = strict_strtol(buf, 10, &time); + ret = kstrtol(buf, 10, &time); if (ret != 0) return ret; @@ -3333,14 +3336,14 @@ static struct gpio_chip wm8962_template_chip = { static void wm8962_init_gpio(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int ret; wm8962->gpio_chip = wm8962_template_chip; wm8962->gpio_chip.ngpio = WM8962_MAX_GPIO; wm8962->gpio_chip.dev = codec->dev; - if (pdata && pdata->gpio_base) + if (pdata->gpio_base) wm8962->gpio_chip.base = pdata->gpio_base; else wm8962->gpio_chip.base = -1; @@ -3373,8 +3376,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); - u16 *reg_cache = codec->reg_cache; + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3421,30 +3423,29 @@ static int wm8962_probe(struct snd_soc_codec *codec) WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, 0); - if (pdata) { - /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { - wm8962_set_gpio_mode(codec, i + 1); - snd_soc_write(codec, 0x200 + i, - pdata->gpio_init[i] & 0xffff); - } + /* Apply static configuration for GPIOs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) + if (pdata->gpio_init[i]) { + wm8962_set_gpio_mode(codec, i + 1); + snd_soc_write(codec, 0x200 + i, + pdata->gpio_init[i] & 0xffff); + } - /* Put the speakers into mono mode? */ - if (pdata->spk_mono) - reg_cache[WM8962_CLASS_D_CONTROL_2] - |= WM8962_SPK_MONO; - /* Micbias setup, detection enable and detection - * threasholds. */ - if (pdata->mic_cfg) - snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, - WM8962_MICDET_ENA | - WM8962_MICDET_THR_MASK | - WM8962_MICSHORT_THR_MASK | - WM8962_MICBIAS_LVL, - pdata->mic_cfg); - } + /* Put the speakers into mono mode? */ + if (pdata->spk_mono) + snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); + + /* Micbias setup, detection enable and detection + * threasholds. */ + if (pdata->mic_cfg) + snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, + WM8962_MICDET_ENA | + WM8962_MICDET_THR_MASK | + WM8962_MICSHORT_THR_MASK | + WM8962_MICBIAS_LVL, + pdata->mic_cfg); /* Latch volume update bits */ snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, @@ -3506,7 +3507,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_init_gpio(codec); if (wm8962->irq) { - if (pdata && pdata->irq_active_low) { + if (pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8962_IRQ_POL; } else { @@ -3584,6 +3585,34 @@ static const struct regmap_config wm8962_regmap = { .cache_type = REGCACHE_RBTREE, }; +static int wm8962_set_pdata_from_of(struct i2c_client *i2c, + struct wm8962_pdata *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + u32 val32; + int i; + + if (of_property_read_bool(np, "spk-mono")) + pdata->spk_mono = true; + + if (of_property_read_u32(np, "mic-cfg", &val32) >= 0) + pdata->mic_cfg = val32; + + if (of_property_read_u32_array(np, "gpio-cfg", pdata->gpio_init, + ARRAY_SIZE(pdata->gpio_init)) >= 0) + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) { + /* + * The range of GPIO register value is [0x0, 0xffff] + * While the default value of each register is 0x0 + * Any other value will be regarded as default value + */ + if (pdata->gpio_init[i] > 0xffff) + pdata->gpio_init[i] = 0x0; + } + + return 0; +} + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3603,6 +3632,15 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, init_completion(&wm8962->fll_lock); wm8962->irq = i2c->irq; + /* If platform data was supplied, update the default data in priv */ + if (pdata) { + memcpy(&wm8962->pdata, pdata, sizeof(struct wm8962_pdata)); + } else if (i2c->dev.of_node) { + ret = wm8962_set_pdata_from_of(i2c, &wm8962->pdata); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; @@ -3666,7 +3704,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, goto err_enable; } - if (pdata && pdata->in4_dc_measure) { + if (wm8962->pdata.in4_dc_measure) { ret = regmap_register_patch(wm8962->regmap, wm8962_dc_measure, ARRAY_SIZE(wm8962_dc_measure)); @@ -3719,8 +3757,34 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*5k (fast start-up) */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_BIAS_ENA | WM8962_VMID_SEL_MASK, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + return 0; } diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 029f31c..d8fc531 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .formats = WM8978_FORMATS, }, .ops = &wm8978_dai_ops, + .symmetric_rates = 1, }; static int wm8978_suspend(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 837978e..253c88b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -151,14 +151,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, } #define SOC_WM899X_OUTPGA_SINGLE_R_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_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + tlv_array) \ + SOC_SINGLE_EXT_TLV(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm899x_outpga_put_volsw_vu, tlv_array) static const char *wm8990_digital_sidetone[] = diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h index 8a942ef..07707d8 100644 --- a/sound/soc/codecs/wm8991.h +++ b/sound/soc/codecs/wm8991.h @@ -822,12 +822,7 @@ #define SOC_WM899X_OUTPGA_SINGLE_R_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_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT_TLV(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm899x_outpga_put_volsw_vu, tlv_array) #endif /* _WM8991_H */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29e95f9..86426a1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> @@ -289,10 +290,8 @@ static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ - .put = wm8994_put_drc_sw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, 1, 0) } + SOC_SINGLE_EXT(xname, reg, shift, 1, 0, \ + snd_soc_get_volsw, wm8994_put_drc_sw) static int wm8994_put_drc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -820,8 +819,9 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, * don't want false reports. */ if (wm8994->jackdet && !wm8994->clk_has_run) { - schedule_delayed_work(&wm8994->jackdet_bootstrap, - msecs_to_jiffies(1000)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->jackdet_bootstrap, + msecs_to_jiffies(1000)); wm8994->clk_has_run = true; } break; @@ -1432,17 +1432,13 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, }; #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = wm8994_put_class_w, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, wm8994_put_class_w) static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); @@ -1498,6 +1494,24 @@ static const char *aif1dac_text[] = { "AIF1DACDAT", "AIF3DACDAT", }; +static const char *loopback_text[] = { + "None", "ADCDAT", +}; + +static const struct soc_enum aif1_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif1_loopback = + SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); + +static const struct soc_enum aif2_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif2_loopback = + SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); + static const struct soc_enum aif1dac_enum = SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); @@ -1744,6 +1758,9 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback), +SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback), + SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1875,9 +1892,9 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, - { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" }, { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, - { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" }, { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, @@ -1928,6 +1945,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + /* Loopback */ + { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, + { "AIF1 Loopback", "None", "AIF1DACDAT" }, + { "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" }, + { "AIF2 Loopback", "None", "AIF2DACDAT" }, + /* Sidetone */ { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, { "Left Sidetone", "DMIC2", "DMIC2L" }, @@ -2010,15 +2033,16 @@ struct fll_div { u16 outdiv; u16 n; u16 k; + u16 lambda; u16 clk_ref_div; u16 fll_fratio; }; -static int wm8994_get_fll_config(struct fll_div *fll, +static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, int freq_in, int freq_out) { u64 Kpart; - unsigned int K, Ndiv, Nmod; + unsigned int K, Ndiv, Nmod, gcd_fll; pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out); @@ -2067,20 +2091,32 @@ static int wm8994_get_fll_config(struct fll_div *fll, Nmod = freq_out % freq_in; pr_debug("Nmod=%d\n", Nmod); - /* Calculate fractional part - scale up so we can round. */ - Kpart = FIXED_FLL_SIZE * (long long)Nmod; + switch (control->type) { + case WM8994: + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, freq_in); + + K = Kpart & 0xFFFFFFFF; - do_div(Kpart, freq_in); + if ((K % 10) >= 5) + K += 5; - K = Kpart & 0xFFFFFFFF; + /* Move down to proper range now rounding is done */ + fll->k = K / 10; + fll->lambda = 0; - if ((K % 10) >= 5) - K += 5; + pr_debug("N=%x K=%x\n", fll->n, fll->k); + break; - /* Move down to proper range now rounding is done */ - fll->k = K / 10; + default: + gcd_fll = gcd(freq_out, freq_in); - pr_debug("N=%x K=%x\n", fll->n, fll->k); + fll->k = (freq_out - (freq_in * fll->n)) / gcd_fll; + fll->lambda = freq_in / gcd_fll; + + } return 0; } @@ -2144,9 +2180,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, * analysis bugs spewing warnings. */ if (freq_out) - ret = wm8994_get_fll_config(&fll, freq_in, freq_out); + ret = wm8994_get_fll_config(control, &fll, freq_in, freq_out); else - ret = wm8994_get_fll_config(&fll, wm8994->fll[id].in, + ret = wm8994_get_fll_config(control, &fll, wm8994->fll[id].in, wm8994->fll[id].out); if (ret < 0) return ret; @@ -2191,6 +2227,17 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_N_MASK, fll.n << WM8994_FLL1_N_SHIFT); + if (fll.lambda) { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_1 + reg_offset, + WM8958_FLL1_LAMBDA_MASK, + fll.lambda); + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, WM8958_FLL1_EFS_ENA); + } else { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, 0); + } + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | @@ -2555,17 +2602,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2584,6 +2638,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2622,12 +2677,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2658,6 +2715,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } @@ -3096,24 +3157,7 @@ static int wm8994_codec_suspend(struct snd_soc_codec *codec) static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - unsigned int val, mask; - - if (control->revision < 4) { - /* force a HW read */ - ret = regmap_read(control->regmap, - WM8994_POWER_MANAGEMENT_5, &val); - - /* modify the cache only */ - codec->cache_only = 1; - mask = WM8994_DAC1R_ENA | WM8994_DAC1L_ENA | - WM8994_DAC2R_ENA | WM8994_DAC2L_ENA; - val &= mask; - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, - mask, val); - codec->cache_only = 0; - } for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { if (!wm8994->fll_suspend[i].out) @@ -3442,7 +3486,8 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) pm_wakeup_event(codec->dev, 300); - schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); + queue_delayed_work(system_power_efficient_wq, + &priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3495,6 +3540,31 @@ static void wm8958_button_det(struct snd_soc_codec *codec, u16 status) wm8994->btn_mask); } +static void wm8958_open_circuit_work(struct work_struct *work) +{ + struct wm8994_priv *wm8994 = container_of(work, + struct wm8994_priv, + open_circuit_work.work); + struct device *dev = wm8994->wm8994->dev; + + wm1811_micd_stop(wm8994->hubs.codec); + + mutex_lock(&wm8994->accdet_lock); + + dev_dbg(dev, "Reporting open circuit\n"); + + wm8994->jack_mic = false; + wm8994->mic_detecting = true; + + wm8958_micd_set_rate(wm8994->hubs.codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + wm8994->btn_mask | + SND_JACK_HEADSET); + + mutex_unlock(&wm8994->accdet_lock); +} + static void wm8958_mic_id(void *data, u16 status) { struct snd_soc_codec *codec = data; @@ -3504,16 +3574,10 @@ static void wm8958_mic_id(void *data, u16 status) if (!(status & WM8958_MICD_STS)) { /* If nothing present then clear our statuses */ dev_dbg(codec->dev, "Detected open circuit\n"); - wm8994->jack_mic = false; - wm8994->mic_detecting = true; - - wm1811_micd_stop(codec); - - wm8958_micd_set_rate(codec); - snd_soc_jack_report(wm8994->micdet[0].jack, 0, - wm8994->btn_mask | - SND_JACK_HEADSET); + queue_delayed_work(system_power_efficient_wq, + &wm8994->open_circuit_work, + msecs_to_jiffies(2500)); return; } @@ -3598,6 +3662,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) pm_runtime_get_sync(codec->dev); + cancel_delayed_work_sync(&wm8994->mic_complete_work); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); @@ -3625,8 +3691,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) WM1811_JACKDET_DB, 0); delay = control->pdata.micdet_delay; - schedule_delayed_work(&wm8994->mic_work, - msecs_to_jiffies(delay)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_work, + msecs_to_jiffies(delay)); } else { dev_dbg(codec->dev, "Jack not detected\n"); @@ -3780,11 +3847,29 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8958_mic_detect); +static void wm8958_mic_work(struct work_struct *work) +{ + struct wm8994_priv *wm8994 = container_of(work, + struct wm8994_priv, + mic_complete_work.work); + struct snd_soc_codec *codec = wm8994->hubs.codec; + + pm_runtime_get_sync(codec->dev); + + mutex_lock(&wm8994->accdet_lock); + + wm8994->mic_id_cb(wm8994->mic_id_cb_data, wm8994->mic_status); + + mutex_unlock(&wm8994->accdet_lock); + + pm_runtime_put(codec->dev); +} + static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->hubs.codec; - int reg, count, ret; + int reg, count, ret, id_delay; /* * Jack detection may have detected a removal simulataneously @@ -3794,6 +3879,9 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + cancel_delayed_work_sync(&wm8994->mic_complete_work); + cancel_delayed_work_sync(&wm8994->open_circuit_work); + pm_runtime_get_sync(codec->dev); /* We may occasionally read a detection without an impedence @@ -3846,8 +3934,13 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) goto out; } + wm8994->mic_status = reg; + id_delay = wm8994->wm8994->pdata.mic_id_delay; + if (wm8994->mic_detecting) - wm8994->mic_id_cb(wm8994->mic_id_cb_data, reg); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_complete_work, + msecs_to_jiffies(id_delay)); else wm8958_button_det(codec, reg); @@ -3899,6 +3992,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, wm1811_jackdet_bootstrap); + INIT_DELAYED_WORK(&wm8994->open_circuit_work, + wm8958_open_circuit_work); switch (control->type) { case WM8994: @@ -3911,14 +4006,13 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + INIT_DELAYED_WORK(&wm8994->mic_complete_work, wm8958_mic_work); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); wm8994->micdet_irq = control->pdata.micdet_irq; - pm_runtime_enable(codec->dev); - pm_runtime_idle(codec->dev); - /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; @@ -4291,8 +4385,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); - pm_runtime_disable(codec->dev); - for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); @@ -4351,6 +4443,9 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994, wm8994_dai, ARRAY_SIZE(wm8994_dai)); } @@ -4358,6 +4453,8 @@ static int wm8994_probe(struct platform_device *pdev) static int wm8994_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + return 0; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 55ddf4d..6536f8d 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -134,6 +134,9 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; struct delayed_work mic_work; + struct delayed_work open_circuit_work; + struct delayed_work mic_complete_work; + u16 mic_status; bool mic_detecting; bool jack_mic; int btn_mask; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 90a65c4..da2899e6 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -549,12 +549,9 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; - codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); return ret; diff --git a/sound/soc/codecs/wm8995.h b/sound/soc/codecs/wm8995.h index 5642121..508ad27 100644 --- a/sound/soc/codecs/wm8995.h +++ b/sound/soc/codecs/wm8995.h @@ -4237,11 +4237,8 @@ #define WM8995_SPK2_MUTE_SEQ1_WIDTH 8 /* SPK2_MUTE_SEQ1 - [7:0] */ #define WM8995_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = wm8995_put_class_w, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) \ -} + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, wm8995_put_class_w) struct wm8995_reg_access { u16 read; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 0000000..6ec3de3 --- /dev/null +++ b/sound/soc/codecs/wm8997.c @@ -0,0 +1,1175 @@ +/* + * wm8997.c -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm8997.h" + +struct wm8997_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +static const struct reg_default wm8997_sysclk_reva_patch[] = { + { 0x301D, 0x7B15 }, + { 0x301B, 0x0050 }, + { 0x305D, 0x7B17 }, + { 0x305B, 0x0050 }, + { 0x3001, 0x08FE }, + { 0x3003, 0x00F4 }, + { 0x3041, 0x08FF }, + { 0x3043, 0x0005 }, + { 0x3020, 0x0225 }, + { 0x3021, 0x0A00 }, + { 0x3022, 0xE24D }, + { 0x3023, 0x0800 }, + { 0x3024, 0xE24D }, + { 0x3025, 0xF000 }, + { 0x3060, 0x0226 }, + { 0x3061, 0x0A00 }, + { 0x3062, 0xE252 }, + { 0x3063, 0x0800 }, + { 0x3064, 0xE252 }, + { 0x3065, 0xF000 }, + { 0x3116, 0x022B }, + { 0x3117, 0xFA00 }, + { 0x3110, 0x246C }, + { 0x3111, 0x0A03 }, + { 0x3112, 0x246E }, + { 0x3113, 0x0A03 }, + { 0x3114, 0x2470 }, + { 0x3115, 0x0A03 }, + { 0x3126, 0x246C }, + { 0x3127, 0x0A02 }, + { 0x3128, 0x246E }, + { 0x3129, 0x0A02 }, + { 0x312A, 0x2470 }, + { 0x312B, 0xFA02 }, + { 0x3125, 0x0800 }, +}; + +static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm8997_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch); + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + default: + break; + } + + return 0; +} + +static const char *wm8997_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm8997_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm8997_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), +}; + +#define WM8997_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static const struct snd_kcontrol_new wm8997_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, + ARIZONA_EQ1_ENA_MASK), +SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, + ARIZONA_EQ2_ENA_MASK), +SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, + ARIZONA_EQ3_ENA_MASK), +SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, + ARIZONA_EQ4_ENA_MASK), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), +WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char *wm8997_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm8997_aec_loopback_values[] = { + 0, 1, 4, 6, 8, 9, +}; + +static const struct soc_enum wm8997_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(wm8997_aec_loopback_texts), + wm8997_aec_loopback_texts, + wm8997_aec_loopback_values); + +static const struct snd_kcontrol_new wm8997_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback); + +static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8997_aec_loopback_mux), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "EPOUT", "OUT3L" }, + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, +}; + +static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8997_FLL1: + return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout); + case WM8997_FLL2: + return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout); + case WM8997_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref, + Fout); + case WM8997_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8997_dai[] = { + { + .name = "wm8997-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-slim1", + .id = 3, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim2", + .id = 4, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim3", + .id = 5, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8997_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = priv->core.arizona->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + if (ret != 0) + return ret; + + arizona_init_spk(codec); + + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); + + priv->core.arizona->dapm = &codec->dapm; + + return 0; +} + +static int wm8997_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8997_DIG_VU 0x0200 + +static unsigned int wm8997_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { + .probe = wm8997_codec_probe, + .remove = wm8997_codec_remove, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8997_set_fll, + + .controls = wm8997_snd_controls, + .num_controls = ARRAY_SIZE(wm8997_snd_controls), + .dapm_widgets = wm8997_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets), + .dapm_routes = wm8997_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), +}; + +static int wm8997_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8997_priv *wm8997; + int i; + + wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), + GFP_KERNEL); + if (wm8997 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8997); + + wm8997->core.arizona = arizona; + wm8997->core.num_inputs = 4; + + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) + wm8997->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8997->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8997->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++) + arizona_init_dai(&wm8997->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], + WM8997_DIG_VU, WM8997_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); +} + +static int wm8997_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8997_codec_driver = { + .driver = { + .name = "wm8997-codec", + .owner = THIS_MODULE, + }, + .probe = wm8997_probe, + .remove = wm8997_remove, +}; + +module_platform_driver(wm8997_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8997 driver"); +MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8997-codec"); diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h new file mode 100644 index 0000000..5e91c6a --- /dev/null +++ b/sound/soc/codecs/wm8997.h @@ -0,0 +1,23 @@ +/* + * wm8997.h -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.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. + */ + +#ifndef _WM8997_H +#define _WM8997_H + +#include "arizona.h" + +#define WM8997_FLL1 1 +#define WM8997_FLL2 2 +#define WM8997_FLL1_REFCLK 3 +#define WM8997_FLL2_REFCLK 4 + +#endif diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 05b1f34..70ce6793 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -209,7 +209,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) case AC97_RESET: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -225,7 +225,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9705_reg))) cache[reg] = val; @@ -294,8 +294,8 @@ static struct snd_soc_dai_driver wm9705_dai[] = { static int wm9705_reset(struct snd_soc_codec *codec) { - if (soc_ac97_ops.reset) { - soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops->reset) { + soc_ac97_ops->reset(codec->ac97); if (ac97_read(codec, 0) == wm9705_reg[0]) return 0; /* Success */ } @@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { - soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + soc_ac97_ops->write(codec->ac97, AC97_POWERDOWN, 0xffff); return 0; } @@ -323,7 +323,7 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) } for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } return 0; @@ -337,9 +337,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9705 SoC Audio Codec\n"); - - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 8e9a6a3..c5eb746 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -455,7 +455,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_REC_GAIN) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -472,7 +472,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; @@ -581,15 +581,15 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; @@ -624,7 +624,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || (i > 0x58 && i != 0x5c)) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -635,7 +635,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f7afa68..a53e175 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -652,7 +652,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_CD) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -668,7 +668,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; @@ -1095,15 +1095,15 @@ static struct snd_soc_dai_driver wm9713_dai[] = { int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; @@ -1180,7 +1180,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || i == AC97_EXTENDED_MSTATUS || i > 0x66) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -1197,7 +1197,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b64..b38f350 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -21,6 +21,7 @@ #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -215,6 +216,29 @@ static struct { [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff_ctl { + const char *name; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int set:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -279,7 +303,7 @@ static const struct soc_enum wm_adsp2_rate_enum[] = { ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, arizona_rate_text, arizona_rate_val), @@ -334,6 +358,163 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(adsp->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->set = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +struct wmfw_ctl_work { + struct wm_adsp *adsp; + struct wm_coeff_ctl *ctl; + struct work_struct work; +}; + +static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!ctl || !ctl->name) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = snd_soc_add_card_controls(adsp->card, + kcontrol, 1); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card, + ctl->name); + + list_add(&ctl->list, &adsp->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -547,6 +728,152 @@ out: return ret; } +static int wm_coeff_init_control_caches(struct wm_adsp *adsp) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &adsp->ctl_list, list) { + if (!ctl->enabled || ctl->set) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_adsp *adsp) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &adsp->ctl_list, list) { + if (!ctl->enabled) + continue; + if (ctl->set) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + } + + return 0; +} + +static void wm_adsp_ctl_work(struct work_struct *work) +{ + struct wmfw_ctl_work *ctl_work = container_of(work, + struct wmfw_ctl_work, + work); + + wmfw_add_ctl(ctl_work->adsp, ctl_work->ctl); + kfree(ctl_work); +} + +static int wm_adsp_create_control(struct wm_adsp *dsp, + const struct wm_adsp_alg_region *region) + +{ + struct wm_coeff_ctl *ctl; + struct wmfw_ctl_work *ctl_work; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + ret = -EINVAL; + goto err_name; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + goto found; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->set = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); + if (!ctl_work) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + ctl_work->adsp = dsp; + ctl_work->ctl = ctl; + INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); + schedule_work(&ctl_work->work); + +found: + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + static int wm_adsp_setup_algs(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; @@ -730,7 +1057,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(dsp, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -738,7 +1074,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(dsp, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -758,7 +1103,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(dsp, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -766,7 +1120,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(dsp, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -774,7 +1137,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(dsp, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -986,9 +1358,12 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1031,6 +1406,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and unset controls */ + ret = wm_coeff_init_control_caches(dsp); + if (ret != 0) + goto err; + + /* Sync set controls */ + ret = wm_coeff_sync_controls(dsp); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -1047,6 +1432,9 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->ctl_list, list) + ctl->enabled = 0; break; default: @@ -1099,9 +1487,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: /* @@ -1172,6 +1563,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and unset controls */ + ret = wm_coeff_init_control_caches(dsp); + if (ret != 0) + goto err; + + /* Sync set controls */ + ret = wm_coeff_sync_controls(dsp); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1209,6 +1610,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->ctl_list, list) + ctl->enabled = 0; + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1246,6 +1650,7 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) } INIT_LIST_HEAD(&adsp->alg_regions); + INIT_LIST_HEAD(&adsp->ctl_list); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea5146..d018dea 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -38,6 +39,7 @@ struct wm_adsp { int type; struct device *dev; struct regmap *regmap; + struct snd_soc_card *card; int base; int sysclk_reg; @@ -55,17 +57,17 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct list_head ctl_list; }; #define WM_ADSP1(wname, num) \ - { .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = num, .event = wm_adsp1_event, \ - .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } + SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ + wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) #define WM_ADSP2(wname, num) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = num, .event = wm_adsp2_event, \ - .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } + SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ + wm_adsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f5d81b9..8b50e59 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -693,17 +693,13 @@ void wm_hubs_update_class_w(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); #define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, class_w_put_volsw) static int class_w_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); @@ -723,9 +719,7 @@ static int class_w_put_volsw(struct snd_kcontrol *kcontrol, static int class_w_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 9e11a14..c82f89c 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -54,16 +54,6 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI -config SND_DAVINCI_SOC_SFFSDR - tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_SFFSDR - select SND_DAVINCI_SOC_I2S - select SND_SOC_PCM3008 - select SFFSDR_FPGA - help - Say Y if you want to add support for SoC audio on - Lyrtech SFFSDR board. - config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a93679d..a396ab6 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -11,10 +11,8 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o -snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 484b22c..fd7c45b 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -14,6 +14,7 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/platform_data/edma.h> #include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 81490fe..32ddb7f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1024,7 +1024,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct device_node *np = pdev->dev.of_node; struct snd_platform_data *pdata = NULL; const struct of_device_id *match = - of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + of_match_device(mcasp_dt_ids, &pdev->dev); const u32 *of_serial_dir32; u8 *of_serial_dir; @@ -1257,7 +1257,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(mcasp_dt_ids), + .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index b2f27c2..8460edc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -17,6 +17,7 @@ #include <linux/dma-mapping.h> #include <linux/kernel.h> #include <linux/genalloc.h> +#include <linux/platform_data/edma.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index b6ef703..fbb710c 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -14,7 +14,7 @@ #include <linux/genalloc.h> #include <linux/platform_data/davinci_asp.h> -#include <mach/edma.h> +#include <linux/platform_data/edma.h> struct davinci_pcm_dma_params { int channel; /* sync dma channel ID */ diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c deleted file mode 100644 index 5be65aa..0000000 --- a/sound/soc/davinci/davinci-sffsdr.c +++ /dev/null @@ -1,181 +0,0 @@ -/* - * ASoC driver for Lyrtech SFFSDR board. - * - * Author: Hugo Villeneuve - * Copyright (C) 2008 Lyrtech inc - * - * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: - * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#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 <asm/dma.h> -#include <asm/mach-types.h> -#ifdef CONFIG_SFFSDR_FPGA -#include <asm/plat-sffsdr/sffsdr-fpga.h> -#endif - -#include <mach/edma.h> - -#include "../codecs/pcm3008.h" -#include "davinci-pcm.h" -#include "davinci-i2s.h" - -/* - * CLKX and CLKR are the inputs for the Sample Rate Generator. - * FSX and FSR are outputs, driven by the sample Rate Generator. - */ -#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ - SND_SOC_DAIFMT_CBM_CFS | \ - SND_SOC_DAIFMT_IB_NF) - -static int sffsdr_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->cpu_dai; - int fs; - int ret = 0; - - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - -#ifndef CONFIG_SFFSDR_FPGA - /* Without the FPGA module, the Fs is fixed at 44100 Hz */ - if (fs != 44100) { - pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); - return -EINVAL; - } -#endif - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); - -#ifndef CONFIG_SFFSDR_FPGA - return 0; -#else - return sffsdr_fpga_set_codec_fs(fs); -#endif -} - -static struct snd_soc_ops sffsdr_ops = { - .hw_params = sffsdr_hw_params, -}; - -/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sffsdr_dai = { - .name = "PCM3008", /* Codec name */ - .stream_name = "PCM3008 HiFi", - .cpu_dai_name = "davinci-mcbsp", - .codec_dai_name = "pcm3008-hifi", - .codec_name = "pcm3008-codec", - .platform_name = "davinci-mcbsp", - .ops = &sffsdr_ops, -}; - -/* davinci-sffsdr audio machine driver */ -static struct snd_soc_card snd_soc_sffsdr = { - .name = "DaVinci SFFSDR", - .owner = THIS_MODULE, - .dai_link = &sffsdr_dai, - .num_links = 1, -}; - -/* sffsdr audio private data */ -static struct pcm3008_setup_data sffsdr_pcm3008_setup = { - .dem0_pin = GPIO(45), - .dem1_pin = GPIO(46), - .pdad_pin = GPIO(47), - .pdda_pin = GPIO(38), -}; - -struct platform_device pcm3008_codec = { - .name = "pcm3008-codec", - .id = 0, - .dev = { - .platform_data = &sffsdr_pcm3008_setup, - }, -}; - -static struct resource sffsdr_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data sffsdr_snd_data = { - .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, - .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, -}; - -static struct platform_device *sffsdr_snd_device; - -static int __init sffsdr_init(void) -{ - int ret; - - if (!machine_is_sffsdr()) - return -EINVAL; - - platform_device_register(&pcm3008_codec); - - sffsdr_snd_device = platform_device_alloc("soc-audio", 0); - if (!sffsdr_snd_device) { - printk(KERN_ERR "platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(sffsdr_snd_device, &snd_soc_sffsdr); - platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data, - sizeof(sffsdr_snd_data)); - - ret = platform_device_add_resources(sffsdr_snd_device, - sffsdr_snd_resources, - ARRAY_SIZE(sffsdr_snd_resources)); - if (ret) { - printk(KERN_ERR "platform device add resources failed\n"); - goto error; - } - - ret = platform_device_add(sffsdr_snd_device); - if (ret) - goto error; - - return ret; - -error: - platform_device_put(sffsdr_snd_device); - return ret; -} - -static void __exit sffsdr_exit(void) -{ - platform_device_unregister(sffsdr_snd_device); - platform_device_unregister(&pcm3008_codec); -} - -module_init(sffsdr_init); -module_exit(sffsdr_exit); - -MODULE_AUTHOR("Hugo Villeneuve"); -MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 593a3ea1..25c31f1 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -1,7 +1,7 @@ /* * ALSA SoC Synopsys I2S Audio Layer * - * sound/soc/spear/designware_i2s.c + * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics * Rajeev Kumar <rajeev-dlh.kumar@st.com> @@ -396,7 +396,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->playback.channels_max = pdata->channel; dw_i2s_dai->playback.formats = pdata->snd_fmts; @@ -404,7 +404,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dev_dbg(&pdev->dev, "designware: record supported\n"); dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->capture.channels_max = pdata->channel; dw_i2s_dai->capture.formats = pdata->snd_fmts; @@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "not able to register dai\n"); - goto err_set_drvdata; + goto err_clk_disable; } return 0; -err_set_drvdata: - dev_set_drvdata(&pdev->dev, NULL); err_clk_disable: clk_disable(dev->clk); err_clk_put: @@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev) struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); clk_put(dev->clk); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3843a18..b7ab71f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,6 +1,9 @@ config SND_SOC_FSL_SSI tristate +config SND_SOC_FSL_SPDIF + tristate + config SND_SOC_FSL_UTILS tristate @@ -98,7 +101,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. @@ -108,18 +111,13 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM - tristate - config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ - select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM - select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate @@ -173,6 +171,17 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C @@ -180,14 +189,24 @@ config SND_SOC_IMX_SGTL5000 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_SPDIF + select REGMAP_MMIO + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" - depends on MFD_MC13783 + depends on MFD_MC13783 && ARM select SND_SOC_IMX_SSI select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index afd3479..8db705b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,9 +12,11 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o +obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o @@ -30,18 +32,11 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o -snd-soc-imx-pcm-objs := imx-pcm.o -ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),) - snd-soc-imx-pcm-objs += imx-pcm-fiq.o -endif -ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),) - snd-soc-imx-pcm-objs += imx-pcm-dma.o -endif - obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o @@ -49,6 +44,8 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -56,4 +53,6 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 75ffdf0..9a4a0ca 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -80,7 +80,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-fiq-pcm-audio.0", + .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c new file mode 100644 index 0000000..3920c3e --- /dev/null +++ b/sound/soc/fsl/fsl_spdif.c @@ -0,0 +1,1225 @@ +/* + * Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * Based on stmp3xxx_spdif_dai.c + * Vladimir Barinov <vbarinov@embeddedalley.com> + * Copyright 2008 SigmaTel, Inc + * Copyright 2008 Embedded Alley Solutions, Inc + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/clk-private.h> +#include <linux/bitrev.h> +#include <linux/regmap.h> +#include <linux/of_address.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include <sound/asoundef.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "fsl_spdif.h" +#include "imx-pcm.h" + +#define FSL_SPDIF_TXFIFO_WML 0x8 +#define FSL_SPDIF_RXFIFO_WML 0x8 + +#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) +#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\ + INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\ + INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED) + +/* Index list for the values that has if (DPLL Locked) condition */ +static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; +#define SRPC_NODPLL_START1 0x5 +#define SRPC_NODPLL_START2 0xc + +#define DEFAULT_RXCLK_SRC 1 + +/* + * SPDIF control structure + * Defines channel status, subcode and Q sub + */ +struct spdif_mixer_control { + /* spinlock to access control data */ + spinlock_t ctl_lock; + + /* IEC958 channel tx status bit */ + unsigned char ch_status[4]; + + /* User bits */ + unsigned char subcode[2 * SPDIF_UBITS_SIZE]; + + /* Q subcode part of user bits */ + unsigned char qsub[2 * SPDIF_QSUB_SIZE]; + + /* Buffer offset for U/Q */ + u32 upos; + u32 qpos; + + /* Ready buffer index of the two buffers */ + u32 ready_buf; +}; + +struct fsl_spdif_priv { + struct spdif_mixer_control fsl_spdif_control; + struct snd_soc_dai_driver cpu_dai_drv; + struct platform_device *pdev; + struct regmap *regmap; + bool dpll_locked; + u8 txclk_div[SPDIF_TXRATE_MAX]; + u8 txclk_src[SPDIF_TXRATE_MAX]; + u8 rxclk_src; + struct clk *txclk[SPDIF_TXRATE_MAX]; + struct clk *rxclk; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_dai_dma_data dma_params_rx; + + /* The name space will be allocated dynamically */ + char name[0]; +}; + + +/* DPLL locked and lock loss interrupt handler */ +static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 locked; + + regmap_read(regmap, REG_SPDIF_SRPC, &locked); + locked &= SRPC_DPLL_LOCKED; + + dev_dbg(&pdev->dev, "isr: Rx dpll %s \n", + locked ? "locked" : "loss lock"); + + spdif_priv->dpll_locked = locked ? true : false; +} + +/* Receiver found illegal symbol interrupt handler */ +static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + + dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n"); + + if (!spdif_priv->dpll_locked) { + /* DPLL unlocked seems no audio stream */ + regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0); + } +} + +/* U/Q Channel receive register full */ +static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 *pos, size, val, reg; + + switch (name) { + case 'U': + pos = &ctrl->upos; + size = SPDIF_UBITS_SIZE; + reg = REG_SPDIF_SRU; + break; + case 'Q': + pos = &ctrl->qpos; + size = SPDIF_QSUB_SIZE; + reg = REG_SPDIF_SRQ; + break; + default: + dev_err(&pdev->dev, "unsupported channel name\n"); + return; + } + + dev_dbg(&pdev->dev, "isr: %c Channel receive register full\n", name); + + if (*pos >= size * 2) { + *pos = 0; + } else if (unlikely((*pos % size) + 3 > size)) { + dev_err(&pdev->dev, "User bit receivce buffer overflow\n"); + return; + } + + regmap_read(regmap, reg, &val); + ctrl->subcode[*pos++] = val >> 16; + ctrl->subcode[*pos++] = val >> 8; + ctrl->subcode[*pos++] = val; +} + +/* U/Q Channel sync found */ +static void spdif_irq_uq_sync(struct fsl_spdif_priv *spdif_priv) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct platform_device *pdev = spdif_priv->pdev; + + dev_dbg(&pdev->dev, "isr: U/Q Channel sync found\n"); + + /* U/Q buffer reset */ + if (ctrl->qpos == 0) + return; + + /* Set ready to this buffer */ + ctrl->ready_buf = (ctrl->qpos - 1) / SPDIF_QSUB_SIZE + 1; +} + +/* U/Q Channel framing error */ +static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 val; + + dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n"); + + /* Read U/Q data to clear the irq and do buffer reset */ + regmap_read(regmap, REG_SPDIF_SRU, &val); + regmap_read(regmap, REG_SPDIF_SRQ, &val); + + /* Drop this U/Q buffer */ + ctrl->ready_buf = 0; + ctrl->upos = 0; + ctrl->qpos = 0; +} + +/* Get spdif interrupt status and clear the interrupt */ +static u32 spdif_intr_status_clear(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + u32 val, val2; + + regmap_read(regmap, REG_SPDIF_SIS, &val); + regmap_read(regmap, REG_SPDIF_SIE, &val2); + + regmap_write(regmap, REG_SPDIF_SIC, val & val2); + + return val; +} + +static irqreturn_t spdif_isr(int irq, void *devid) +{ + struct fsl_spdif_priv *spdif_priv = (struct fsl_spdif_priv *)devid; + struct platform_device *pdev = spdif_priv->pdev; + u32 sis; + + sis = spdif_intr_status_clear(spdif_priv); + + if (sis & INT_DPLL_LOCKED) + spdif_irq_dpll_lock(spdif_priv); + + if (sis & INT_TXFIFO_UNOV) + dev_dbg(&pdev->dev, "isr: Tx FIFO under/overrun\n"); + + if (sis & INT_TXFIFO_RESYNC) + dev_dbg(&pdev->dev, "isr: Tx FIFO resync\n"); + + if (sis & INT_CNEW) + dev_dbg(&pdev->dev, "isr: cstatus new\n"); + + if (sis & INT_VAL_NOGOOD) + dev_dbg(&pdev->dev, "isr: validity flag no good\n"); + + if (sis & INT_SYM_ERR) + spdif_irq_sym_error(spdif_priv); + + if (sis & INT_BIT_ERR) + dev_dbg(&pdev->dev, "isr: receiver found parity bit error\n"); + + if (sis & INT_URX_FUL) + spdif_irq_uqrx_full(spdif_priv, 'U'); + + if (sis & INT_URX_OV) + dev_dbg(&pdev->dev, "isr: U Channel receive register overrun\n"); + + if (sis & INT_QRX_FUL) + spdif_irq_uqrx_full(spdif_priv, 'Q'); + + if (sis & INT_QRX_OV) + dev_dbg(&pdev->dev, "isr: Q Channel receive register overrun\n"); + + if (sis & INT_UQ_SYNC) + spdif_irq_uq_sync(spdif_priv); + + if (sis & INT_UQ_ERR) + spdif_irq_uq_err(spdif_priv); + + if (sis & INT_RXFIFO_UNOV) + dev_dbg(&pdev->dev, "isr: Rx FIFO under/overrun\n"); + + if (sis & INT_RXFIFO_RESYNC) + dev_dbg(&pdev->dev, "isr: Rx FIFO resync\n"); + + if (sis & INT_LOSS_LOCK) + spdif_irq_dpll_lock(spdif_priv); + + /* FIXME: Write Tx FIFO to clear TxEm */ + if (sis & INT_TX_EM) + dev_dbg(&pdev->dev, "isr: Tx FIFO empty\n"); + + /* FIXME: Read Rx FIFO to clear RxFIFOFul */ + if (sis & INT_RXFIFO_FUL) + dev_dbg(&pdev->dev, "isr: Rx FIFO full\n"); + + return IRQ_HANDLED; +} + +static int spdif_softreset(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + u32 val, cycle = 1000; + + regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET); + + /* + * RESET bit would be cleared after finishing its reset procedure, + * which typically lasts 8 cycles. 1000 cycles will keep it safe. + */ + do { + regmap_read(regmap, REG_SPDIF_SCR, &val); + } while ((val & SCR_SOFT_RESET) && cycle--); + + if (cycle) + return 0; + else + return -EBUSY; +} + +static void spdif_set_cstatus(struct spdif_mixer_control *ctrl, + u8 mask, u8 cstatus) +{ + ctrl->ch_status[3] &= ~mask; + ctrl->ch_status[3] |= cstatus & mask; +} + +static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 ch_status; + + ch_status = (bitrev8(ctrl->ch_status[0]) << 16) | + (bitrev8(ctrl->ch_status[1]) << 8) | + bitrev8(ctrl->ch_status[2]); + regmap_write(regmap, REG_SPDIF_STCSCH, ch_status); + + dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status); + + ch_status = bitrev8(ctrl->ch_status[3]) << 16; + regmap_write(regmap, REG_SPDIF_STCSCL, ch_status); + + dev_dbg(&pdev->dev, "STCSCL: 0x%06x\n", ch_status); +} + +/* Set SPDIF PhaseConfig register for rx clock */ +static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv, + enum spdif_gainsel gainsel, int dpll_locked) +{ + struct regmap *regmap = spdif_priv->regmap; + u8 clksrc = spdif_priv->rxclk_src; + + if (clksrc >= SRPC_CLKSRC_MAX || gainsel >= GAINSEL_MULTI_MAX) + return -EINVAL; + + regmap_update_bits(regmap, REG_SPDIF_SRPC, + SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK, + SRPC_CLKSRC_SEL_SET(clksrc) | SRPC_GAINSEL_SET(gainsel)); + + return 0; +} + +static int spdif_set_sample_rate(struct snd_pcm_substream *substream, + int sample_rate) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + unsigned long csfs = 0; + u32 stc, mask, rate; + u8 clk, div; + int ret; + + switch (sample_rate) { + case 32000: + rate = SPDIF_TXRATE_32000; + csfs = IEC958_AES3_CON_FS_32000; + break; + case 44100: + rate = SPDIF_TXRATE_44100; + csfs = IEC958_AES3_CON_FS_44100; + break; + case 48000: + rate = SPDIF_TXRATE_48000; + csfs = IEC958_AES3_CON_FS_48000; + break; + default: + dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate); + return -EINVAL; + } + + clk = spdif_priv->txclk_src[rate]; + if (clk >= STC_TXCLK_SRC_MAX) { + dev_err(&pdev->dev, "tx clock source is out of range\n"); + return -EINVAL; + } + + div = spdif_priv->txclk_div[rate]; + if (div == 0) { + dev_err(&pdev->dev, "the divisor can't be zero\n"); + return -EINVAL; + } + + /* + * The S/PDIF block needs a clock of 64 * fs * div. The S/PDIF block + * will divide by (div). So request 64 * fs * (div+1) which will + * get rounded. + */ + ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (div + 1)); + if (ret) { + dev_err(&pdev->dev, "failed to set tx clock rate\n"); + return ret; + } + + dev_dbg(&pdev->dev, "expected clock rate = %d\n", + (64 * sample_rate * div)); + dev_dbg(&pdev->dev, "actual clock rate = %ld\n", + clk_get_rate(spdif_priv->txclk[rate])); + + /* set fs field in consumer channel status */ + spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); + + /* select clock source and divisor */ + stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DIV(div); + mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DIV_MASK; + regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc); + + dev_dbg(&pdev->dev, "set sample rate to %d\n", sample_rate); + + return 0; +} + +static int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct platform_device *pdev = spdif_priv->pdev; + struct regmap *regmap = spdif_priv->regmap; + u32 scr, mask, i; + int ret; + + /* Reset module and interrupts only for first initialization */ + if (!cpu_dai->active) { + ret = spdif_softreset(spdif_priv); + if (ret) { + dev_err(&pdev->dev, "failed to soft reset\n"); + return ret; + } + + /* Disable all the interrupts */ + regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL | + SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP | + SCR_TXFIFO_FSEL_IF8; + mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | + SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | + SCR_TXFIFO_FSEL_MASK; + for (i = 0; i < SPDIF_TXRATE_MAX; i++) + clk_prepare_enable(spdif_priv->txclk[i]); + } else { + scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; + clk_prepare_enable(spdif_priv->rxclk); + } + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Power up SPDIF module */ + regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0); + + return 0; +} + +static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 scr, mask, i; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + scr = 0; + mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | + SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | + SCR_TXFIFO_FSEL_MASK; + for (i = 0; i < SPDIF_TXRATE_MAX; i++) + clk_disable_unprepare(spdif_priv->txclk[i]); + } else { + scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; + clk_disable_unprepare(spdif_priv->rxclk); + } + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Power down SPDIF module only if tx&rx are both inactive */ + if (!cpu_dai->active) { + spdif_intr_status_clear(spdif_priv); + regmap_update_bits(regmap, REG_SPDIF_SCR, + SCR_LOW_POWER, SCR_LOW_POWER); + } +} + +static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct platform_device *pdev = spdif_priv->pdev; + u32 sample_rate = params_rate(params); + int ret = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = spdif_set_sample_rate(substream, sample_rate); + if (ret) { + dev_err(&pdev->dev, "%s: set sample rate failed: %d\n", + __func__, sample_rate); + return ret; + } + spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK, + IEC958_AES3_CON_CLOCK_1000PPM); + spdif_write_channel_status(spdif_priv); + } else { + /* Setup rx clock source */ + ret = spdif_set_rx_clksrc(spdif_priv, SPDIF_DEFAULT_GAINSEL, 1); + } + + return ret; +} + +static int fsl_spdif_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE; + u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(regmap, REG_SPDIF_SIE, intr, intr); + regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, dmaen); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0); + regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_spdif_dai_ops = { + .startup = fsl_spdif_startup, + .hw_params = fsl_spdif_hw_params, + .trigger = fsl_spdif_trigger, + .shutdown = fsl_spdif_shutdown, +}; + + +/* + * FSL SPDIF IEC958 controller(mixer) functions + * + * Channel status get/put control + * User bit value get/put control + * Valid bit value get control + * DPLL lock status get control + * User bit sync mode selection control + */ + +static int fsl_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int fsl_spdif_pb_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + + uvalue->value.iec958.status[0] = ctrl->ch_status[0]; + uvalue->value.iec958.status[1] = ctrl->ch_status[1]; + uvalue->value.iec958.status[2] = ctrl->ch_status[2]; + uvalue->value.iec958.status[3] = ctrl->ch_status[3]; + + return 0; +} + +static int fsl_spdif_pb_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + + ctrl->ch_status[0] = uvalue->value.iec958.status[0]; + ctrl->ch_status[1] = uvalue->value.iec958.status[1]; + ctrl->ch_status[2] = uvalue->value.iec958.status[2]; + ctrl->ch_status[3] = uvalue->value.iec958.status[3]; + + spdif_write_channel_status(spdif_priv); + + return 0; +} + +/* Get channel status from SPDIF_RX_CCHAN register */ +static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 cstatus, val; + + regmap_read(regmap, REG_SPDIF_SIS, &val); + if (!(val & INT_CNEW)) { + return -EAGAIN; + } + + regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); + ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; + ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF; + ucontrol->value.iec958.status[2] = cstatus & 0xFF; + + regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus); + ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF; + ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF; + ucontrol->value.iec958.status[5] = cstatus & 0xFF; + + /* Clear intr */ + regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW); + + return 0; +} + +/* + * Get User bits (subcode) from chip value which readed out + * in UChannel register. + */ +static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&ctrl->ctl_lock, flags); + if (ctrl->ready_buf) { + int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE; + memcpy(&ucontrol->value.iec958.subcode[0], + &ctrl->subcode[idx], SPDIF_UBITS_SIZE); + } else { + ret = -EAGAIN; + } + spin_unlock_irqrestore(&ctrl->ctl_lock, flags); + + return ret; +} + +/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */ +static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = SPDIF_QSUB_SIZE; + + return 0; +} + +/* Get Q subcode from chip value which readed out in QChannel register */ +static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&ctrl->ctl_lock, flags); + if (ctrl->ready_buf) { + int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE; + memcpy(&ucontrol->value.bytes.data[0], + &ctrl->qsub[idx], SPDIF_QSUB_SIZE); + } else { + ret = -EAGAIN; + } + spin_unlock_irqrestore(&ctrl->ctl_lock, flags); + + return ret; +} + +/* Valid bit infomation */ +static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +/* Get valid good bit from interrupt status register */ +static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val; + + val = regmap_read(regmap, REG_SPDIF_SIS, &val); + ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0; + regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD); + + return 0; +} + +/* DPLL lock infomation */ +static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 16000; + uinfo->value.integer.max = 96000; + + return 0; +} + +static u32 gainsel_multi[GAINSEL_MULTI_MAX] = { + 24, 16, 12, 8, 6, 4, 3, +}; + +/* Get RX data clock rate given the SPDIF bus_clk */ +static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, + enum spdif_gainsel gainsel) +{ + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u64 tmpval64, busclk_freq = 0; + u32 freqmeas, phaseconf; + u8 clksrc; + + regmap_read(regmap, REG_SPDIF_SRFM, &freqmeas); + regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf); + + clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; + if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { + /* Get bus clock from system */ + busclk_freq = clk_get_rate(spdif_priv->rxclk); + } + + /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ + tmpval64 = (u64) busclk_freq * freqmeas; + do_div(tmpval64, gainsel_multi[gainsel] * 1024); + do_div(tmpval64, 128 * 1024); + + dev_dbg(&pdev->dev, "FreqMeas: %d\n", freqmeas); + dev_dbg(&pdev->dev, "BusclkFreq: %lld\n", busclk_freq); + dev_dbg(&pdev->dev, "RxRate: %lld\n", tmpval64); + + return (int)tmpval64; +} + +/* + * Get DPLL lock or not info from stable interrupt status register. + * User application must use this control to get locked, + * then can do next PCM operation + */ +static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + + if (spdif_priv->dpll_locked) + ucontrol->value.integer.value[0] = rate; + else + ucontrol->value.integer.value[0] = 0; + + return 0; +} + +/* User bit sync mode info */ +static int fsl_spdif_usync_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +/* + * User bit sync mode: + * 1 CD User channel subcode + * 0 Non-CD data + */ +static int fsl_spdif_usync_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val; + + regmap_read(regmap, REG_SPDIF_SRCD, &val); + ucontrol->value.integer.value[0] = (val & SRCD_CD_USER) != 0; + + return 0; +} + +/* + * User bit sync mode: + * 1 CD User channel subcode + * 0 Non-CD data + */ +static int fsl_spdif_usync_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val = ucontrol->value.integer.value[0] << SRCD_CD_USER_OFFSET; + + regmap_update_bits(regmap, REG_SPDIF_SRCD, SRCD_CD_USER, val); + + return 0; +} + +/* FSL SPDIF IEC958 controller defines */ +static struct snd_kcontrol_new fsl_spdif_ctrls[] = { + /* Status cchanel controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_info, + .get = fsl_spdif_pb_get, + .put = fsl_spdif_pb_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_info, + .get = fsl_spdif_capture_get, + }, + /* User bits controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Subcode Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_info, + .get = fsl_spdif_subcode_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Q-subcode Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_qinfo, + .get = fsl_spdif_qget, + }, + /* Valid bit error controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 V-Bit Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_vbit_info, + .get = fsl_spdif_vbit_get, + }, + /* DPLL lock info get controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "RX Sample Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_rxrate_info, + .get = fsl_spdif_rxrate_get, + }, + /* User bit sync mode set/get controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 USyncMode CDText", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_usync_info, + .get = fsl_spdif_usync_get, + .put = fsl_spdif_usync_put, + }, +}; + +static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &spdif_private->dma_params_tx; + dai->capture_dma_data = &spdif_private->dma_params_rx; + + snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); + + return 0; +} + +static struct snd_soc_dai_driver fsl_spdif_dai = { + .probe = &fsl_spdif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = FSL_SPDIF_RATES_PLAYBACK, + .formats = FSL_SPDIF_FORMATS_PLAYBACK, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = FSL_SPDIF_RATES_CAPTURE, + .formats = FSL_SPDIF_FORMATS_CAPTURE, + }, + .ops = &fsl_spdif_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_spdif_component = { + .name = "fsl-spdif", +}; + +/* FSL SPDIF REGMAP */ + +static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_SPDIF_SCR: + case REG_SPDIF_SRCD: + case REG_SPDIF_SRPC: + case REG_SPDIF_SIE: + case REG_SPDIF_SIS: + case REG_SPDIF_SRL: + case REG_SPDIF_SRR: + case REG_SPDIF_SRCSH: + case REG_SPDIF_SRCSL: + case REG_SPDIF_SRU: + case REG_SPDIF_SRQ: + case REG_SPDIF_STCSCH: + case REG_SPDIF_STCSCL: + case REG_SPDIF_SRFM: + case REG_SPDIF_STC: + return true; + default: + return false; + }; +} + +static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_SPDIF_SCR: + case REG_SPDIF_SRCD: + case REG_SPDIF_SRPC: + case REG_SPDIF_SIE: + case REG_SPDIF_SIC: + case REG_SPDIF_STL: + case REG_SPDIF_STR: + case REG_SPDIF_STCSCH: + case REG_SPDIF_STCSCL: + case REG_SPDIF_STC: + return true; + default: + return false; + }; +} + +static const struct regmap_config fsl_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_SPDIF_STC, + .readable_reg = fsl_spdif_readable_reg, + .writeable_reg = fsl_spdif_writeable_reg, +}; + +static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, + struct clk *clk, u64 savesub, + enum spdif_txrate index) +{ + const u32 rate[] = { 32000, 44100, 48000 }; + u64 rate_ideal, rate_actual, sub; + u32 div, arate; + + for (div = 1; div <= 128; div++) { + rate_ideal = rate[index] * (div + 1) * 64; + rate_actual = clk_round_rate(clk, rate_ideal); + + arate = rate_actual / 64; + arate /= div; + + if (arate == rate[index]) { + /* We are lucky */ + savesub = 0; + spdif_priv->txclk_div[index] = div; + break; + } else if (arate / rate[index] == 1) { + /* A little bigger than expect */ + sub = (arate - rate[index]) * 100000; + do_div(sub, rate[index]); + if (sub < savesub) { + savesub = sub; + spdif_priv->txclk_div[index] = div; + } + } else if (rate[index] / arate == 1) { + /* A little smaller than expect */ + sub = (rate[index] - arate) * 100000; + do_div(sub, rate[index]); + if (sub < savesub) { + savesub = sub; + spdif_priv->txclk_div[index] = div; + } + } + } + + return savesub; +} + +static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, + enum spdif_txrate index) +{ + const u32 rate[] = { 32000, 44100, 48000 }; + struct platform_device *pdev = spdif_priv->pdev; + struct device *dev = &pdev->dev; + u64 savesub = 100000, ret; + struct clk *clk; + char tmp[16]; + int i; + + for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { + sprintf(tmp, "rxtx%d", i); + clk = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(clk)) { + dev_err(dev, "no rxtx%d clock in devicetree\n", i); + return PTR_ERR(clk); + } + if (!clk_get_rate(clk)) + continue; + + ret = fsl_spdif_txclk_caldiv(spdif_priv, clk, savesub, index); + if (savesub == ret) + continue; + + savesub = ret; + spdif_priv->txclk[index] = clk; + spdif_priv->txclk_src[index] = i; + + /* To quick catch a divisor, we allow a 0.1% deviation */ + if (savesub < 100) + break; + } + + dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n", + spdif_priv->txclk_src[index], rate[index]); + dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n", + spdif_priv->txclk_div[index], rate[index]); + + return 0; +} + +static int fsl_spdif_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_spdif_priv *spdif_priv; + struct spdif_mixer_control *ctrl; + struct resource *res; + void __iomem *regs; + int irq, ret, i; + + if (!np) + return -ENODEV; + + spdif_priv = devm_kzalloc(&pdev->dev, + sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, + GFP_KERNEL); + if (!spdif_priv) + return -ENOMEM; + + strcpy(spdif_priv->name, np->name); + + spdif_priv->pdev = pdev; + + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); + spdif_priv->cpu_dai_drv.name = spdif_priv->name; + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (IS_ERR(res)) { + dev_err(&pdev->dev, "could not determine device resources\n"); + return PTR_ERR(res); + } + + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_spdif_regmap_config); + if (IS_ERR(spdif_priv->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(spdif_priv->regmap); + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, + spdif_priv->name, spdif_priv); + if (ret) { + dev_err(&pdev->dev, "could not claim irq %u\n", irq); + return ret; + } + + /* Select clock source for rx/tx clock */ + spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1"); + if (IS_ERR(spdif_priv->rxclk)) { + dev_err(&pdev->dev, "no rxtx1 clock in devicetree\n"); + return PTR_ERR(spdif_priv->rxclk); + } + spdif_priv->rxclk_src = DEFAULT_RXCLK_SRC; + + for (i = 0; i < SPDIF_TXRATE_MAX; i++) { + ret = fsl_spdif_probe_txclk(spdif_priv, i); + if (ret) + return ret; + } + + /* Initial spinlock for control data */ + ctrl = &spdif_priv->fsl_spdif_control; + spin_lock_init(&ctrl->ctl_lock); + + /* Init tx channel status default value */ + ctrl->ch_status[0] = + IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015; + ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID; + ctrl->ch_status[2] = 0x00; + ctrl->ch_status[3] = + IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM; + + spdif_priv->dpll_locked = false; + + spdif_priv->dma_params_tx.maxburst = FSL_SPDIF_TXFIFO_WML; + spdif_priv->dma_params_rx.maxburst = FSL_SPDIF_RXFIFO_WML; + spdif_priv->dma_params_tx.addr = res->start + REG_SPDIF_STL; + spdif_priv->dma_params_rx.addr = res->start + REG_SPDIF_SRL; + + /* Register with ASoC */ + dev_set_drvdata(&pdev->dev, spdif_priv); + + ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + return ret; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) { + dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); + goto error_component; + } + + return ret; + +error_component: + snd_soc_unregister_component(&pdev->dev); + + return ret; +} + +static int fsl_spdif_remove(struct platform_device *pdev) +{ + imx_pcm_dma_exit(pdev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct of_device_id fsl_spdif_dt_ids[] = { + { .compatible = "fsl,imx35-spdif", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); + +static struct platform_driver fsl_spdif_driver = { + .driver = { + .name = "fsl-spdif-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_spdif_dt_ids, + }, + .probe = fsl_spdif_probe, + .remove = fsl_spdif_remove, +}; + +module_platform_driver(fsl_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale S/PDIF CPU DAI Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-spdif-dai"); diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h new file mode 100644 index 0000000..b126679 --- /dev/null +++ b/sound/soc/fsl/fsl_spdif.h @@ -0,0 +1,191 @@ +/* + * fsl_spdif.h - ALSA S/PDIF interface for the Freescale i.MX SoC + * + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <b42378@freescale.com> + * + * Based on fsl_ssi.h + * Author: Timur Tabi <timur@freescale.com> + * Copyright 2007-2008 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_SPDIF_DAI_H +#define _FSL_SPDIF_DAI_H + +/* S/PDIF Register Map */ +#define REG_SPDIF_SCR 0x0 /* SPDIF Configuration Register */ +#define REG_SPDIF_SRCD 0x4 /* CDText Control Register */ +#define REG_SPDIF_SRPC 0x8 /* PhaseConfig Register */ +#define REG_SPDIF_SIE 0xc /* InterruptEn Register */ +#define REG_SPDIF_SIS 0x10 /* InterruptStat Register */ +#define REG_SPDIF_SIC 0x10 /* InterruptClear Register */ +#define REG_SPDIF_SRL 0x14 /* SPDIFRxLeft Register */ +#define REG_SPDIF_SRR 0x18 /* SPDIFRxRight Register */ +#define REG_SPDIF_SRCSH 0x1c /* SPDIFRxCChannel_h Register */ +#define REG_SPDIF_SRCSL 0x20 /* SPDIFRxCChannel_l Register */ +#define REG_SPDIF_SRU 0x24 /* UchannelRx Register */ +#define REG_SPDIF_SRQ 0x28 /* QchannelRx Register */ +#define REG_SPDIF_STL 0x2C /* SPDIFTxLeft Register */ +#define REG_SPDIF_STR 0x30 /* SPDIFTxRight Register */ +#define REG_SPDIF_STCSCH 0x34 /* SPDIFTxCChannelCons_h Register */ +#define REG_SPDIF_STCSCL 0x38 /* SPDIFTxCChannelCons_l Register */ +#define REG_SPDIF_SRFM 0x44 /* FreqMeas Register */ +#define REG_SPDIF_STC 0x50 /* SPDIFTxClk Register */ + + +/* SPDIF Configuration register */ +#define SCR_RXFIFO_CTL_OFFSET 23 +#define SCR_RXFIFO_CTL_MASK (1 << SCR_RXFIFO_CTL_OFFSET) +#define SCR_RXFIFO_CTL_ZERO (1 << SCR_RXFIFO_CTL_OFFSET) +#define SCR_RXFIFO_OFF_OFFSET 22 +#define SCR_RXFIFO_OFF_MASK (1 << SCR_RXFIFO_OFF_OFFSET) +#define SCR_RXFIFO_OFF (1 << SCR_RXFIFO_OFF_OFFSET) +#define SCR_RXFIFO_RST_OFFSET 21 +#define SCR_RXFIFO_RST_MASK (1 << SCR_RXFIFO_RST_OFFSET) +#define SCR_RXFIFO_RST (1 << SCR_RXFIFO_RST_OFFSET) +#define SCR_RXFIFO_FSEL_OFFSET 19 +#define SCR_RXFIFO_FSEL_MASK (0x3 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF0 (0x0 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF4 (0x1 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF8 (0x2 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF12 (0x3 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_AUTOSYNC_OFFSET 18 +#define SCR_RXFIFO_AUTOSYNC_MASK (1 << SCR_RXFIFO_AUTOSYNC_OFFSET) +#define SCR_RXFIFO_AUTOSYNC (1 << SCR_RXFIFO_AUTOSYNC_OFFSET) +#define SCR_TXFIFO_AUTOSYNC_OFFSET 17 +#define SCR_TXFIFO_AUTOSYNC_MASK (1 << SCR_TXFIFO_AUTOSYNC_OFFSET) +#define SCR_TXFIFO_AUTOSYNC (1 << SCR_TXFIFO_AUTOSYNC_OFFSET) +#define SCR_TXFIFO_FSEL_OFFSET 15 +#define SCR_TXFIFO_FSEL_MASK (0x3 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF0 (0x0 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF4 (0x1 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF8 (0x2 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF12 (0x3 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_LOW_POWER (1 << 13) +#define SCR_SOFT_RESET (1 << 12) +#define SCR_TXFIFO_CTRL_OFFSET 10 +#define SCR_TXFIFO_CTRL_MASK (0x3 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_TXFIFO_CTRL_ZERO (0x0 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_TXFIFO_CTRL_NORMAL (0x1 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_TXFIFO_CTRL_ONESAMPLE (0x2 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_DMA_RX_EN_OFFSET 9 +#define SCR_DMA_RX_EN_MASK (1 << SCR_DMA_RX_EN_OFFSET) +#define SCR_DMA_RX_EN (1 << SCR_DMA_RX_EN_OFFSET) +#define SCR_DMA_TX_EN_OFFSET 8 +#define SCR_DMA_TX_EN_MASK (1 << SCR_DMA_TX_EN_OFFSET) +#define SCR_DMA_TX_EN (1 << SCR_DMA_TX_EN_OFFSET) +#define SCR_VAL_OFFSET 5 +#define SCR_VAL_MASK (1 << SCR_VAL_OFFSET) +#define SCR_VAL_CLEAR (1 << SCR_VAL_OFFSET) +#define SCR_TXSEL_OFFSET 2 +#define SCR_TXSEL_MASK (0x7 << SCR_TXSEL_OFFSET) +#define SCR_TXSEL_OFF (0 << SCR_TXSEL_OFFSET) +#define SCR_TXSEL_RX (1 << SCR_TXSEL_OFFSET) +#define SCR_TXSEL_NORMAL (0x5 << SCR_TXSEL_OFFSET) +#define SCR_USRC_SEL_OFFSET 0x0 +#define SCR_USRC_SEL_MASK (0x3 << SCR_USRC_SEL_OFFSET) +#define SCR_USRC_SEL_NONE (0x0 << SCR_USRC_SEL_OFFSET) +#define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET) +#define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET) + +/* SPDIF CDText control */ +#define SRCD_CD_USER_OFFSET 1 +#define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET) + +/* SPDIF Phase Configuration register */ +#define SRPC_DPLL_LOCKED (1 << 6) +#define SRPC_CLKSRC_SEL_OFFSET 7 +#define SRPC_CLKSRC_SEL_MASK (0xf << SRPC_CLKSRC_SEL_OFFSET) +#define SRPC_CLKSRC_SEL_SET(x) ((x << SRPC_CLKSRC_SEL_OFFSET) & SRPC_CLKSRC_SEL_MASK) +#define SRPC_CLKSRC_SEL_LOCKED_OFFSET1 5 +#define SRPC_CLKSRC_SEL_LOCKED_OFFSET2 2 +#define SRPC_GAINSEL_OFFSET 3 +#define SRPC_GAINSEL_MASK (0x7 << SRPC_GAINSEL_OFFSET) +#define SRPC_GAINSEL_SET(x) ((x << SRPC_GAINSEL_OFFSET) & SRPC_GAINSEL_MASK) + +#define SRPC_CLKSRC_MAX 16 + +enum spdif_gainsel { + GAINSEL_MULTI_24 = 0, + GAINSEL_MULTI_16, + GAINSEL_MULTI_12, + GAINSEL_MULTI_8, + GAINSEL_MULTI_6, + GAINSEL_MULTI_4, + GAINSEL_MULTI_3, +}; +#define GAINSEL_MULTI_MAX (GAINSEL_MULTI_3 + 1) +#define SPDIF_DEFAULT_GAINSEL GAINSEL_MULTI_8 + +/* SPDIF interrupt mask define */ +#define INT_DPLL_LOCKED (1 << 20) +#define INT_TXFIFO_UNOV (1 << 19) +#define INT_TXFIFO_RESYNC (1 << 18) +#define INT_CNEW (1 << 17) +#define INT_VAL_NOGOOD (1 << 16) +#define INT_SYM_ERR (1 << 15) +#define INT_BIT_ERR (1 << 14) +#define INT_URX_FUL (1 << 10) +#define INT_URX_OV (1 << 9) +#define INT_QRX_FUL (1 << 8) +#define INT_QRX_OV (1 << 7) +#define INT_UQ_SYNC (1 << 6) +#define INT_UQ_ERR (1 << 5) +#define INT_RXFIFO_UNOV (1 << 4) +#define INT_RXFIFO_RESYNC (1 << 3) +#define INT_LOSS_LOCK (1 << 2) +#define INT_TX_EM (1 << 1) +#define INT_RXFIFO_FUL (1 << 0) + +/* SPDIF Clock register */ +#define STC_SYSCLK_DIV_OFFSET 11 +#define STC_SYSCLK_DIV_MASK (0x1ff << STC_TXCLK_SRC_OFFSET) +#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK) +#define STC_TXCLK_SRC_OFFSET 8 +#define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET) +#define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK) +#define STC_TXCLK_ALL_EN_OFFSET 7 +#define STC_TXCLK_ALL_EN_MASK (1 << STC_TXCLK_ALL_EN_OFFSET) +#define STC_TXCLK_ALL_EN (1 << STC_TXCLK_ALL_EN_OFFSET) +#define STC_TXCLK_DIV_OFFSET 0 +#define STC_TXCLK_DIV_MASK (0x7ff << STC_TXCLK_DIV_OFFSET) +#define STC_TXCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_TXCLK_DIV_MASK) +#define STC_TXCLK_SRC_MAX 8 + +/* SPDIF tx rate */ +enum spdif_txrate { + SPDIF_TXRATE_32000 = 0, + SPDIF_TXRATE_44100, + SPDIF_TXRATE_48000, +}; +#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1) + + +#define SPDIF_CSTATUS_BYTE 6 +#define SPDIF_UBITS_SIZE 96 +#define SPDIF_QSUB_SIZE (SPDIF_UBITS_SIZE / 8) + + +#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_96000) + +#define FSL_SPDIF_FORMATS_PLAYBACK (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define FSL_SPDIF_FORMATS_CAPTURE (SNDRV_PCM_FMTBIT_S24_LE) + +#endif /* _FSL_SPDIF_DAI_H */ diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0f0bed6..c6b7439 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -8,6 +8,26 @@ * This file is licensed under the terms of the GNU General Public License * version 2. This program is licensed "as is" without any warranty of any * kind, whether express or implied. + * + * + * Some notes why imx-pcm-fiq is used instead of DMA on some boards: + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developed with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challenge. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. */ #include <linux/init.h> @@ -36,7 +56,7 @@ #define read_ssi(addr) in_be32(addr) #define write_ssi(val, addr) out_be32(addr, val) #define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) -#elif defined ARM +#else #define read_ssi(addr) readl(addr) #define write_ssi(val, addr) writel(val, addr) /* @@ -121,12 +141,14 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; + bool imx_ac97; + bool use_dma; struct clk *clk; - struct platform_device *imx_pcm_pdev; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; struct imx_dma_data filter_data_tx; struct imx_dma_data filter_data_rx; + struct imx_pcm_fiq_params fiq_params; struct { unsigned int rfrc; @@ -299,6 +321,102 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) return ret; } +static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) +{ + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u8 i2s_mode; + u8 wm; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; + + if (ssi_private->imx_ac97) + i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + else + i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + + /* + * Section 16.5 of the MPC8610 reference manual says that the SSI needs + * to be disabled before updating the registers we set here. + */ + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + + /* + * Program the SSI into I2S Slave Non-Network Synchronous mode. Also + * enable the transmit and receive FIFO. + * + * FIXME: Little-endian samples require a different shift dir + */ + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, + CCSR_SSI_SCR_TFR_CLK_DIS | + i2s_mode | + (synchronous ? CCSR_SSI_SCR_SYN : 0)); + + write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | + CCSR_SSI_STCR_TSCKP, &ssi->stcr); + + write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | + CCSR_SSI_SRCR_RSCKP, &ssi->srcr); + /* + * The DC and PM bits are only used if the SSI is the clock master. + */ + + /* + * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't + * use FIFO 1. We program the transmit water to signal a DMA transfer + * if there are only two (or fewer) elements left in the FIFO. Two + * elements equals one frame (left channel, right channel). This value, + * however, depends on the depth of the transmit buffer. + * + * We set the watermark on the same level as the DMA burstsize. For + * fiq it is probably better to use the biggest possible watermark + * size. + */ + if (ssi_private->use_dma) + wm = ssi_private->fifo_depth - 2; + else + wm = ssi_private->fifo_depth; + + write_ssi(CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) | + CCSR_SSI_SFCSR_TFWM1(wm) | CCSR_SSI_SFCSR_RFWM1(wm), + &ssi->sfcsr); + + /* + * For ac97 interrupts are enabled with the startup of the substream + * because it is also running without an active substream. Normally SSI + * is only enabled when there is a substream. + */ + if (ssi_private->imx_ac97) { + /* + * Setup the clock control register + */ + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->stccr); + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->srccr); + + /* + * Enable AC97 mode and startup the SSI + */ + write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV, + &ssi->sacnt); + write_ssi(0xff, &ssi->saccdis); + write_ssi(0x300, &ssi->saccen); + + /* + * Enable SSI, Transmit and Receive + */ + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | + CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); + + write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); + } + + return 0; +} + + /** * fsl_ssi_startup: create a new substream * @@ -320,70 +438,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * and initialize the SSI registers. */ if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - ssi_private->first_stream = substream; /* - * Section 16.5 of the MPC8610 reference manual says that the - * SSI needs to be disabled before updating the registers we set - * here. - */ - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - - /* - * Program the SSI into I2S Slave Non-Network Synchronous mode. - * Also enable the transmit and receive FIFO. - * - * FIXME: Little-endian samples require a different shift dir - */ - write_ssi_mask(&ssi->scr, - CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, - CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE - | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - - write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | - CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP, &ssi->stcr); - - write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | - CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP, &ssi->srcr); - - /* - * The DC and PM bits are only used if the SSI is the clock - * master. - */ - - /* Enable the interrupts and DMA requests */ - write_ssi(SIER_FLAGS, &ssi->sier); - - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We - * don't use FIFO 1. We program the transmit water to signal a - * DMA transfer if there are only two (or fewer) elements left - * in the FIFO. Two elements equals one frame (left channel, - * right channel). This value, however, depends on the depth of - * the transmit buffer. - * - * We program the receive FIFO to notify us if at least two - * elements (one frame) have been written to the FIFO. We could - * make this value larger (and maybe we should), but this way - * data will be written to memory as soon as it's available. - */ - write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | - CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2), - &ssi->sfcsr); - - /* - * We keep the SSI disabled because if we enable it, then the - * DMA controller will start. It's not supposed to start until - * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The - * DMA controller will transfer one "BWC" of data (i.e. the - * amount of data that the MR.BWC bits are set to). The reason - * this is bad is because at this point, the PCM driver has not - * finished initializing the DMA controller. + * fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ + if (!ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); } else { if (synchronous) { struct snd_pcm_runtime *first_runtime = @@ -493,6 +555,27 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sier_bits; + + /* + * Enable only the interrupts and DMA requests + * that are needed for the channel. As the fiq + * is polling for this bits, we have to ensure + * that this are aligned with the preallocated + * buffers + */ + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi_private->use_dma) + sier_bits = SIER_FLAGS; + else + sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN; + } else { + if (ssi_private->use_dma) + sier_bits = SIER_FLAGS; + else + sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN; + } switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -511,12 +594,18 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); + + if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); break; default: return -EINVAL; } + write_ssi(sier_bits, &ssi->sier); + return 0; } @@ -535,22 +624,13 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; - - /* - * If this is the last active substream, disable the SSI. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - } } static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx && ssi_private->use_dma) { dai->playback_dma_data = &ssi_private->dma_params_tx; dai->capture_dma_data = &ssi_private->dma_params_rx; } @@ -588,6 +668,133 @@ static const struct snd_soc_component_driver fsl_ssi_component = { .name = "fsl-ssi", }; +/** + * fsl_ssi_ac97_trigger: start and stop the AC97 receive/transmit. + * + * This function is called by ALSA to start, stop, pause, and resume the + * transfer of data. + */ +static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata( + rtd->cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TIE | + CCSR_SSI_SIER_TFE0_EN); + else + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_RFF0_EN); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TIE | + CCSR_SSI_SIER_TFE0_EN, 0); + else + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_RFF0_EN, 0); + break; + + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor); + else + write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor); + + return 0; +} + +static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { + .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_ac97_trigger, +}; + +static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &fsl_ssi_ac97_dai_ops, +}; + + +static struct fsl_ssi_private *fsl_ac97_data; + +static void fsl_ssi_ac97_init(void) +{ + fsl_ssi_setup(fsl_ac97_data); +} + +void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + + lreg = reg << 12; + write_ssi(lreg, &ssi->sacadd); + + lval = val << 4; + write_ssi(lval , &ssi->sacdat); + + write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK, + CCSR_SSI_SACNT_WR); + udelay(100); +} + +unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12; + write_ssi(lreg, &ssi->sacadd); + write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK, + CCSR_SSI_SACNT_RD); + + udelay(100); + + val = (read_ssi(&ssi->sacdat) >> 4) & 0xffff; + + return val; +} + +static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = { + .read = fsl_ssi_ac97_read, + .write = fsl_ssi_ac97_write, +}; + /* Show the statistics of a flag only if its interrupt is enabled. The * compiler will optimze this code to a no-op if the interrupt is not * enabled. @@ -664,6 +871,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource res; char name[64]; bool shared; + bool ac97 = false; /* SSIs that are not connected on the board should have a * status = "disabled" @@ -674,14 +882,20 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop || strcmp(sprop, "i2s-slave")) { + if (!sprop) { + dev_err(&pdev->dev, "fsl,mode property is necessary\n"); + return -EINVAL; + } + if (!strcmp(sprop, "ac97-slave")) { + ac97 = true; + } else if (strcmp(sprop, "i2s-slave")) { dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop); return -ENODEV; } /* The DAI name is the last part of the full name of the node. */ p = strrchr(np->full_name, '/') + 1; - ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), + ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private) + strlen(p), GFP_KERNEL); if (!ssi_private) { dev_err(&pdev->dev, "could not allocate DAI object\n"); @@ -690,38 +904,41 @@ static int fsl_ssi_probe(struct platform_device *pdev) strcpy(ssi_private->name, p); - /* Initialize this copy of the CPU DAI driver structure */ - memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, - sizeof(fsl_ssi_dai_template)); + ssi_private->use_dma = !of_property_read_bool(np, + "fsl,fiq-stream-filter"); + + if (ac97) { + memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai, + sizeof(fsl_ssi_ac97_dai)); + + fsl_ac97_data = ssi_private; + ssi_private->imx_ac97 = true; + + snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + } else { + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, + sizeof(fsl_ssi_dai_template)); + } ssi_private->cpu_dai_drv.name = ssi_private->name; /* Get the addresses and IRQ */ ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - goto error_kmalloc; + return ret; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - ret = -ENOMEM; - goto error_kmalloc; + return -ENOMEM; } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); if (ssi_private->irq == NO_IRQ) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - ret = -ENXIO; - goto error_iomap; - } - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, - ssi_private); - if (ret < 0) { - dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irqmap; + return -ENXIO; } /* Are the RX and the TX clocks locked? */ @@ -740,13 +957,18 @@ static int fsl_ssi_probe(struct platform_device *pdev) u32 dma_events[2]; ssi_private->ssi_on_imx = true; - ssi_private->clk = clk_get(&pdev->dev, NULL); + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); - goto error_irq; + goto error_irqmap; + } + ret = clk_prepare_enable(ssi_private->clk); + if (ret) { + dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", + ret); + goto error_irqmap; } - clk_prepare_enable(ssi_private->clk); /* * We have burstsize be "fifo_depth - 2" to match the SSI @@ -764,24 +986,38 @@ static int fsl_ssi_probe(struct platform_device *pdev) &ssi_private->filter_data_tx; ssi_private->dma_params_rx.filter_data = &ssi_private->filter_data_rx; - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32_array(pdev->dev.of_node, + if (!of_property_read_bool(pdev->dev.of_node, "dmas") && + ssi_private->use_dma) { + /* + * FIXME: This is a temporary solution until all + * necessary dma drivers support the generic dma + * bindings. + */ + ret = of_property_read_u32_array(pdev->dev.of_node, "fsl,ssi-dma-events", dma_events, 2); - if (ret) { - dev_err(&pdev->dev, "could not get dma events\n"); - goto error_clk; + if (ret && ssi_private->use_dma) { + dev_err(&pdev->dev, "could not get dma events but fsl-ssi is configured to use DMA\n"); + goto error_clk; + } } shared = of_device_is_compatible(of_get_parent(np), "fsl,spba-bus"); imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx, - dma_events[0], shared); + dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, - dma_events[1], shared); + dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); + } else if (ssi_private->use_dma) { + /* The 'name' should not have any slashes in it. */ + ret = devm_request_irq(&pdev->dev, ssi_private->irq, + fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", + ssi_private->irq); + goto error_irqmap; + } } /* Initialize the the device_attribute structure */ @@ -795,7 +1031,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error_irq; + goto error_clk; } /* Register with ASoC */ @@ -809,12 +1045,29 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (ssi_private->ssi_on_imx) { - ssi_private->imx_pcm_pdev = - platform_device_register_simple("imx-pcm-audio", - -1, NULL, 0); - if (IS_ERR(ssi_private->imx_pcm_pdev)) { - ret = PTR_ERR(ssi_private->imx_pcm_pdev); - goto error_dev; + if (!ssi_private->use_dma) { + + /* + * Some boards use an incompatible codec. To get it + * working, we are using imx-fiq-pcm-audio, that + * can handle those codecs. DMA is not possible in this + * situation. + */ + + ssi_private->fiq_params.irq = ssi_private->irq; + ssi_private->fiq_params.base = ssi_private->ssi; + ssi_private->fiq_params.dma_params_rx = + &ssi_private->dma_params_rx; + ssi_private->fiq_params.dma_params_tx = + &ssi_private->dma_params_tx; + + ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params); + if (ret) + goto error_dev; + } else { + ret = imx_pcm_dma_init(pdev); + if (ret) + goto error_dev; } } @@ -850,35 +1103,26 @@ static int fsl_ssi_probe(struct platform_device *pdev) } done: + if (ssi_private->imx_ac97) + fsl_ssi_ac97_init(); + return 0; error_dai: if (ssi_private->ssi_on_imx) - platform_device_unregister(ssi_private->imx_pcm_pdev); + imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); error_dev: - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } - -error_irq: - free_irq(ssi_private->irq, ssi_private); error_irqmap: irq_dispose_mapping(ssi_private->irq); -error_iomap: - iounmap(ssi_private->ssi); - -error_kmalloc: - kfree(ssi_private); - return ret; } @@ -888,20 +1132,15 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) { - platform_device_unregister(ssi_private->imx_pcm_pdev); - clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } + if (ssi_private->ssi_on_imx) + imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - - free_irq(ssi_private->irq, ssi_private); + if (ssi_private->ssi_on_imx) + clk_disable_unprepare(ssi_private->clk); irq_dispose_mapping(ssi_private->irq); - kfree(ssi_private); - dev_set_drvdata(&pdev->dev, NULL); - return 0; } @@ -924,6 +1163,7 @@ static struct platform_driver fsl_ssi_driver = { module_platform_driver(fsl_ssi_driver); +MODULE_ALIAS("platform:fsl-ssi-dai"); MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 47f046a..d3bf71a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,7 +26,6 @@ #include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/pinctrl/consumer.h> #include "imx-audmux.h" @@ -74,8 +73,11 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); @@ -225,14 +227,19 @@ EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, unsigned int pdcr) { + int ret; + if (audmux_type != IMX31_AUDMUX) return -EINVAL; if (!audmux_base) return -ENOSYS; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); @@ -244,10 +251,69 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); +static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, + struct device_node *of_node) +{ + struct device_node *child; + + for_each_available_child_of_node(of_node, child) { + unsigned int port; + unsigned int ptcr = 0; + unsigned int pdcr = 0; + unsigned int pcr = 0; + unsigned int val; + int ret; + int i = 0; + + ret = of_property_read_u32(child, "fsl,audmux-port", &port); + if (ret) { + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", + child->full_name); + continue; + } + if (!of_property_read_bool(child, "fsl,port-config")) { + dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", + child->full_name); + continue; + } + + for (i = 0; (ret = of_property_read_u32_index(child, + "fsl,port-config", i, &val)) == 0; + ++i) { + if (audmux_type == IMX31_AUDMUX) { + if (i % 2) + pdcr |= val; + else + ptcr |= val; + } else { + pcr |= val; + } + } + + if (ret != -EOVERFLOW) { + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", + i, child->full_name); + continue; + } + + if (audmux_type == IMX31_AUDMUX) { + if (i % 2) { + dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", + child->full_name); + continue; + } + imx_audmux_v2_configure_port(port, ptcr, pdcr); + } else { + imx_audmux_v1_configure_port(port, pcr); + } + } + + return 0; +} + static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; - struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -256,12 +322,6 @@ static int imx_audmux_probe(struct platform_device *pdev) if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "setup pinctrl failed!"); - return PTR_ERR(pinctrl); - } - audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", @@ -275,6 +335,9 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); + if (of_id) + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + return 0; } diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index b8ff44b..38a4209 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -1,57 +1,7 @@ #ifndef __IMX_AUDMUX_H #define __IMX_AUDMUX_H -#define MX27_AUDMUX_HPCR1_SSI0 0 -#define MX27_AUDMUX_HPCR2_SSI1 1 -#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 -#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 -#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 -#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 - -#define MX31_AUDMUX_PORT1_SSI0 0 -#define MX31_AUDMUX_PORT2_SSI1 1 -#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 -#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 -#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 -#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 -#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 - -#define MX51_AUDMUX_PORT1_SSI0 0 -#define MX51_AUDMUX_PORT2_SSI1 1 -#define MX51_AUDMUX_PORT3 2 -#define MX51_AUDMUX_PORT4 3 -#define MX51_AUDMUX_PORT5 4 -#define MX51_AUDMUX_PORT6 5 -#define MX51_AUDMUX_PORT7 6 - -/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) -#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) - -/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) +#include <dt-bindings/sound/fsl-imx-audmux.h> int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 4ae30f2..a3d60d4 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -64,7 +64,7 @@ static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = { .codec_dai_name = "mc13783-hifi", .codec_name = "mc13783-codec", .cpu_dai_name = "imx-ssi.0", - .platform_name = "imx-pcm-audio.0", + .platform_name = "imx-ssi.0", .ops = &imx_mc13783_hifi_ops, .symmetric_rates = 1, .dai_fmt = FMT_SSI, @@ -90,6 +90,7 @@ static const struct snd_soc_dapm_route imx_mc13783_routes[] = { static struct snd_soc_card imx_mc13783 = { .name = "imx_mc13783", + .owner = THIS_MODULE, .dai_link = imx_mc13783_dai_mc13783, .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783), .dapm_widgets = imx_mc13783_widget, diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index c246fb5..4dc1296 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -14,6 +14,7 @@ #include <linux/platform_device.h> #include <linux/dmaengine.h> #include <linux/types.h> +#include <linux/module.h> #include <sound/core.h> #include <sound/pcm.h> @@ -64,11 +65,14 @@ int imx_pcm_dma_init(struct platform_device *pdev) { return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } +EXPORT_SYMBOL_GPL(imx_pcm_dma_init); void imx_pcm_dma_exit(struct platform_device *pdev) { snd_dmaengine_pcm_unregister(&pdev->dev); } +EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 670b96b..34043c5 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -22,6 +22,7 @@ #include <linux/slab.h> #include <sound/core.h> +#include <sound/dmaengine_pcm.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -32,6 +33,7 @@ #include <linux/platform_data/asoc-imx-ssi.h> #include "imx-ssi.h" +#include "imx-pcm.h" struct imx_pcm_runtime_data { unsigned int period; @@ -225,6 +227,22 @@ static int snd_imx_close(struct snd_pcm_substream *substream) return 0; } +static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} + static struct snd_pcm_ops imx_pcm_ops = { .open = snd_imx_open, .close = snd_imx_close, @@ -236,6 +254,54 @@ static struct snd_pcm_ops imx_pcm_ops = { .mmap = snd_imx_pcm_mmap, }; +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) @@ -268,6 +334,27 @@ static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) return 0; } +static void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + static void imx_pcm_fiq_free(struct snd_pcm *pcm) { mxc_set_irq_fiq(ssi_irq, 0); @@ -281,9 +368,9 @@ static struct snd_soc_platform_driver imx_soc_platform_fiq = { .pcm_free = imx_pcm_fiq_free, }; -int imx_pcm_fiq_init(struct platform_device *pdev) +int imx_pcm_fiq_init(struct platform_device *pdev, + struct imx_pcm_fiq_params *params) { - struct imx_ssi *ssi = platform_get_drvdata(pdev); int ret; ret = claim_fiq(&fh); @@ -292,15 +379,15 @@ int imx_pcm_fiq_init(struct platform_device *pdev) return ret; } - mxc_set_irq_fiq(ssi->irq, 1); - ssi_irq = ssi->irq; + mxc_set_irq_fiq(params->irq, 1); + ssi_irq = params->irq; - imx_pcm_fiq = ssi->irq; + imx_pcm_fiq = params->irq; - imx_ssi_fiq_base = (unsigned long)ssi->base; + imx_ssi_fiq_base = (unsigned long)params->base; - ssi->dma_params_tx.maxburst = 4; - ssi->dma_params_rx.maxburst = 6; + params->dma_params_tx->maxburst = 4; + params->dma_params_rx->maxburst = 6; ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq); if (ret) @@ -314,3 +401,12 @@ failed_register: return ret; } +EXPORT_SYMBOL_GPL(imx_pcm_fiq_init); + +void imx_pcm_fiq_exit(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); +} +EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c deleted file mode 100644 index c498964..0000000 --- a/sound/soc/fsl/imx-pcm.c +++ /dev/null @@ -1,145 +0,0 @@ -/* - * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de> - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * 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/dma-mapping.h> -#include <linux/module.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include "imx-pcm.h" - -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - - pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); - return ret; -} -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); - -static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = IMX_SSI_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - - return 0; -} - -static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); - -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &imx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - -out: - return ret; -} -EXPORT_SYMBOL_GPL(imx_pcm_new); - -void imx_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(imx_pcm_free); - -static int imx_pcm_probe(struct platform_device *pdev) -{ - if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0) - return imx_pcm_fiq_init(pdev); - - return imx_pcm_dma_init(pdev); -} - -static int imx_pcm_remove(struct platform_device *pdev) -{ - if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0) - snd_soc_unregister_platform(&pdev->dev); - else - imx_pcm_dma_exit(pdev); - - return 0; -} - -static struct platform_device_id imx_pcm_devtype[] = { - { .name = "imx-pcm-audio", }, - { .name = "imx-fiq-pcm-audio", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(platform, imx_pcm_devtype); - -static struct platform_driver imx_pcm_driver = { - .driver = { - .name = "imx-pcm", - .owner = THIS_MODULE, - }, - .id_table = imx_pcm_devtype, - .probe = imx_pcm_probe, - .remove = imx_pcm_remove, -}; -module_platform_driver(imx_pcm_driver); - -MODULE_DESCRIPTION("Freescale i.MX PCM driver"); -MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index b7fa0d7..5d5b733 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,22 +22,23 @@ static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, - int dma, bool shared) + int dma, enum sdma_peripheral_type peripheral_type) { dma_data->dma_request = dma; dma_data->priority = DMA_PRIO_HIGH; - if (shared) - dma_data->peripheral_type = IMX_DMATYPE_SSI_SP; - else - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = peripheral_type; } -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); -void imx_pcm_free(struct snd_pcm *pcm); +struct imx_pcm_fiq_params { + int irq; + void __iomem *base; -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA + /* Pointer to original ssi driver to setup tx rx sizes */ + struct snd_dmaengine_dai_dma_data *dma_params_rx; + struct snd_dmaengine_dai_dma_data *dma_params_tx; +}; + +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); #else @@ -51,13 +52,20 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) } #endif -#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ -int imx_pcm_fiq_init(struct platform_device *pdev); +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) +int imx_pcm_fiq_init(struct platform_device *pdev, + struct imx_pcm_fiq_params *params); +void imx_pcm_fiq_exit(struct platform_device *pdev); #else -static inline int imx_pcm_fiq_init(struct platform_device *pdev) +static inline int imx_pcm_fiq_init(struct platform_device *pdev, + struct imx_pcm_fiq_params *params) { return -ENODEV; } + +static inline void imx_pcm_fiq_exit(struct platform_device *pdev) +{ +} #endif #endif /* _IMX_PCM_H */ diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..46c5b4f 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -13,7 +13,7 @@ #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> -#include <linux/of_i2c.h> +#include <linux/i2c.h> #include <linux/clk.h> #include <sound/soc.h> @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); @@ -128,28 +128,20 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = clk_get(&codec_dev->dev, NULL); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { - /* assuming clock enabled by default */ - data->codec_clk = NULL; - ret = of_property_read_u32(codec_np, "clock-frequency", - &data->clk_frequency); - if (ret) { - dev_err(&codec_dev->dev, - "clock-frequency missing or invalid\n"); - goto fail; - } - } else { - data->clk_frequency = clk_get_rate(data->codec_clk); - clk_prepare_enable(data->codec_clk); + ret = PTR_ERR(data->codec_clk); + goto fail; } + data->clk_frequency = clk_get_rate(data->codec_clk); + data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; data->dai.codec_dai_name = "sgtl5000"; data->dai.codec_of_node = codec_np; data->dai.cpu_of_node = ssi_np; - data->dai.platform_name = "imx-pcm-audio"; + data->dai.platform_of_node = ssi_np; data->dai.init = &imx_sgtl5000_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; @@ -157,10 +149,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -170,12 +162,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } platform_set_drvdata(pdev, data); -clk_fail: - clk_put(data->codec_clk); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + fail: if (ssi_np) of_node_put(ssi_np); @@ -189,10 +184,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - if (data->codec_clk) { - clk_disable_unprepare(data->codec_clk); - clk_put(data->codec_clk); - } snd_soc_unregister_card(&data->card); return 0; diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 0000000..816013b --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,148 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register dit failed: %d\n", ret); + goto end; + } + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register dir failed: %d\n", ret); + goto error_dit; + } + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto error_dir; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto error_dir; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + if (data->rxdev) + platform_device_unregister(data->rxdev); +error_dit: + if (data->txdev) + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + if (data->rxdev) + platform_device_unregister(data->rxdev); + if (data->txdev) + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index c6fa03e..f58bcd8 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -501,13 +501,12 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) imx_ssi_ac97_read(ac97, 0); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops imx_ssi_ac97_ops = { .read = imx_ssi_ac97_read, .write = imx_ssi_ac97_write, .reset = imx_ssi_ac97_reset, .warm_reset = imx_ssi_ac97_warm_reset }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int imx_ssi_probe(struct platform_device *pdev) { @@ -572,17 +571,23 @@ static int imx_ssi_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start, - false); + IMX_DMATYPE_SSI); } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start, - false); + IMX_DMATYPE_SSI); } platform_set_drvdata(pdev, ssi); + ret = snd_soc_set_ac97_ops(&imx_ssi_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto failed_register; + } + ret = snd_soc_register_component(&pdev->dev, &imx_component, dai, 1); if (ret) { @@ -590,46 +595,30 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_register; } - ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev_fiq) { - ret = -ENOMEM; - goto failed_pdev_fiq_alloc; - } - - platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi); - ret = platform_device_add(ssi->soc_platform_pdev_fiq); - if (ret) { - dev_err(&pdev->dev, "failed to add platform device\n"); - goto failed_pdev_fiq_add; - } + ssi->fiq_params.irq = ssi->irq; + ssi->fiq_params.base = ssi->base; + ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx; + ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx; - ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev) { - ret = -ENOMEM; - goto failed_pdev_alloc; - } + ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params); + if (ret) + goto failed_pcm_fiq; - platform_set_drvdata(ssi->soc_platform_pdev, ssi); - ret = platform_device_add(ssi->soc_platform_pdev); - if (ret) { - dev_err(&pdev->dev, "failed to add platform device\n"); - goto failed_pdev_add; - } + ret = imx_pcm_dma_init(pdev); + if (ret) + goto failed_pcm_dma; return 0; -failed_pdev_add: - platform_device_put(ssi->soc_platform_pdev); -failed_pdev_alloc: - platform_device_del(ssi->soc_platform_pdev_fiq); -failed_pdev_fiq_add: - platform_device_put(ssi->soc_platform_pdev_fiq); -failed_pdev_fiq_alloc: +failed_pcm_dma: + imx_pcm_fiq_exit(pdev); +failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -639,8 +628,8 @@ static int imx_ssi_remove(struct platform_device *pdev) struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); - platform_device_unregister(ssi->soc_platform_pdev); - platform_device_unregister(ssi->soc_platform_pdev_fiq); + imx_pcm_dma_exit(pdev); + imx_pcm_fiq_exit(pdev); snd_soc_unregister_component(&pdev->dev); @@ -649,6 +638,7 @@ static int imx_ssi_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index bb6b3db..fb1616b 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -209,11 +209,9 @@ struct imx_ssi { struct snd_dmaengine_dai_dma_data dma_params_tx; struct imx_dma_data filter_data_tx; struct imx_dma_data filter_data_rx; + struct imx_pcm_fiq_params fiq_params; int enabled; - - struct platform_device *soc_platform_pdev; - struct platform_device *soc_platform_pdev_fiq; }; #endif /* _IMX_SSI_H */ diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..722afe6 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,324 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + struct platform_device *pdev; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct device *dev = &priv->pdev->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, + "failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct device *dev = &priv->pdev->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(dev, "failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); + goto fail; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 4141b35..3ef7a0c 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -131,13 +131,12 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97) psc_ac97_warm_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = psc_ac97_read, .write = psc_ac97_write, .reset = psc_ac97_cold_reset, .warm_reset = psc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -290,6 +289,12 @@ static int psc_ac97_of_probe(struct platform_device *op) if (rc != 0) return rc; + rc = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (rc != 0) { + dev_err(&op->dev, "Failed to set AC'97 ops: %d\n", ret); + return rc; + } + rc = snd_soc_register_component(&op->dev, &psc_ac97_component, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { @@ -318,6 +323,7 @@ static int psc_ac97_of_remove(struct platform_device *op) { mpc5200_audio_dma_destroy(op); snd_soc_unregister_component(&op->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index 3d10741..f4c3bda 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -161,7 +161,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .name = "tlv320aic32x4", .stream_name = "TLV320AIC32X4", .codec_dai_name = "tlv320aic32x4-hifi", - .platform_name = "imx-pcm-audio.0", + .platform_name = "imx-ssi.0", .codec_name = "tlv320aic32x4.0-0018", .cpu_dai_name = "imx-ssi.0", .ops = &mx27vis_aic32x4_snd_ops, diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index f8da6dd..ae403c2 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -33,7 +33,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .cpu_dai_name = "imx-ssi.0", - .platform_name = "imx-fiq-pcm-audio.0", + .platform_name = "imx-ssi.0", .ops = &imx_phycore_hifi_ops, }, }; diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fe54a69..fce6325 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -245,7 +245,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { .stream_name = "Audio", .cpu_dai_name = "imx-ssi.0", .codec_dai_name = "wm8350-hifi", - .platform_name = "imx-fiq-pcm-audio.0", + .platform_name = "imx-ssi.0", .codec_name = "wm8350-codec.0-0x1a", .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6cf8355..8c49147 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .owner = THIS_MODULE, }, .probe = asoc_simple_card_probe, .remove = asoc_simple_card_remove, @@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = { module_platform_driver(asoc_simple_card); +MODULE_ALIAS("platform:asoc-simple-card"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ASoC Simple Sound Card"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9a12644..4c849a4 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -118,7 +118,7 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_FLUSH; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -140,7 +140,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -314,10 +314,10 @@ static int jz4740_i2s_suspend(struct snd_soc_dai *dai) conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } @@ -327,10 +327,10 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); if (dai->active) { - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -368,7 +368,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); jz4740_i2c_init_pcm_config(i2s); @@ -388,7 +388,7 @@ static int jz4740_i2s_dai_remove(struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } @@ -509,7 +509,6 @@ static int jz4740_i2s_dev_remove(struct platform_device *pdev) iounmap(i2s->base); release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - platform_set_drvdata(pdev, NULL); kfree(i2s); return 0; diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715..78ed4a4 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,19 +1,15 @@ config SND_KIRKWOOD_SOC - tristate "SoC Audio for the Marvell Kirkwood chip" - depends on ARCH_KIRKWOOD + tristate "SoC Audio for the Marvell Kirkwood and Dove chips" + depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. -config SND_KIRKWOOD_SOC_I2S - tristate - config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) + depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) depends on I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help Say Y if you want to add support for SoC audio on @@ -21,8 +17,7 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C - select SND_KIRKWOOD_SOC_I2S + depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C select SND_SOC_ALC5623 help Say Y if you want to add support for SoC audio on diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 3e62ae9..9e78138 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,6 @@ -snd-soc-kirkwood-objs := kirkwood-dma.o -snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o +snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index d3d4bdc..b238434 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -33,11 +33,11 @@ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE) -struct kirkwood_dma_priv { - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *rec_stream; - struct kirkwood_dma_data *data; -}; +static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) +{ + struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; + return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai); +} static struct snd_pcm_hardware kirkwood_dma_snd_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | @@ -51,7 +51,7 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { .rate_max = 384000, .channels_min = 1, .channels_max = 8, - .buffer_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS, + .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, .periods_min = KIRKWOOD_SND_MIN_PERIODS, @@ -63,8 +63,7 @@ static u64 kirkwood_dma_dmamask = DMA_BIT_MASK(32); static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) { - struct kirkwood_dma_priv *prdata = dev_id; - struct kirkwood_dma_data *priv = prdata->data; + struct kirkwood_dma_data *priv = dev_id; unsigned long mask, status, cause; mask = readl(priv->io + KIRKWOOD_INT_MASK); @@ -89,10 +88,10 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) writel(status, priv->io + KIRKWOOD_INT_CAUSE); if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES) - snd_pcm_period_elapsed(prdata->play_stream); + snd_pcm_period_elapsed(priv->substream_play); if (status & KIRKWOOD_INT_CAUSE_REC_BYTES) - snd_pcm_period_elapsed(prdata->rec_stream); + snd_pcm_period_elapsed(priv->substream_rec); return IRQ_HANDLED; } @@ -126,15 +125,10 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) { int err; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_platform *platform = soc_runtime->platform; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct kirkwood_dma_data *priv; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + struct kirkwood_dma_data *priv = kirkwood_priv(substream); const struct mbus_dram_target_info *dram; unsigned long addr; - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); /* Ensure that all constraints linked to dma burst are fulfilled */ @@ -157,21 +151,11 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) if (err < 0) return err; - if (prdata == NULL) { - prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL); - if (prdata == NULL) - return -ENOMEM; - - prdata->data = priv; - + if (!priv->substream_play && !priv->substream_rec) { err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, - "kirkwood-i2s", prdata); - if (err) { - kfree(prdata); + "kirkwood-i2s", priv); + if (err) return -EBUSY; - } - - snd_soc_platform_set_drvdata(platform, prdata); /* * Enable Error interrupts. We're only ack'ing them but @@ -183,11 +167,11 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) dram = mv_mbus_dram_info(); addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - prdata->play_stream = substream; + priv->substream_play = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { - prdata->rec_stream = substream; + priv->substream_rec = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_RECORD_WIN, addr, dram); } @@ -197,27 +181,19 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) static int kirkwood_dma_close(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct snd_soc_platform *platform = soc_runtime->platform; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); - struct kirkwood_dma_data *priv; - - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); + struct kirkwood_dma_data *priv = kirkwood_priv(substream); - if (!prdata || !priv) + if (!priv) return 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - prdata->play_stream = NULL; + priv->substream_play = NULL; else - prdata->rec_stream = NULL; + priv->substream_rec = NULL; - if (!prdata->play_stream && !prdata->rec_stream) { + if (!priv->substream_play && !priv->substream_rec) { writel(0, priv->io + KIRKWOOD_ERR_MASK); - free_irq(priv->irq, prdata); - kfree(prdata); - snd_soc_platform_set_drvdata(platform, NULL); + free_irq(priv->irq, priv); } return 0; @@ -243,13 +219,9 @@ static int kirkwood_dma_hw_free(struct snd_pcm_substream *substream) static int kirkwood_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct kirkwood_dma_data *priv; + struct kirkwood_dma_data *priv = kirkwood_priv(substream); unsigned long size, count; - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); - /* compute buffer size in term of "words" as requested in specs */ size = frames_to_bytes(runtime, runtime->buffer_size); size = (size>>2)-1; @@ -272,13 +244,9 @@ static int kirkwood_dma_prepare(struct snd_pcm_substream *substream) static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct kirkwood_dma_data *priv; + struct kirkwood_dma_data *priv = kirkwood_priv(substream); snd_pcm_uframes_t count; - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = bytes_to_frames(substream->runtime, readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT)); @@ -289,7 +257,7 @@ static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream return count; } -struct snd_pcm_ops kirkwood_dma_ops = { +static struct snd_pcm_ops kirkwood_dma_ops = { .open = kirkwood_dma_open, .close = kirkwood_dma_close, .ioctl = snd_pcm_lib_ioctl, @@ -366,36 +334,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver kirkwood_soc_platform = { +struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, }; - -static int kirkwood_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); -} - -static int kirkwood_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver kirkwood_pcm_driver = { - .driver = { - .name = "kirkwood-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = kirkwood_soc_platform_probe, - .remove = kirkwood_soc_platform_remove, -}; - -module_platform_driver(kirkwood_pcm_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); -MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-pcm-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..0f3d73d 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,13 +22,12 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" -#define DRV_NAME "kirkwood-i2s" +#define DRV_NAME "mvebu-audio" -#define KIRKWOOD_I2S_RATES \ - (SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) #define KIRKWOOD_I2S_FORMATS \ (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ @@ -105,14 +104,16 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for supported rates */ + /* use internal dco for the supported rates + * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { - /* use optional external clk for other rates */ + } else { + /* use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate); @@ -199,8 +200,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, ctl_play |= KIRKWOOD_PLAYCTL_MONO_OFF; priv->ctl_play &= ~(KIRKWOOD_PLAYCTL_MONO_MASK | - KIRKWOOD_PLAYCTL_I2S_EN | - KIRKWOOD_PLAYCTL_SPDIF_EN | + KIRKWOOD_PLAYCTL_ENABLE_MASK | KIRKWOOD_PLAYCTL_SIZE_MASK); priv->ctl_play |= ctl_play; } else { @@ -244,8 +244,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_play; - value = ctl & ~(KIRKWOOD_PLAYCTL_I2S_EN | - KIRKWOOD_PLAYCTL_SPDIF_EN); + value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); /* enable interrupts */ @@ -267,7 +266,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, writel(value, priv->io + KIRKWOOD_INT_MASK); /* disable all playbacks */ - ctl &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); + ctl &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -387,7 +386,7 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) /* disable playback/record */ value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN); + value &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_RECCTL); @@ -398,11 +397,6 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) } -static int kirkwood_i2s_remove(struct snd_soc_dai *dai) -{ - return 0; -} - static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .startup = kirkwood_i2s_startup, .trigger = kirkwood_i2s_trigger, @@ -413,17 +407,18 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { static struct snd_soc_dai_driver kirkwood_i2s_dai = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, - .rates = KIRKWOOD_I2S_RATES, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, .formats = KIRKWOOD_I2S_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = KIRKWOOD_I2S_RATES, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, @@ -431,7 +426,6 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, @@ -461,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -481,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -498,10 +495,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (err < 0) return err; - priv->extclk = clk_get(&pdev->dev, "extclk"); + priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { - clk_put(priv->extclk); + devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); @@ -515,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -525,14 +522,22 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, soc_dai, 1); - if (!err) - return 0; - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + goto err_component; + } - if (!IS_ERR(priv->extclk)) { - clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); + err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); + goto err_platform; } + return 0; + err_platform: + snd_soc_unregister_component(&pdev->dev); + err_component: + if (!IS_ERR(priv->extclk)) + clk_disable_unprepare(priv->extclk); clk_disable_unprepare(priv->clk); return err; @@ -542,23 +547,32 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - if (!IS_ERR(priv->extclk)) { + if (!IS_ERR(priv->extclk)) clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); - } clk_disable_unprepare(priv->clk); return 0; } +#ifdef CONFIG_OF +static struct of_device_id mvebu_audio_of_match[] = { + { .compatible = "marvell,kirkwood-audio" }, + { .compatible = "marvell,dove-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mvebu_audio_of_match), }, }; @@ -568,4 +582,4 @@ module_platform_driver(kirkwood_i2s_driver); MODULE_AUTHOR("Arnaud Patard, <arnaud.patard@rtp-net.org>"); MODULE_DESCRIPTION("Kirkwood I2S SoC Interface"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-i2s"); +MODULE_ALIAS("platform:mvebu-audio"); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..025be0e 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -16,9 +16,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/cs42l51.h" static int openrd_client_hw_params(struct snd_pcm_substream *substream, @@ -54,8 +52,8 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .cpu_dai_name = "mvebu-audio", + .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..27545b0 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -15,9 +15,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/alc5623.h" static int t5325_hw_params(struct snd_pcm_substream *substream, @@ -70,8 +68,8 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .cpu_dai_name = "mvebu-audio", + .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 4d92637..f8e1ccc 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -54,7 +54,7 @@ #define KIRKWOOD_PLAYCTL_MONO_OFF (0<<5) #define KIRKWOOD_PLAYCTL_I2S_MUTE (1<<7) #define KIRKWOOD_PLAYCTL_SPDIF_EN (1<<4) -#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3) +#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3) #define KIRKWOOD_PLAYCTL_SIZE_MASK (7<<0) #define KIRKWOOD_PLAYCTL_SIZE_16 (7<<0) #define KIRKWOOD_PLAYCTL_SIZE_16_C (3<<0) @@ -62,6 +62,9 @@ #define KIRKWOOD_PLAYCTL_SIZE_24 (1<<0) #define KIRKWOOD_PLAYCTL_SIZE_32 (0<<0) +#define KIRKWOOD_PLAYCTL_ENABLE_MASK (KIRKWOOD_PLAYCTL_SPDIF_EN | \ + KIRKWOOD_PLAYCTL_I2S_EN) + #define KIRKWOOD_PLAY_BUF_ADDR 0x1104 #define KIRKWOOD_PLAY_BUF_SIZE 0x1108 #define KIRKWOOD_PLAY_BYTE_COUNT 0x110C @@ -122,6 +125,8 @@ #define KIRKWOOD_SND_MAX_PERIODS 16 #define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 #define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ + * KIRKWOOD_SND_MAX_PERIODS) struct kirkwood_dma_data { void __iomem *io; @@ -129,8 +134,12 @@ struct kirkwood_dma_data { struct clk *extclk; uint32_t ctl_play; uint32_t ctl_rec; + struct snd_pcm_substream *substream_play; + struct snd_pcm_substream *substream_rec; int irq; int burst; }; +extern struct snd_soc_platform_driver kirkwood_soc_platform; + #endif diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 4139116..ee36384 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -371,7 +371,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) /* audio interrupt base of SRAM location where * interrupts are stored by System FW */ - mc_drv_ctx = kzalloc(sizeof(*mc_drv_ctx), GFP_ATOMIC); + mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); if (!mc_drv_ctx) { pr_err("allocation failed\n"); return -ENOMEM; @@ -381,51 +381,39 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) pdev, IORESOURCE_MEM, "IRQ_BASE"); if (!irq_mem) { pr_err("no mem resource given\n"); - ret_val = -ENODEV; - goto unalloc; + return -ENODEV; } - mc_drv_ctx->int_base = ioremap_nocache(irq_mem->start, - resource_size(irq_mem)); + mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, + resource_size(irq_mem)); if (!mc_drv_ctx->int_base) { pr_err("Mapping of cache failed\n"); - ret_val = -ENOMEM; - goto unalloc; + return -ENOMEM; } /* register for interrupt */ - ret_val = request_threaded_irq(irq, snd_mfld_jack_intr_handler, + ret_val = devm_request_threaded_irq(&pdev->dev, irq, + snd_mfld_jack_intr_handler, snd_mfld_jack_detection, IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); if (ret_val) { pr_err("cannot register IRQ\n"); - goto unalloc; + return ret_val; } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; ret_val = snd_soc_register_card(&snd_soc_card_mfld); if (ret_val) { pr_debug("snd_soc_register_card failed %d\n", ret_val); - goto freeirq; + return ret_val; } platform_set_drvdata(pdev, mc_drv_ctx); pr_debug("successfully exited probe\n"); - return ret_val; - -freeirq: - free_irq(irq, mc_drv_ctx); -unalloc: - kfree(mc_drv_ctx); - return ret_val; + return 0; } static int snd_mfld_mc_remove(struct platform_device *pdev) { - struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); - free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); snd_soc_unregister_card(&snd_soc_card_mfld); - kfree(mc_drv_ctx); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 78d321c..219235c 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" - depends on ARCH_MXS + depends on ARCH_MXS || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b41fffc..b16abbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { .fifo_size = 32, }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_params *dma_params = param; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &dma_params->dma_data; - - return true; -} - static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { .pcm_hardware = &snd_mxs_hardware, - .compat_filter_fn = filter, .prealloc_buffer_size = 64 * 1024, }; @@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_COMPAT | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 3aa918f..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,13 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include <linux/fsl/mxs-dma.h> - -struct mxs_pcm_dma_params { - struct mxs_dma_data dma_data; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d31dc52..b56b8a0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -24,15 +24,13 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> -#include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> #include "mxs-saif.h" @@ -605,8 +603,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); - dai->playback_dma_data = &saif->dma_param; - dai->capture_dma_data = &saif->dma_param; return 0; } @@ -662,12 +658,38 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) return IRQ_HANDLED; } +static int mxs_saif_mclk_init(struct platform_device *pdev) +{ + struct mxs_saif *saif = platform_get_drvdata(pdev); + struct device_node *np = pdev->dev.of_node; + struct clk *clk; + int ret; + + clk = clk_register_divider(&pdev->dev, "mxs_saif_mclk", + __clk_get_name(saif->clk), 0, + saif->base + SAIF_CTRL, + BP_SAIF_CTRL_BITCLK_MULT_RATE, 3, + 0, NULL); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + if (ret == -EEXIST) + return 0; + dev_err(&pdev->dev, "failed to register mclk: %d\n", ret); + return PTR_ERR(clk); + } + + ret = of_clk_add_provider(np, of_clk_src_simple_get, clk); + if (ret) + return ret; + + return 0; +} + static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; - struct pinctrl *pinctrl; int ret = 0; struct device_node *master; @@ -707,12 +729,6 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[saif->id] = saif; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - ret = PTR_ERR(pinctrl); - return ret; - } - saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); @@ -727,22 +743,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -759,16 +759,15 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.dma_data.chan_irq < 0) { - ret = saif->dma_param.dma_data.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } - platform_set_drvdata(pdev, saif); + /* We only support saif0 being tx and clock master */ + if (saif->id == 0) { + ret = mxs_saif_mclk_init(pdev); + if (ret) + dev_warn(&pdev->dev, "failed to init clocks\n"); + } + ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, &mxs_saif_dai, 1); if (ret) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..53eaa4b 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -117,7 +117,6 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5e..4bb2737 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -25,7 +25,6 @@ #include <sound/soc.h> #include <sound/jack.h> #include <sound/soc-dapm.h> -#include <asm/mach-types.h> #include "../codecs/sgtl5000.h" #include "mxs-saif.h" @@ -51,18 +50,27 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, } /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ - if (mclk < 8000000 || mclk > 27000000) + if (mclk < 8000000 || mclk > 27000000) { + dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); return -EINVAL; + } /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); - if (ret) + if (ret) { + dev_err(codec_dai->dev, "Failed to set sysclk to %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); return ret; + } /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */ ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0); - if (ret) + if (ret) { + dev_err(cpu_dai->dev, "Failed to set sysclk to %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); return ret; + } /* set codec to slave mode */ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | @@ -70,13 +78,19 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, dai_format); - if (ret) + if (ret) { + dev_err(codec_dai->dev, "Failed to set dai format to %08x\n", + dai_format); return ret; + } /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); - if (ret) + if (ret) { + dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n", + dai_format); return ret; + } return 0; } @@ -90,18 +104,14 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", - .codec_name = "sgtl5000.0-000a", - .cpu_dai_name = "mxs-saif.0", - .platform_name = "mxs-saif.0", .ops = &mxs_sgtl5000_hifi_ops, + .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", - .codec_name = "sgtl5000.0-000a", - .cpu_dai_name = "mxs-saif.1", - .platform_name = "mxs-saif.1", .ops = &mxs_sgtl5000_hifi_ops, + .capture_only = true, }, }; @@ -116,7 +126,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +152,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) @@ -160,8 +170,10 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) * should be >= 8MHz and <= 27M. */ ret = mxs_saif_get_mclk(0, 44100 * 256, 44100); - if (ret) + if (ret) { + dev_err(&pdev->dev, "failed to get mclk\n"); return ret; + } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index fe3285c..8987bf9 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -197,13 +197,12 @@ static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops nuc900_ac97_ops = { .read = nuc900_ac97_read, .write = nuc900_ac97_write, .reset = nuc900_ac97_cold_reset, .warm_reset = nuc900_ac97_warm_reset, -} -EXPORT_SYMBOL_GPL(soc_ac97_ops); +}; static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -326,64 +325,49 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (nuc900_ac97_data) return -EBUSY; - nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); + nuc900_audio = devm_kzalloc(&pdev->dev, sizeof(struct nuc900_audio), + GFP_KERNEL); if (!nuc900_audio) return -ENOMEM; spin_lock_init(&nuc900_audio->lock); nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!nuc900_audio->res) { - ret = -ENODEV; - goto out0; - } - - if (!request_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res), pdev->name)) { - ret = -EBUSY; - goto out0; - } - - nuc900_audio->mmio = ioremap(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); - if (!nuc900_audio->mmio) { - ret = -ENOMEM; - goto out1; - } + nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev, + nuc900_audio->res); + if (IS_ERR(nuc900_audio->mmio)) + return PTR_ERR(nuc900_audio->mmio); - nuc900_audio->clk = clk_get(&pdev->dev, NULL); + nuc900_audio->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(nuc900_audio->clk)) { ret = PTR_ERR(nuc900_audio->clk); - goto out2; + goto out; } nuc900_audio->irq_num = platform_get_irq(pdev, 0); if (!nuc900_audio->irq_num) { ret = -EBUSY; - goto out3; + goto out; } nuc900_ac97_data = nuc900_audio; + ret = snd_soc_set_ac97_ops(&nuc900_ac97_ops); + if (ret) + goto out; + ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) - goto out3; + goto out; /* enbale ac97 multifunction pin */ mfp_set_groupg(nuc900_audio->dev, NULL); return 0; -out3: - clk_put(nuc900_audio->clk); -out2: - iounmap(nuc900_audio->mmio); -out1: - release_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); -out0: - kfree(nuc900_audio); +out: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -391,13 +375,8 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); - clk_put(nuc900_ac97_data->clk); - iounmap(nuc900_ac97_data->mmio); - release_mem_region(nuc900_ac97_data->res->start, - resource_size(nuc900_ac97_data->res)); - - kfree(nuc900_ac97_data); nuc900_ac97_data = NULL; + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 60259f2..daa78a0 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,7 +1,7 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on ARCH_OMAP && DMA_OMAP - select SND_SOC_DMAENGINE_PCM + depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) + select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC tristate @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 @@ -87,7 +87,7 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 @@ -103,7 +103,7 @@ config SND_OMAP_SOC_OMAP_HDMI tristate "SoC Audio support for Texas Instruments OMAP HDMI" depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS select SND_OMAP_SOC_HDMI - select SND_SOC_OMAP_HDMI_CODEC + select SND_SOC_HDMI_CODEC select OMAP4_DSS_HDMI_AUDIO help Say Y if you want to add support for SoC HDMI audio on Texas Instruments diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 2b22594..a725905 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index eb68c7d..83433fd 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \ unsigned long val; \ int status; \ \ - status = strict_strtoul(buf, 0, &val); \ + status = kstrtoul(buf, 0, &val); \ if (status) \ return status; \ \ @@ -1012,28 +1012,33 @@ int omap_mcbsp_init(struct platform_device *pdev) } } - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); - if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; - } - /* RX DMA request number, and port address configuration */ - mcbsp->dma_req[1] = res->start; - mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); - mcbsp->dma_data[1].maxburst = 4; + if (!pdev->dev.of_node) { + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); - if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + } else { + mcbsp->dma_data[0].filter_data = "tx"; + mcbsp->dma_data[1].filter_data = "rx"; } - /* TX DMA request number, and port address configuration */ - mcbsp->dma_req[0] = res->start; - mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0); mcbsp->dma_data[0].maxburst = 4; + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); + mcbsp->dma_data[1].maxburst = 4; + mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7..ebb1390 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/mfd/twl6040.h> -#include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> #include <linux/of.h> @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 2ad0370..12e566b 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -57,7 +57,6 @@ struct omap_dmic { struct mutex mutex; struct snd_dmaengine_dai_dma_data dma_data; - unsigned int dma_req; }; static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) @@ -478,26 +477,15 @@ static int asoc_dmic_probe(struct platform_device *pdev) } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dmic->dev, "invalid dma resource\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->dma_req = res->start; - dmic->dma_data.filter_data = &dmic->dma_req; + dmic->dma_data.filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (!res) { - dev_err(dmic->dev, "invalid memory resource\n"); - ret = -ENODEV; + dmic->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dmic->io_base)) { + ret = PTR_ERR(dmic->io_base); goto err_put_clk; } - dmic->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dmic->io_base)) - return PTR_ERR(dmic->io_base); ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, &omap_dmic_dai, 1); diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c index d4eaa92..7e66e9c 100644 --- a/sound/soc/omap/omap-hdmi-card.c +++ b/sound/soc/omap/omap-hdmi-card.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = { .cpu_dai_name = "omap-hdmi-audio-dai", .platform_name = "omap-pcm-audio", .codec_name = "hdmi-audio-codec", - .codec_dai_name = "omap-hdmi-hifi", + .codec_dai_name = "hdmi-hifi", }; static struct snd_soc_card snd_soc_omap_hdmi = { diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index eadbfb6..6c19bba 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -433,6 +433,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* Sample rate generator drives the FS */ regs->srgr2 |= FSGM; break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP slave. FS clock as output */ + regs->srgr2 |= FSGM; + regs->pcr0 |= FSXM; + break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ break; @@ -814,8 +819,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) clk_put(mcbsp->fclk); - platform_set_drvdata(pdev, NULL); - return 0; } diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index eb05c7e..90d2a7c 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -66,7 +66,6 @@ struct omap_mcpdm { bool restart; struct snd_dmaengine_dai_dma_data dma_data[2]; - unsigned int dma_req[2]; }; /* @@ -477,24 +476,10 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA; mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); - if (!res) - return -ENODEV; - - mcpdm->dma_req[0] = res->start; - mcpdm->dma_data[0].filter_data = &mcpdm->dma_req[0]; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link"); - if (!res) - return -ENODEV; - - mcpdm->dma_req[1] = res->start; - mcpdm->dma_data[1].filter_data = &mcpdm->dma_req[1]; + mcpdm->dma_data[0].filter_data = "dn_link"; + mcpdm->dma_data[1].filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (res == NULL) - return -ENOMEM; - mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(mcpdm->io_base)) return PTR_ERR(mcpdm->io_base); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index c28e042..a11405d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -113,14 +113,25 @@ static int omap_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma_data; + int ret; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - return snd_dmaengine_pcm_open_request_chan(substream, - omap_dma_filter_fn, - dma_data->filter_data); + /* DT boot: filter_data is the DMA name */ + if (rtd->cpu_dai->dev->of_node) { + struct dma_chan *chan; + + chan = dma_request_slave_channel(rtd->cpu_dai->dev, + dma_data->filter_data); + ret = snd_dmaengine_pcm_open(substream, chan); + } else { + ret = snd_dmaengine_pcm_open_request_chan(substream, + omap_dma_filter_fn, + dma_data->filter_data); + } + return ret; } static int omap_pcm_mmap(struct snd_pcm_substream *substream, diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 249cd23..611179c 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -396,7 +396,7 @@ static int __init rx51_soc_init(void) { int err; - if (!machine_is_nokia_rx51()) + if (!machine_is_nokia_rx51() && !of_machine_is_compatible("nokia,omap3-n900")) return -ENODEV; err = gpio_request_one(RX51_TVOUT_SEL_GPIO, diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4d2e46f..4db74a0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,7 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM select SND_ARM help Say Y if you want to add support for codecs attached to @@ -130,26 +130,6 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. -config SND_SOC_SAARB - tristate "SoC Audio support for Marvell Saarb" - depends on SND_PXA2XX_SOC && MACH_SAARB - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - -config SND_SOC_TAVOREVB3 - tristate "SoC Audio support for Marvell Tavor EVB3" - depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - config SND_PXA910_SOC tristate "SoC Audio for Marvell PXA910 chip" depends on ARCH_MMP && SND diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index d8a265d..2cff67b 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o -snd-soc-saarb-objs := saarb.o -snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-hx4700-objs := hx4700.o snd-soc-magician-objs := magician.o @@ -48,8 +46,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o -obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o -obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 4ad7609..5b7d969 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -129,6 +129,7 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { /* audio machine driver */ static struct snd_soc_card brownstone = { .name = "brownstone", + .owner = THIS_MODULE, .dai_link = brownstone_wm8994_dai, .num_links = ARRAY_SIZE(brownstone_wm8994_dai), diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..bbea778 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -56,8 +56,6 @@ #include "pxa2xx-ac97.h" #include "../codecs/wm9713.h" -#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) - #define AC97_GPIO_PULL 0x58 /* Use GPIO8 for rear speaker amplifier */ @@ -133,10 +131,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 3499300..8235e23 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/platform_data/dma-mmp_tdma.h> #include <linux/platform_data/mmp_audio.h> + #include <sound/pxa2xx-lib.h> #include <sound/core.h> #include <sound/pcm.h> @@ -67,7 +68,7 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config slave_config; int ret; @@ -80,10 +81,10 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->dev_addr; + slave_config.dst_addr = dma_params->addr; slave_config.dst_maxburst = 4; } else { - slave_config.src_addr = dma_params->dev_addr; + slave_config.src_addr = dma_params->addr; slave_config.src_maxburst = 4; } @@ -147,7 +148,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream, vma->vm_end - vma->vm_start, vma->vm_page_prot); } -struct snd_pcm_ops mmp_pcm_ops = { +static struct snd_pcm_ops mmp_pcm_ops = { .open = mmp_pcm_open, .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, @@ -208,7 +209,7 @@ static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, return 0; } -int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm_substream *substream; struct snd_pcm *pcm = rtd->pcm; @@ -229,7 +230,7 @@ err: return ret; } -struct snd_soc_platform_driver mmp_soc_platform = { +static struct snd_soc_platform_driver mmp_soc_platform = { .ops = &mmp_pcm_ops, .pcm_new = mmp_pcm_new, .pcm_free = mmp_pcm_free_dma_buffers, diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index a647799..41752a5 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -27,12 +27,15 @@ #include <linux/slab.h> #include <linux/pxa2xx_ssp.h> #include <linux/io.h> +#include <linux/dmaengine.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include "mmp-sspa.h" /* @@ -40,7 +43,7 @@ */ struct sspa_priv { struct ssp_device *sspa; - struct pxa2xx_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_params; struct clk *audio_clk; struct clk *sysclk; int dai_fmt; @@ -266,7 +269,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); struct ssp_device *sspa = sspa_priv->sspa; - struct pxa2xx_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_params; u32 sspa_ctrl; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -309,7 +312,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, } dma_params = &sspa_priv->dma_params[substream->stream]; - dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? (sspa->phys_base + SSPA_TXD) : (sspa->phys_base + SSPA_RXD); snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params); @@ -388,7 +391,7 @@ static struct snd_soc_dai_ops mmp_sspa_dai_ops = { .set_fmt = mmp_sspa_set_dai_fmt, }; -struct snd_soc_dai_driver mmp_sspa_dai = { +static struct snd_soc_dai_driver mmp_sspa_dai = { .probe = mmp_sspa_probe, .playback = { .channels_min = 1, @@ -425,14 +428,12 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) return -ENOMEM; priv->dma_params = devm_kzalloc(&pdev->dev, - 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL); + 2 * sizeof(struct snd_dmaengine_dai_dma_data), + GFP_KERNEL); if (priv->dma_params == NULL) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) - return -ENOMEM; - priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(priv->sspa->mmio_base)) return PTR_ERR(priv->sspa->mmio_base); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6f4dd75..a3119a0 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -21,6 +21,8 @@ #include <linux/clk.h> #include <linux/io.h> #include <linux/pxa2xx_ssp.h> +#include <linux/of.h> +#include <linux/dmaengine.h> #include <asm/irq.h> @@ -30,9 +32,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include <mach/hardware.h> -#include <mach/dma.h> #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" @@ -79,27 +81,13 @@ static void pxa_ssp_disable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -struct pxa2xx_pcm_dma_data { - struct pxa2xx_pcm_dma_params params; - char name[20]; -}; - static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, - int out, struct pxa2xx_pcm_dma_params *dma_data) + int out, struct snd_dmaengine_dai_dma_data *dma) { - struct pxa2xx_pcm_dma_data *dma; - - dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params); - - snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, - width4 ? "32-bit" : "16-bit", out ? "out" : "in"); - - dma->params.name = dma->name; - dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); - dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : - (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | - (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; - dma->params.dev_addr = ssp->phys_base + SSDR; + dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES : + DMA_SLAVE_BUSWIDTH_2_BYTES; + dma->maxburst = 16; + dma->addr = ssp->phys_base + SSDR; } static int pxa_ssp_startup(struct snd_pcm_substream *substream, @@ -107,7 +95,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; - struct pxa2xx_pcm_dma_data *dma; + struct snd_dmaengine_dai_dma_data *dma; int ret = 0; if (!cpu_dai->active) { @@ -115,10 +103,14 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params); + + dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + &ssp->drcmr_tx : &ssp->drcmr_rx; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma); return ret; } @@ -559,7 +551,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -719,6 +711,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, static int pxa_ssp_probe(struct snd_soc_dai *dai) { + struct device *dev = dai->dev; struct ssp_priv *priv; int ret; @@ -726,10 +719,26 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (!priv) return -ENOMEM; - priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); - if (priv->ssp == NULL) { - ret = -ENODEV; - goto err_priv; + if (dev->of_node) { + struct device_node *ssp_handle; + + ssp_handle = of_parse_phandle(dev->of_node, "port", 0); + if (!ssp_handle) { + dev_err(dev, "unable to get 'port' phandle\n"); + return -ENODEV; + } + + priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + } else { + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } } priv->dai_fmt = (unsigned int) -1; @@ -798,6 +807,12 @@ static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", }; +#ifdef CONFIG_OF +static const struct of_device_id pxa_ssp_of_ids[] = { + { .compatible = "mrvl,pxa-ssp-dai" }, +}; +#endif + static int asoc_ssp_probe(struct platform_device *pdev) { return snd_soc_register_component(&pdev->dev, &pxa_ssp_component, @@ -812,8 +827,9 @@ static int asoc_ssp_remove(struct platform_device *pdev) static struct platform_driver asoc_ssp_driver = { .driver = { - .name = "pxa-ssp-dai", - .owner = THIS_MODULE, + .name = "pxa-ssp-dai", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pxa_ssp_of_ids), }, .probe = asoc_ssp_probe, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 57ea8e6..f1059d9 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -14,15 +14,16 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/dmaengine.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include <mach/hardware.h> #include <mach/regs-ac97.h> -#include <mach/dma.h> #include <mach/audio.h> #include "pxa2xx-ac97.h" @@ -41,52 +42,51 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) pxa2xx_ac97_finish_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .read = pxa2xx_ac97_read, .write = pxa2xx_ac97_write, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { - .name = "AC97 PCM Stereo out", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(12), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST32 | DCMD_WIDTH4, + +static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { - .name = "AC97 PCM Stereo in", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(11), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -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 = &DRCMR(10), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, +static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { + .addr = __PREG(MODR), + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .maxburst = 16, + .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -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 = &DRCMR(9), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, +static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { + .addr = __PREG(MODR), + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .maxburst = 16, + .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -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 = &DRCMR(8), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, +static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { + .addr = __PREG(MCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .maxburst = 16, + .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; #ifdef CONFIG_PM @@ -120,7 +120,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dma_data = &pxa2xx_ac97_pcm_stereo_out; @@ -136,7 +136,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dma_data = &pxa2xx_ac97_pcm_aux_mono_out; @@ -239,11 +239,17 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } + ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); + if (ret != 0) + return ret; + /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -255,6 +261,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h index eda891e..a49c21b 100644 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ b/sound/soc/pxa/pxa2xx-ac97.h @@ -14,7 +14,4 @@ #define PXA2XX_DAI_AC97_AUX 1 #define PXA2XX_DAI_AC97_MIC 2 -/* platform data */ -extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; - #endif diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index f7ca716..d5340a0 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -23,9 +23,9 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include <mach/hardware.h> -#include <mach/dma.h> #include <mach/audio.h> #include "pxa2xx-i2s.h" @@ -82,20 +82,20 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { - .name = "I2S PCM Stereo out", - .dev_addr = __PREG(SADR), - .drcmr = &DRCMR(3), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; +static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { + .addr = __PREG(SADR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { - .name = "I2S PCM Stereo in", - .dev_addr = __PREG(SADR), - .drcmr = &DRCMR(2), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; +static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { + .addr = __PREG(SADR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, @@ -163,7 +163,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; BUG_ON(IS_ERR(clk_i2s)); clk_prepare_enable(clk_i2s); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index ecff116..806da27 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -12,10 +12,13 @@ #include <linux/dma-mapping.h> #include <linux/module.h> +#include <linux/dmaengine.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> #include "../../arm/pxa2xx-pcm.h" @@ -25,7 +28,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma; + struct snd_dmaengine_dai_dma_data *dma; int ret; dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -39,7 +42,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, * with different params */ if (prtd->params == NULL) { prtd->params = dma; - ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, + ret = pxa_request_dma("name", DMA_PRIO_LOW, pxa2xx_pcm_dma_irq, substream); if (ret < 0) return ret; @@ -47,7 +50,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, } else if (prtd->params != dma) { pxa_free_dma(prtd->dma_ch); prtd->params = dma; - ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, + ret = pxa_request_dma("name", DMA_PRIO_LOW, pxa2xx_pcm_dma_irq, substream); if (ret < 0) return ret; @@ -131,10 +134,18 @@ static int pxa2xx_soc_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id snd_soc_pxa_audio_match[] = { + { .compatible = "mrvl,pxa-pcm-audio" }, + { } +}; +#endif + static struct platform_driver pxa_pcm_driver = { .driver = { - .name = "pxa-pcm-audio", - .owner = THIS_MODULE, + .name = "pxa-pcm-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(snd_soc_pxa_audio_match), }, .probe = pxa2xx_soc_platform_probe, diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c deleted file mode 100644 index c34146b..0000000 --- a/sound/soc/pxa/saarb.c +++ /dev/null @@ -1,190 +0,0 @@ -/* - * saarb.c -- SoC audio for saarb - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang <haojian.zhuang@marvell.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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *saarb_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* saarb machine dapm widgets */ -static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* saarb machine audio map */ -static const struct snd_soc_dapm_route saarb_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int saarb_i2s_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->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - - return ret; -} - -static struct snd_soc_ops saarb_i2s_ops = { - .hw_params = saarb_i2s_hw_params, -}; - -static struct snd_soc_dai_link saarb_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = saarb_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &saarb_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_saarb = { - .name = "Saarb", - .owner = THIS_MODULE, - .dai_link = saarb_dai, - .num_links = ARRAY_SIZE(saarb_dai), - - .dapm_widgets = saarb_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), - .dapm_routes = saarb_audio_map, - .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), -}; - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init saarb_init(void) -{ - int ret; - - if (!machine_is_saarb()) - return -ENODEV; - saarb_snd_device = platform_device_alloc("soc-audio", -1); - if (!saarb_snd_device) - return -ENOMEM; - - platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); - - ret = platform_device_add(saarb_snd_device); - if (ret) - platform_device_put(saarb_snd_device); - - return ret; -} - -static void __exit saarb_exit(void) -{ - platform_device_unregister(saarb_snd_device); -} - -module_init(saarb_init); -module_exit(saarb_exit); - -MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c deleted file mode 100644 index 8b5ab8f..0000000 --- a/sound/soc/pxa/tavorevb3.c +++ /dev/null @@ -1,189 +0,0 @@ -/* - * tavorevb3.c -- SoC audio for Tavor EVB3 - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang <haojian.zhuang@marvell.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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *evb3_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* tavorevb3 machine dapm widgets */ -static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic 2", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* tavorevb3 machine audio map */ -static const struct snd_soc_dapm_route evb3_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int evb3_i2s_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->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - return ret; -} - -static struct snd_soc_ops evb3_i2s_ops = { - .hw_params = evb3_i2s_hw_params, -}; - -static struct snd_soc_dai_link evb3_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = evb3_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &evb3_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_evb3 = { - .name = "Tavor EVB3", - .owner = THIS_MODULE, - .dai_link = evb3_dai, - .num_links = ARRAY_SIZE(evb3_dai), - - .dapm_widgets = evb3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets), - .dapm_routes = evb3_audio_map, - .num_dapm_routes = ARRAY_SIZE(evb3_audio_map), -}; - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init tavorevb3_init(void) -{ - int ret; - - if (!machine_is_tavorevb3()) - return -ENODEV; - evb3_snd_device = platform_device_alloc("soc-audio", -1); - if (!evb3_snd_device) - return -ENOMEM; - - platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); - - ret = platform_device_add(evb3_snd_device); - if (ret) - platform_device_put(evb3_snd_device); - - return ret; -} - -static void __exit tavorevb3_exit(void) -{ - platform_device_unregister(evb3_snd_device); -} - -module_init(tavorevb3_init); -module_exit(tavorevb3_exit); - -MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index f4ea4f6..13c9ee0 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -122,6 +122,7 @@ static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { /* ttc/td audio machine driver */ static struct snd_soc_card ttc_dkb_card = { .name = "ttc-dkb-hifi", + .owner = THIS_MODULE, .dai_link = ttc_pm860x_hifi_dai, .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ceb6566..db8aadf 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -256,7 +256,6 @@ static struct snd_soc_card zylonite = { .resume_pre = &zylonite_resume_pre, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), - .owner = THIS_MODULE, }; static struct platform_device *zylonite_snd_ac97_device; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1358c7d..d0740a7 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -128,7 +128,9 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) substream->runtime && snd_pcm_running(substream)) { dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 58cfb1e..945e8ab 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -192,7 +192,7 @@ static struct snd_soc_card snd_soc_card_s6105 = { .num_links = 1, }; -static struct s6000_snd_platform_data __initdata s6105_snd_data = { +static struct s6000_snd_platform_data s6105_snd_data __initdata = { .wide = 0, .channel_in = 0, .channel_out = 1, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 475fb0d..2eea184 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG - depends on I2C=y && GENERIC_HARDIRQS + depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S @@ -151,7 +151,7 @@ config SND_SOC_SMARTQ config SND_SOC_GONI_AQUILA_WM8994 tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) - depends on I2C=y && GENERIC_HARDIRQS + depends on I2C=y select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -177,7 +177,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG - depends on I2C=y && GENERIC_HARDIRQS + depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_PCM diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index cb88ead..2acf987 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -214,13 +214,12 @@ static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops s3c_ac97_ops = { .read = s3c_ac97_read, .write = s3c_ac97_write, .warm_reset = s3c_ac97_warm_reset, .reset = s3c_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -405,23 +404,16 @@ static int s3c_ac97_probe(struct platform_device *pdev) return -ENXIO; } - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!irq_res) { dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); return -ENXIO; } - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(s3c_ac97.regs)) + return PTR_ERR(s3c_ac97.regs); s3c_ac97_pcm_out.channel = dmatx_res->start; s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; @@ -433,14 +425,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) init_completion(&s3c_ac97.done); mutex_init(&s3c_ac97.lock); - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); ret = -ENODEV; @@ -461,12 +446,18 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err4; } + ret = snd_soc_set_ac97_ops(&s3c_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err4; + } + ret = snd_soc_register_component(&pdev->dev, &s3c_ac97_component, s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); if (ret) goto err5; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -480,20 +471,16 @@ err5: err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); - + snd_soc_set_ac97_ops(NULL); return ret; } static int s3c_ac97_remove(struct platform_device *pdev) { - struct resource *mem_res, *irq_res; + struct resource *irq_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); @@ -501,13 +488,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); - - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index ceed466..29e2468 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -350,8 +350,16 @@ static struct snd_soc_codec_conf bells_codec_conf[] = { }, }; +static struct snd_soc_dapm_widget bells_widgets[] = { + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + + { "DMIC", NULL, "MICBIAS2" }, + { "IN2L", NULL, "DMIC" }, + { "IN2R", NULL, "DMIC" }, }; static struct snd_soc_card bells_cards[] = { @@ -365,6 +373,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -383,6 +393,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -401,6 +413,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21b7926..9338d11 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream) dma_info.period = prtd->dma_period; dma_info.len = prtd->dma_period*limit; + if (dma_info.cap == DMA_CYCLIC) { + dma_info.buf = pos; + prtd->params->ops->prepare(prtd->params->ch, &dma_info); + prtd->dma_loaded += limit; + return; + } + while (prtd->dma_loaded < limit) { pr_debug("dma_loaded: %d\n", prtd->dma_loaded); @@ -176,6 +183,10 @@ static int dma_hw_params(struct snd_pcm_substream *substream, prtd->params->ch = prtd->params->ops->request( prtd->params->channel, &req, rtd->cpu_dai->dev, prtd->params->ch_name); + if (!prtd->params->ch) { + pr_err("Failed to allocate DMA channel\n"); + return -ENXIO; + } prtd->params->ops->config(prtd->params->ch, &config); } @@ -433,17 +444,17 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; -int asoc_dma_platform_register(struct device *dev) +int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_register); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); -void asoc_dma_platform_unregister(struct device *dev) +void samsung_asoc_dma_platform_unregister(struct device *dev) { snd_soc_unregister_platform(dev); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..0e86315 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,7 +22,7 @@ struct s3c_dma_params { char *ch_name; }; -int asoc_dma_platform_register(struct device *dev); -void asoc_dma_platform_unregister(struct device *dev); +int samsung_asoc_dma_platform_register(struct device *dev); +void samsung_asoc_dma_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index c0e6d9a..821a502 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -31,6 +31,10 @@ #define I2SLVL1ADDR 0x34 #define I2SLVL2ADDR 0x38 #define I2SLVL3ADDR 0x3c +#define I2SSTR1 0x40 +#define I2SVER 0x44 +#define I2SFIC2 0x48 +#define I2STDM 0x4c #define CON_RSTCLR (1 << 31) #define CON_FRXOFSTATUS (1 << 26) @@ -95,24 +99,39 @@ #define MOD_RXONLY (1 << 8) #define MOD_TXRX (2 << 8) #define MOD_MASK (3 << 8) -#define MOD_LR_LLOW (0 << 7) -#define MOD_LR_RLOW (1 << 7) -#define MOD_SDF_IIS (0 << 5) -#define MOD_SDF_MSB (1 << 5) -#define MOD_SDF_LSB (2 << 5) -#define MOD_SDF_MASK (3 << 5) -#define MOD_RCLK_256FS (0 << 3) -#define MOD_RCLK_512FS (1 << 3) -#define MOD_RCLK_384FS (2 << 3) -#define MOD_RCLK_768FS (3 << 3) -#define MOD_RCLK_MASK (3 << 3) -#define MOD_BCLK_32FS (0 << 1) -#define MOD_BCLK_48FS (1 << 1) -#define MOD_BCLK_16FS (2 << 1) -#define MOD_BCLK_24FS (3 << 1) -#define MOD_BCLK_MASK (3 << 1) +#define MOD_LRP_SHIFT 7 +#define MOD_LR_LLOW 0 +#define MOD_LR_RLOW 1 +#define MOD_SDF_SHIFT 5 +#define MOD_SDF_IIS 0 +#define MOD_SDF_MSB 1 +#define MOD_SDF_LSB 2 +#define MOD_SDF_MASK 3 +#define MOD_RCLK_SHIFT 3 +#define MOD_RCLK_256FS 0 +#define MOD_RCLK_512FS 1 +#define MOD_RCLK_384FS 2 +#define MOD_RCLK_768FS 3 +#define MOD_RCLK_MASK 3 +#define MOD_BCLK_SHIFT 1 +#define MOD_BCLK_32FS 0 +#define MOD_BCLK_48FS 1 +#define MOD_BCLK_16FS 2 +#define MOD_BCLK_24FS 3 +#define MOD_BCLK_MASK 3 #define MOD_8BIT (1 << 0) +#define EXYNOS5420_MOD_LRP_SHIFT 15 +#define EXYNOS5420_MOD_SDF_SHIFT 6 +#define EXYNOS5420_MOD_RCLK_SHIFT 4 +#define EXYNOS5420_MOD_BCLK_SHIFT 0 +#define EXYNOS5420_MOD_BCLK_64FS 4 +#define EXYNOS5420_MOD_BCLK_96FS 5 +#define EXYNOS5420_MOD_BCLK_128FS 6 +#define EXYNOS5420_MOD_BCLK_192FS 7 +#define EXYNOS5420_MOD_BCLK_256FS 8 +#define EXYNOS5420_MOD_BCLK_MASK 0xf + #define MOD_CDCLKCON (1 << 12) #define PSR_PSREN (1 << 15) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 82ebb1a..b302f3b 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -40,6 +40,7 @@ enum samsung_dai_type { struct samsung_i2s_dai_data { int dai_type; + u32 quirks; }; struct i2s_dai { @@ -198,7 +199,13 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { - u32 rfs = (readl(i2s->addr + I2SMOD) >> 3) & 0x3; + u32 rfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT); + rfs &= MOD_RCLK_MASK; switch (rfs) { case 3: return 768; @@ -212,21 +219,26 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { u32 mod = readl(i2s->addr + I2SMOD); + int rfs_shift; - mod &= ~MOD_RCLK_MASK; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs_shift = MOD_RCLK_SHIFT; + mod &= ~(MOD_RCLK_MASK << rfs_shift); switch (rfs) { case 768: - mod |= MOD_RCLK_768FS; + mod |= (MOD_RCLK_768FS << rfs_shift); break; case 512: - mod |= MOD_RCLK_512FS; + mod |= (MOD_RCLK_512FS << rfs_shift); break; case 384: - mod |= MOD_RCLK_384FS; + mod |= (MOD_RCLK_384FS << rfs_shift); break; default: - mod |= MOD_RCLK_256FS; + mod |= (MOD_RCLK_256FS << rfs_shift); break; } @@ -236,9 +248,22 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { - u32 bfs = (readl(i2s->addr + I2SMOD) >> 1) & 0x3; + u32 bfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT; + bfs &= EXYNOS5420_MOD_BCLK_MASK; + } else { + bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; + bfs &= MOD_BCLK_MASK; + } switch (bfs) { + case 8: return 256; + case 7: return 192; + case 6: return 128; + case 5: return 96; + case 4: return 64; case 3: return 24; case 2: return 16; case 1: return 48; @@ -250,21 +275,50 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { u32 mod = readl(i2s->addr + I2SMOD); + int bfs_shift; + int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; - mod &= ~MOD_BCLK_MASK; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT; + mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift); + } else { + bfs_shift = MOD_BCLK_SHIFT; + mod &= ~(MOD_BCLK_MASK << bfs_shift); + } + + /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ + if (!tdm && bfs > 48) { + dev_err(&i2s->pdev->dev, "Unsupported BCLK divider\n"); + return; + } switch (bfs) { case 48: - mod |= MOD_BCLK_48FS; + mod |= (MOD_BCLK_48FS << bfs_shift); break; case 32: - mod |= MOD_BCLK_32FS; + mod |= (MOD_BCLK_32FS << bfs_shift); break; case 24: - mod |= MOD_BCLK_24FS; + mod |= (MOD_BCLK_24FS << bfs_shift); break; case 16: - mod |= MOD_BCLK_16FS; + mod |= (MOD_BCLK_16FS << bfs_shift); + break; + case 64: + mod |= (EXYNOS5420_MOD_BCLK_64FS << bfs_shift); + break; + case 96: + mod |= (EXYNOS5420_MOD_BCLK_96FS << bfs_shift); + break; + case 128: + mod |= (EXYNOS5420_MOD_BCLK_128FS << bfs_shift); + break; + case 192: + mod |= (EXYNOS5420_MOD_BCLK_192FS << bfs_shift); + break; + case 256: + mod |= (EXYNOS5420_MOD_BCLK_256FS << bfs_shift); break; default: dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n"); @@ -491,20 +545,32 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, { struct i2s_dai *i2s = to_info(dai); u32 mod = readl(i2s->addr + I2SMOD); + int lrp_shift, sdf_shift, sdf_mask, lrp_rlow; u32 tmp = 0; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + lrp_shift = EXYNOS5420_MOD_LRP_SHIFT; + sdf_shift = EXYNOS5420_MOD_SDF_SHIFT; + } else { + lrp_shift = MOD_LRP_SHIFT; + sdf_shift = MOD_SDF_SHIFT; + } + + sdf_mask = MOD_SDF_MASK << sdf_shift; + lrp_rlow = MOD_LR_RLOW << lrp_shift; + /* Format is priority */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_MSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_MSB << sdf_shift); break; case SND_SOC_DAIFMT_LEFT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_LSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_LSB << sdf_shift); break; case SND_SOC_DAIFMT_I2S: - tmp |= MOD_SDF_IIS; + tmp |= (MOD_SDF_IIS << sdf_shift); break; default: dev_err(&i2s->pdev->dev, "Format not supported\n"); @@ -519,10 +585,10 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_NB_IF: - if (tmp & MOD_LR_RLOW) - tmp &= ~MOD_LR_RLOW; + if (tmp & lrp_rlow) + tmp &= ~lrp_rlow; else - tmp |= MOD_LR_RLOW; + tmp |= lrp_rlow; break; default: dev_err(&i2s->pdev->dev, "Polarity not supported\n"); @@ -544,15 +610,18 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } + /* + * Don't change the I2S mode if any controller is active on this + * channel. + */ if (any_active(i2s) && - ((mod & (MOD_SDF_MASK | MOD_LR_RLOW - | MOD_SLAVE)) != tmp)) { + ((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; } - mod &= ~(MOD_SDF_MASK | MOD_LR_RLOW | MOD_SLAVE); + mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE); mod |= tmp; writel(mod, i2s->addr + I2SMOD); @@ -742,13 +811,13 @@ static int config_setup(struct i2s_dai *i2s) return -EAGAIN; } - /* Don't bother RFS, BFS & PSR in Slave mode */ - if (is_slave(i2s)) - return 0; - set_bfs(i2s, bfs); set_rfs(i2s, rfs); + /* Don't bother with PSR in Slave mode */ + if (is_slave(i2s)) + return 0; + if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { psr = i2s->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); @@ -1007,6 +1076,8 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) if (IS_ERR(i2s->pdev)) return NULL; + i2s->pdev->dev.parent = &pdev->dev; + platform_set_drvdata(i2s->pdev, i2s); ret = platform_device_add(i2s->pdev); if (ret < 0) @@ -1016,66 +1087,20 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } -#ifdef CONFIG_OF -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) -{ - struct device *dev = &i2s->pdev->dev; - int index, gpio, ret; - - for (index = 0; index < 7; index++) { - gpio = of_get_gpio(dev->of_node, index); - if (!gpio_is_valid(gpio)) { - dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); - goto free_gpio; - } - - ret = gpio_request(gpio, dev_name(dev)); - if (ret) { - dev_err(dev, "gpio [%d] request failed\n", gpio); - goto free_gpio; - } - i2s->gpios[index] = gpio; - } - return 0; - -free_gpio: - while (--index >= 0) - gpio_free(i2s->gpios[index]); - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) -{ - unsigned int index; - for (index = 0; index < 7; index++) - gpio_free(i2s->gpios[index]); -} -#else -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) -{ - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) -{ -} - -#endif - static const struct of_device_id exynos_i2s_match[]; -static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) +static inline const struct samsung_i2s_dai_data *samsung_i2s_get_driver_data( + struct platform_device *pdev) { #ifdef CONFIG_OF - struct samsung_i2s_dai_data *data; if (pdev->dev.of_node) { const struct of_device_id *match; match = of_match_node(exynos_i2s_match, pdev->dev.of_node); - data = (struct samsung_i2s_dai_data *) match->data; - return data->dai_type; + return match->data; } else #endif - return platform_get_device_id(pdev)->driver_data; + return (struct samsung_i2s_dai_data *) + platform_get_device_id(pdev)->driver_data; } #ifdef CONFIG_PM_RUNTIME @@ -1106,13 +1131,13 @@ static int samsung_i2s_probe(struct platform_device *pdev) struct resource *res; u32 regs_base, quirks = 0, idma_addr = 0; struct device_node *np = pdev->dev.of_node; - enum samsung_dai_type samsung_dai_type; + const struct samsung_i2s_dai_data *i2s_dai_data; int ret = 0; /* Call during Seconday interface registration */ - samsung_dai_type = samsung_i2s_get_driver_data(pdev); + i2s_dai_data = samsung_i2s_get_driver_data(pdev); - if (samsung_dai_type == TYPE_SEC) { + if (i2s_dai_data->dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); if (!sec_dai) { dev_err(&pdev->dev, "Unable to get drvdata\n"); @@ -1121,7 +1146,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_component(&sec_dai->pdev->dev, &samsung_i2s_component, &sec_dai->i2s_dai_drv, 1); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1161,15 +1186,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) idma_addr = i2s_cfg->idma_addr; } } else { - if (of_find_property(np, "samsung,supports-6ch", NULL)) - quirks |= QUIRK_PRI_6CHAN; - - if (of_find_property(np, "samsung,supports-secdai", NULL)) - quirks |= QUIRK_SEC_DAI; - - if (of_find_property(np, "samsung,supports-rstclr", NULL)) - quirks |= QUIRK_NEED_RSTCLR; - + quirks = i2s_dai_data->quirks; if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { if (quirks & QUIRK_SEC_DAI) { @@ -1235,18 +1252,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->sec_dai = sec_dai; } - if (np) { - if (samsung_i2s_parse_dt_gpio(pri_dai)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } - } else { - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } + if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; } snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, @@ -1254,7 +1263,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; err: @@ -1267,14 +1276,10 @@ static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; struct resource *res; - struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; - if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) - samsung_i2s_dt_gpio_free(i2s->pri_dai); - if (other) { other->pri_dai = NULL; other->sec_dai = NULL; @@ -1288,33 +1293,59 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } +static const struct samsung_i2s_dai_data i2sv3_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_NO_MUXPSR, +}; + +static const struct samsung_i2s_dai_data i2sv5_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, +}; + +static const struct samsung_i2s_dai_data i2sv6_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | + QUIRK_SUPPORTS_TDM, +}; + +static const struct samsung_i2s_dai_data samsung_dai_type_pri = { + .dai_type = TYPE_PRI, +}; + +static const struct samsung_i2s_dai_data samsung_dai_type_sec = { + .dai_type = TYPE_SEC, +}; + static struct platform_device_id samsung_i2s_driver_ids[] = { { .name = "samsung-i2s", - .driver_data = TYPE_PRI, + .driver_data = (kernel_ulong_t)&samsung_dai_type_pri, }, { .name = "samsung-i2s-sec", - .driver_data = TYPE_SEC, + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, }, {}, }; MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); #ifdef CONFIG_OF -static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { - [TYPE_PRI] = { TYPE_PRI }, - [TYPE_SEC] = { TYPE_SEC }, -}; - static const struct of_device_id exynos_i2s_match[] = { - { .compatible = "samsung,i2s-v5", - .data = &samsung_i2s_dai_data_array[TYPE_PRI], + { + .compatible = "samsung,s3c6410-i2s", + .data = &i2sv3_dai_type, + }, { + .compatible = "samsung,s5pv210-i2s", + .data = &i2sv5_dai_type, + }, { + .compatible = "samsung,exynos5420-i2s", + .data = &i2sv6_dai_type, }, {}, }; diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 6e5fed3..ce1e1e1 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -257,7 +257,6 @@ static int idma_mmap(struct snd_pcm_substream *substream, /* From snd_pcm_lib_mmap_iomem */ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); - vma->vm_flags |= VM_IO; size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; ret = io_remap_pfn_range(vma, vma->vm_start, diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 1566afe..e54256f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -594,7 +594,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev) goto err5; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -623,7 +623,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev) struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); pm_runtime_disable(&pdev->dev); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 20e98d1..e5e81b1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -1,6 +1,4 @@ -/* sound/soc/samsung/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. +/* ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. * * Copyright (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 47e2386..ea885cb 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -176,7 +176,7 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the DMA: %d\n", ret); goto err; @@ -190,7 +190,7 @@ err: static int s3c2412_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 8b34145..9c8ebd8 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -480,7 +480,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the dma: %d\n", ret); goto err; @@ -494,7 +494,7 @@ err: static int s3c24xx_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index e43bd42..23a9204 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -176,7 +176,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 581ea4a..5fd7a05 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -11,6 +11,7 @@ #include <sound/pcm_params.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_device.h> /* * Default CFG switch settings to use this driver: @@ -37,11 +38,19 @@ /* SMDK has a 16.934MHZ crystal attached to WM8994 */ #define SMDK_WM8994_FREQ 16934000 +struct smdk_wm8994_data { + int mclk1_rate; +}; + +/* Default SMDKs */ +static struct smdk_wm8994_data smdk_board_data = { + .mclk1_rate = SMDK_WM8994_FREQ, +}; + static int smdk_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->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int pll_out; int ret; @@ -54,18 +63,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, else pll_out = params_rate(params) * 256; - 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; - - 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; - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, SMDK_WM8994_FREQ, pll_out); if (ret < 0) @@ -131,6 +128,8 @@ static struct snd_soc_dai_link smdk_dai[] = { .platform_name = "samsung-i2s.0", .codec_name = "wm8994-codec", .init = smdk_wm8994_init_paiftx, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &smdk_ops, }, { /* Sec_Fifo Playback i/f */ .name = "Sec_FIFO TX", @@ -139,6 +138,8 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-i2s-sec", .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &smdk_ops, }, }; @@ -150,15 +151,28 @@ static struct snd_soc_card smdk = { .num_links = ARRAY_SIZE(smdk_dai), }; +#ifdef CONFIG_OF +static const struct of_device_id samsung_wm8994_of_match[] = { + { .compatible = "samsung,smdk-wm8994", .data = &smdk_board_data }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); +#endif /* CONFIG_OF */ static int smdk_audio_probe(struct platform_device *pdev) { int ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &smdk; + struct smdk_wm8994_data *board; + const struct of_device_id *id; card->dev = &pdev->dev; + board = devm_kzalloc(&pdev->dev, sizeof(*board), GFP_KERNEL); + if (!board) + return -ENOMEM; + if (np) { smdk_dai[0].cpu_dai_name = NULL; smdk_dai[0].cpu_of_node = of_parse_phandle(np, @@ -173,6 +187,12 @@ static int smdk_audio_probe(struct platform_device *pdev) smdk_dai[0].platform_of_node = smdk_dai[0].cpu_of_node; } + id = of_match_device(samsung_wm8994_of_match, &pdev->dev); + if (id) + *board = *((struct smdk_wm8994_data *)id->data); + + platform_set_drvdata(pdev, board); + ret = snd_soc_register_card(card); if (ret) @@ -190,17 +210,9 @@ static int smdk_audio_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_OF -static const struct of_device_id samsung_wm8994_of_match[] = { - { .compatible = "samsung,smdk-wm8994", }, - {}, -}; -MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); -#endif /* CONFIG_OF */ - static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio", + .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, @@ -212,4 +224,4 @@ module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:smdk-audio"); +MODULE_ALIAS("platform:smdk-audio-wm8994"); diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 3688a32..0c84ca0 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -146,7 +146,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 2e5ebb2..28487dc 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -395,7 +395,7 @@ static int spdif_probe(struct platform_device *pdev) spin_lock_init(&spdif->lock); - spdif->pclk = clk_get(&pdev->dev, "spdif"); + spdif->pclk = devm_clk_get(&pdev->dev, "spdif"); if (IS_ERR(spdif->pclk)) { dev_err(&pdev->dev, "failed to get peri-clock\n"); ret = -ENOENT; @@ -403,7 +403,7 @@ static int spdif_probe(struct platform_device *pdev) } clk_prepare_enable(spdif->pclk); - spdif->sclk = clk_get(&pdev->dev, "sclk_spdif"); + spdif->sclk = devm_clk_get(&pdev->dev, "sclk_spdif"); if (IS_ERR(spdif->sclk)) { dev_err(&pdev->dev, "failed to get internal source clock\n"); ret = -ENOENT; @@ -442,7 +442,7 @@ static int spdif_probe(struct platform_device *pdev) spdif->dma_playback = &spdif_stereo_out; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to register DMA: %d\n", ret); goto err5; @@ -457,10 +457,8 @@ err3: release_mem_region(mem_res->start, resource_size(mem_res)); err2: clk_disable_unprepare(spdif->sclk); - clk_put(spdif->sclk); err1: clk_disable_unprepare(spdif->pclk); - clk_put(spdif->pclk); err0: return ret; } @@ -470,7 +468,7 @@ static int spdif_remove(struct platform_device *pdev) struct samsung_spdif_info *spdif = &spdif_info; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); iounmap(spdif->regs); @@ -480,9 +478,7 @@ static int spdif_remove(struct platform_device *pdev) release_mem_region(mem_res->start, resource_size(mem_res)); clk_disable_unprepare(spdif->sclk); - clk_put(spdif->sclk); clk_disable_unprepare(spdif->pclk); - clk_put(spdif->pclk); return 0; } diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 6bcb116..56d8ff6 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -34,6 +34,13 @@ config SND_SOC_SH4_SIU select SH_DMAE select FW_LOADER +config SND_SOC_RCAR + tristate "R-Car series SRU/SCU/SSIU/SSI support" + select SND_SIMPLE_CARD + select RCAR_CLK_ADG + help + This option enables R-Car SUR/SCU/SSIU/SSI sound support + ## ## Boards ## diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 849b387..aaf3dcd 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,6 +12,9 @@ obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o +## audio units for R-Car +obj-$(CONFIG_SND_SOC_RCAR) += rcar/ + ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-migor-objs := migor.o diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index f830c41..b33ca7c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -235,6 +235,8 @@ struct fsi_stream { struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int loop_cnt; + int additional_pos; }; struct fsi_clk { @@ -276,7 +278,7 @@ struct fsi_stream_handler { int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); - void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, + int (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, int enable); }; #define fsi_stream_handler_call(io, func, args...) \ @@ -1188,7 +1190,7 @@ static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) samples); } -static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int enable) { struct fsi_master *master = fsi_get_master(fsi); @@ -1201,6 +1203,8 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io) @@ -1287,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ + io->additional_pos = 0; io->dma = dma_map_single(dai->dev, runtime->dma_area, snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; @@ -1303,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) +static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) { struct snd_pcm_runtime *runtime = io->substream->runtime; + int period = io->period_pos + additional; - return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); + if (period >= runtime->periods) + period = 0; + + return io->dma + samples_to_bytes(runtime, period * io->period_samples); } static void fsi_dma_complete(void *data) @@ -1319,7 +1329,7 @@ static void fsi_dma_complete(void *data) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io), + dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0), samples_to_bytes(runtime, io->period_samples), dir); io->buff_sample_pos += io->period_samples; @@ -1345,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work) struct snd_pcm_runtime *runtime; enum dma_data_direction dir; int is_play = fsi_stream_is_play(fsi, io); - int len; + int len, i; dma_addr_t buf; if (!fsi_stream_is_working(fsi, io)) @@ -1355,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work) runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); - buf = fsi_dma_get_area(io); - dma_sync_single_for_device(dai->dev, buf, len, dir); + for (i = 0; i < io->loop_cnt; i++) { + buf = fsi_dma_get_area(io, io->additional_pos); - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } + dma_sync_single_for_device(dai->dev, buf, len, dir); - desc->callback = fsi_dma_complete; - desc->callback_param = io; + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } - if (dmaengine_submit(desc) < 0) { - dev_err(dai->dev, "tx_submit() fail\n"); - return; + desc->callback = fsi_dma_complete; + desc->callback_param = io; + + if (dmaengine_submit(desc) < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + return; + } + + dma_async_issue_pending(io->chan); + + io->additional_pos = 1; } - dma_async_issue_pending(io->chan); + io->loop_cnt = 1; /* * FIXME @@ -1409,7 +1426,7 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { struct fsi_master *master = fsi_get_master(fsi); @@ -1422,6 +1439,8 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index af19f77..0af2e4d 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -227,13 +227,12 @@ static void hac_ac97_coldrst(struct snd_ac97 *ac97) hac_ac97_warmrst(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops hac_ac97_ops = { .read = hac_ac97_read, .write = hac_ac97_write, .reset = hac_ac97_coldrst, .warm_reset = hac_ac97_warmrst, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -316,6 +315,10 @@ static const struct snd_soc_component_driver sh4_hac_component = { static int hac_soc_platform_probe(struct platform_device *pdev) { + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); + if (ret != 0) + return ret; + return snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } @@ -323,6 +326,7 @@ static int hac_soc_platform_probe(struct platform_device *pdev) static int hac_soc_platform_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile new file mode 100644 index 0000000..0ff492d --- /dev/null +++ b/sound/soc/sh/rcar/Makefile @@ -0,0 +1,2 @@ +snd-soc-rcar-objs := core.o gen.o scu.o adg.o ssi.o +obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o
\ No newline at end of file diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c new file mode 100644 index 0000000..d80deb7 --- /dev/null +++ b/sound/soc/sh/rcar/adg.c @@ -0,0 +1,234 @@ +/* + * Helper routines for R-Car sound ADG. + * + * Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ +#include <linux/sh_clk.h> +#include <mach/clock.h> +#include "rsnd.h" + +#define CLKA 0 +#define CLKB 1 +#define CLKC 2 +#define CLKI 3 +#define CLKMAX 4 + +struct rsnd_adg { + struct clk *clk[CLKMAX]; + + int rate_of_441khz_div_6; + int rate_of_48khz_div_6; +}; + +#define for_each_rsnd_clk(pos, adg, i) \ + for (i = 0, (pos) = adg->clk[i]; \ + i < CLKMAX; \ + i++, (pos) = adg->clk[i]) +#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) + +static enum rsnd_reg rsnd_adg_ssi_reg_get(int id) +{ + enum rsnd_reg reg; + + /* + * SSI 8 is not connected to ADG. + * it works with SSI 7 + */ + if (id == 8) + return RSND_REG_MAX; + + if (0 <= id && id <= 3) + reg = RSND_REG_AUDIO_CLK_SEL0; + else if (4 <= id && id <= 7) + reg = RSND_REG_AUDIO_CLK_SEL1; + else + reg = RSND_REG_AUDIO_CLK_SEL2; + + return reg; +} + +int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + enum rsnd_reg reg; + int id; + + /* + * "mod" = "ssi" here. + * we can get "ssi id" from mod + */ + id = rsnd_mod_id(mod); + reg = rsnd_adg_ssi_reg_get(id); + + rsnd_write(priv, mod, reg, 0); + + return 0; +} + +int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + enum rsnd_reg reg; + int id, shift, i; + u32 data; + int sel_table[] = { + [CLKA] = 0x1, + [CLKB] = 0x2, + [CLKC] = 0x3, + [CLKI] = 0x0, + }; + + dev_dbg(dev, "request clock = %d\n", rate); + + /* + * find suitable clock from + * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. + */ + data = 0; + for_each_rsnd_clk(clk, adg, i) { + if (rate == clk_get_rate(clk)) { + data = sel_table[i]; + goto found_clock; + } + } + + /* + * find 1/6 clock from BRGA/BRGB + */ + if (rate == adg->rate_of_441khz_div_6) { + data = 0x10; + goto found_clock; + } + + if (rate == adg->rate_of_48khz_div_6) { + data = 0x20; + goto found_clock; + } + + return -EIO; + +found_clock: + + /* + * This "mod" = "ssi" here. + * we can get "ssi id" from mod + */ + id = rsnd_mod_id(mod); + reg = rsnd_adg_ssi_reg_get(id); + + dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate); + + /* + * Enable SSIx clock + */ + shift = (id % 4) * 8; + + rsnd_bset(priv, mod, reg, + 0xFF << shift, + data << shift); + + return 0; +} + +static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) +{ + struct clk *clk; + unsigned long rate; + u32 ckr; + int i; + int brg_table[] = { + [CLKA] = 0x0, + [CLKB] = 0x1, + [CLKC] = 0x4, + [CLKI] = 0x2, + }; + + /* + * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC + * have 44.1kHz or 48kHz base clocks for now. + * + * SSI itself can divide parent clock by 1/1 - 1/16 + * So, BRGA outputs 44.1kHz base parent clock 1/32, + * and, BRGB outputs 48.0kHz base parent clock 1/32 here. + * see + * rsnd_adg_ssi_clk_try_start() + */ + ckr = 0; + adg->rate_of_441khz_div_6 = 0; + adg->rate_of_48khz_div_6 = 0; + for_each_rsnd_clk(clk, adg, i) { + rate = clk_get_rate(clk); + + if (0 == rate) /* not used */ + continue; + + /* RBGA */ + if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) { + adg->rate_of_441khz_div_6 = rate / 6; + ckr |= brg_table[i] << 20; + } + + /* RBGB */ + if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) { + adg->rate_of_48khz_div_6 = rate / 6; + ckr |= brg_table[i] << 16; + } + } + + rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr); + rsnd_priv_write(priv, BRRA, 0x00000002); /* 1/6 */ + rsnd_priv_write(priv, BRRB, 0x00000002); /* 1/6 */ +} + +int rsnd_adg_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct rsnd_adg *adg; + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + int i; + + adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); + if (!adg) { + dev_err(dev, "ADG allocate failed\n"); + return -ENOMEM; + } + + adg->clk[CLKA] = clk_get(NULL, "audio_clk_a"); + adg->clk[CLKB] = clk_get(NULL, "audio_clk_b"); + adg->clk[CLKC] = clk_get(NULL, "audio_clk_c"); + adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal"); + for_each_rsnd_clk(clk, adg, i) { + if (IS_ERR(clk)) { + dev_err(dev, "Audio clock failed\n"); + return -EIO; + } + } + + rsnd_adg_ssi_clk_init(priv, adg); + + priv->adg = adg; + + dev_dbg(dev, "adg probed\n"); + + return 0; +} + +void rsnd_adg_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_adg *adg = priv->adg; + struct clk *clk; + int i; + + for_each_rsnd_clk(clk, adg, i) + clk_put(clk); +} diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c new file mode 100644 index 0000000..a357060 --- /dev/null +++ b/sound/soc/sh/rcar/core.c @@ -0,0 +1,861 @@ +/* + * Renesas R-Car SRU/SCU/SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * Based on fsi.c + * Kuninori Morimoto <morimoto.kuninori@renesas.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. + */ + +/* + * Renesas R-Car sound device structure + * + * Gen1 + * + * SRU : Sound Routing Unit + * - SRC : Sampling Rate Converter + * - CMD + * - CTU : Channel Count Conversion Unit + * - MIX : Mixer + * - DVC : Digital Volume and Mute Function + * - SSI : Serial Sound Interface + * + * Gen2 + * + * SCU : Sampling Rate Converter Unit + * - SRC : Sampling Rate Converter + * - CMD + * - CTU : Channel Count Conversion Unit + * - MIX : Mixer + * - DVC : Digital Volume and Mute Function + * SSIU : Serial Sound Interface Unit + * - SSI : Serial Sound Interface + */ + +/* + * driver data Image + * + * rsnd_priv + * | + * | ** this depends on Gen1/Gen2 + * | + * +- gen + * | + * | ** these depend on data path + * | ** gen and platform data control it + * | + * +- rdai[0] + * | | sru ssiu ssi + * | +- playback -> [mod] -> [mod] -> [mod] -> ... + * | | + * | | sru ssiu ssi + * | +- capture -> [mod] -> [mod] -> [mod] -> ... + * | + * +- rdai[1] + * | | sru ssiu ssi + * | +- playback -> [mod] -> [mod] -> [mod] -> ... + * | | + * | | sru ssiu ssi + * | +- capture -> [mod] -> [mod] -> [mod] -> ... + * ... + * | + * | ** these control ssi + * | + * +- ssi + * | | + * | +- ssi[0] + * | +- ssi[1] + * | +- ssi[2] + * | ... + * | + * | ** these control scu + * | + * +- scu + * | + * +- scu[0] + * +- scu[1] + * +- scu[2] + * ... + * + * + * for_each_rsnd_dai(xx, priv, xx) + * rdai[0] => rdai[1] => rdai[2] => ... + * + * for_each_rsnd_mod(xx, rdai, xx) + * [mod] => [mod] => [mod] => ... + * + * rsnd_dai_call(xxx, fn ) + * [mod]->fn() -> [mod]->fn() -> [mod]->fn()... + * + */ +#include <linux/pm_runtime.h> +#include "rsnd.h" + +#define RSND_RATES SNDRV_PCM_RATE_8000_96000 +#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/* + * rsnd_platform functions + */ +#define rsnd_platform_call(priv, dai, func, param...) \ + (!(priv->info->func) ? -ENODEV : \ + priv->info->func(param)) + + +/* + * basic function + */ +u32 rsnd_read(struct rsnd_priv *priv, + struct rsnd_mod *mod, enum rsnd_reg reg) +{ + void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); + + BUG_ON(!base); + + return ioread32(base); +} + +void rsnd_write(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) +{ + void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); + struct device *dev = rsnd_priv_to_dev(priv); + + BUG_ON(!base); + + dev_dbg(dev, "w %p : %08x\n", base, data); + + iowrite32(data, base); +} + +void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) +{ + void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); + struct device *dev = rsnd_priv_to_dev(priv); + u32 val; + + BUG_ON(!base); + + val = ioread32(base); + val &= ~mask; + val |= data & mask; + iowrite32(val, base); + + dev_dbg(dev, "s %p : %08x\n", base, val); +} + +/* + * rsnd_mod functions + */ +char *rsnd_mod_name(struct rsnd_mod *mod) +{ + if (!mod || !mod->ops) + return "unknown"; + + return mod->ops->name; +} + +void rsnd_mod_init(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_mod_ops *ops, + int id) +{ + mod->priv = priv; + mod->id = id; + mod->ops = ops; + INIT_LIST_HEAD(&mod->list); +} + +/* + * rsnd_dma functions + */ +static void rsnd_dma_continue(struct rsnd_dma *dma) +{ + /* push next A or B plane */ + dma->submit_loop = 1; + schedule_work(&dma->work); +} + +void rsnd_dma_start(struct rsnd_dma *dma) +{ + /* push both A and B plane*/ + dma->submit_loop = 2; + schedule_work(&dma->work); +} + +void rsnd_dma_stop(struct rsnd_dma *dma) +{ + dma->submit_loop = 0; + cancel_work_sync(&dma->work); + dmaengine_terminate_all(dma->chan); +} + +static void rsnd_dma_complete(void *data) +{ + struct rsnd_dma *dma = (struct rsnd_dma *)data; + struct rsnd_priv *priv = dma->priv; + unsigned long flags; + + rsnd_lock(priv, flags); + + dma->complete(dma); + + if (dma->submit_loop) + rsnd_dma_continue(dma); + + rsnd_unlock(priv, flags); +} + +static void rsnd_dma_do_work(struct work_struct *work) +{ + struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work); + struct rsnd_priv *priv = dma->priv; + struct device *dev = rsnd_priv_to_dev(priv); + struct dma_async_tx_descriptor *desc; + dma_addr_t buf; + size_t len; + int i; + + for (i = 0; i < dma->submit_loop; i++) { + + if (dma->inquiry(dma, &buf, &len) < 0) + return; + + desc = dmaengine_prep_slave_single( + dma->chan, buf, len, dma->dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } + + desc->callback = rsnd_dma_complete; + desc->callback_param = dma; + + if (dmaengine_submit(desc) < 0) { + dev_err(dev, "dmaengine_submit() fail\n"); + return; + } + + } + + dma_async_issue_pending(dma->chan); +} + +int rsnd_dma_available(struct rsnd_dma *dma) +{ + return !!dma->chan; +} + +static bool rsnd_dma_filter(struct dma_chan *chan, void *param) +{ + chan->private = param; + + return true; +} + +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, + int is_play, int id, + int (*inquiry)(struct rsnd_dma *dma, + dma_addr_t *buf, int *len), + int (*complete)(struct rsnd_dma *dma)) +{ + struct device *dev = rsnd_priv_to_dev(priv); + dma_cap_mask_t mask; + + if (dma->chan) { + dev_err(dev, "it already has dma channel\n"); + return -EIO; + } + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dma->slave.shdma_slave.slave_id = id; + + dma->chan = dma_request_channel(mask, rsnd_dma_filter, + &dma->slave.shdma_slave); + if (!dma->chan) { + dev_err(dev, "can't get dma channel\n"); + return -EIO; + } + + dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + dma->priv = priv; + dma->inquiry = inquiry; + dma->complete = complete; + INIT_WORK(&dma->work, rsnd_dma_do_work); + + return 0; +} + +void rsnd_dma_quit(struct rsnd_priv *priv, + struct rsnd_dma *dma) +{ + if (dma->chan) + dma_release_channel(dma->chan); + + dma->chan = NULL; +} + +/* + * rsnd_dai functions + */ +#define rsnd_dai_call(rdai, io, fn) \ +({ \ + struct rsnd_mod *mod, *n; \ + int ret = 0; \ + for_each_rsnd_mod(mod, n, io) { \ + ret = rsnd_mod_call(mod, fn, rdai, io); \ + if (ret < 0) \ + break; \ + } \ + ret; \ +}) + +int rsnd_dai_connect(struct rsnd_dai *rdai, + struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + if (!mod) { + dev_err(dev, "NULL mod\n"); + return -EIO; + } + + if (!list_empty(&mod->list)) { + dev_err(dev, "%s%d is not empty\n", + rsnd_mod_name(mod), + rsnd_mod_id(mod)); + return -EIO; + } + + list_add_tail(&mod->list, &io->head); + + return 0; +} + +int rsnd_dai_disconnect(struct rsnd_mod *mod) +{ + list_del_init(&mod->list); + + return 0; +} + +int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) +{ + int id = rdai - priv->rdai; + + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return -EINVAL; + + return id; +} + +struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) +{ + return priv->rdai + id; +} + +static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) +{ + struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai); + + return rsnd_dai_get(priv, dai->id); +} + +int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io) +{ + return &rdai->playback == io; +} + +/* + * rsnd_soc_dai functions + */ +int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional) +{ + struct snd_pcm_substream *substream = io->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + int pos = io->byte_pos + additional; + + pos %= (runtime->periods * io->byte_per_period); + + return pos; +} + +void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int byte) +{ + io->byte_pos += byte; + + if (io->byte_pos >= io->next_period_byte) { + struct snd_pcm_substream *substream = io->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + io->period_pos++; + io->next_period_byte += io->byte_per_period; + + if (io->period_pos >= runtime->periods) { + io->byte_pos = 0; + io->period_pos = 0; + io->next_period_byte = io->byte_per_period; + } + + snd_pcm_period_elapsed(substream); + } +} + +static int rsnd_dai_stream_init(struct rsnd_dai_stream *io, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (!list_empty(&io->head)) + return -EIO; + + INIT_LIST_HEAD(&io->head); + io->substream = substream; + io->byte_pos = 0; + io->period_pos = 0; + io->byte_per_period = runtime->period_size * + runtime->channels * + samples_to_bytes(runtime, 1); + io->next_period_byte = io->byte_per_period; + + return 0; +} + +static +struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + return rtd->cpu_dai; +} + +static +struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai, + struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rdai->playback; + else + return &rdai->capture; +} + +static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + struct rsnd_mod *mod = rsnd_ssi_mod_get_frm_dai(priv, + rsnd_dai_id(priv, rdai), + rsnd_dai_is_play(rdai, io)); + int ssi_id = rsnd_mod_id(mod); + int ret; + unsigned long flags; + + rsnd_lock(priv, flags); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = rsnd_dai_stream_init(io, substream); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_platform_call(priv, dai, start, ssi_id); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_gen_path_init(priv, rdai, io); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_dai_call(rdai, io, init); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_dai_call(rdai, io, start); + if (ret < 0) + goto dai_trigger_end; + break; + case SNDRV_PCM_TRIGGER_STOP: + ret = rsnd_dai_call(rdai, io, stop); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_dai_call(rdai, io, quit); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_gen_path_exit(priv, rdai, io); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_platform_call(priv, dai, stop, ssi_id); + if (ret < 0) + goto dai_trigger_end; + break; + default: + ret = -EINVAL; + } + +dai_trigger_end: + rsnd_unlock(priv, flags); + + return ret; +} + +static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rdai->clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rdai->clk_master = 0; + break; + default: + return -EINVAL; + } + + /* set clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + rdai->bit_clk_inv = 0; + rdai->frm_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + rdai->bit_clk_inv = 1; + rdai->frm_clk_inv = 0; + break; + case SND_SOC_DAIFMT_IB_IF: + rdai->bit_clk_inv = 1; + rdai->frm_clk_inv = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + default: + rdai->bit_clk_inv = 0; + rdai->frm_clk_inv = 0; + break; + } + + /* set format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rdai->sys_delay = 0; + rdai->data_alignment = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + rdai->sys_delay = 1; + rdai->data_alignment = 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + rdai->sys_delay = 1; + rdai->data_alignment = 1; + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { + .trigger = rsnd_soc_dai_trigger, + .set_fmt = rsnd_soc_dai_set_fmt, +}; + +static int rsnd_dai_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct snd_soc_dai_driver *drv; + struct rsnd_dai *rdai; + struct rsnd_mod *pmod, *cmod; + struct device *dev = rsnd_priv_to_dev(priv); + int dai_nr; + int i; + + /* get max dai nr */ + for (dai_nr = 0; dai_nr < 32; dai_nr++) { + pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1); + cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0); + + if (!pmod && !cmod) + break; + } + + if (!dai_nr) { + dev_err(dev, "no dai\n"); + return -EIO; + } + + drv = devm_kzalloc(dev, sizeof(*drv) * dai_nr, GFP_KERNEL); + rdai = devm_kzalloc(dev, sizeof(*rdai) * dai_nr, GFP_KERNEL); + if (!drv || !rdai) { + dev_err(dev, "dai allocate failed\n"); + return -ENOMEM; + } + + for (i = 0; i < dai_nr; i++) { + + pmod = rsnd_ssi_mod_get_frm_dai(priv, i, 1); + cmod = rsnd_ssi_mod_get_frm_dai(priv, i, 0); + + /* + * init rsnd_dai + */ + INIT_LIST_HEAD(&rdai[i].playback.head); + INIT_LIST_HEAD(&rdai[i].capture.head); + + snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i); + + /* + * init snd_soc_dai_driver + */ + drv[i].name = rdai[i].name; + drv[i].ops = &rsnd_soc_dai_ops; + if (pmod) { + drv[i].playback.rates = RSND_RATES; + drv[i].playback.formats = RSND_FMTS; + drv[i].playback.channels_min = 2; + drv[i].playback.channels_max = 2; + } + if (cmod) { + drv[i].capture.rates = RSND_RATES; + drv[i].capture.formats = RSND_FMTS; + drv[i].capture.channels_min = 2; + drv[i].capture.channels_max = 2; + } + + dev_dbg(dev, "%s (%s/%s)\n", rdai[i].name, + pmod ? "play" : " -- ", + cmod ? "capture" : " -- "); + } + + priv->dai_nr = dai_nr; + priv->daidrv = drv; + priv->rdai = rdai; + + return 0; +} + +static void rsnd_dai_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ +} + +/* + * pcm ops + */ +static struct snd_pcm_hardware rsnd_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = RSND_FMTS, + .rates = RSND_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int rsnd_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int rsnd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_dai *dai = rsnd_substream_to_dai(substream); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + + return bytes_to_frames(runtime, io->byte_pos); +} + +static struct snd_pcm_ops rsnd_pcm_ops = { + .open = rsnd_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = rsnd_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .pointer = rsnd_pointer, +}; + +/* + * snd_soc_platform + */ + +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + return snd_pcm_lib_preallocate_pages_for_all( + rtd->pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +static void rsnd_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static struct snd_soc_platform_driver rsnd_soc_platform = { + .ops = &rsnd_pcm_ops, + .pcm_new = rsnd_pcm_new, + .pcm_free = rsnd_pcm_free, +}; + +static const struct snd_soc_component_driver rsnd_soc_component = { + .name = "rsnd", +}; + +/* + * rsnd probe + */ +static int rsnd_probe(struct platform_device *pdev) +{ + struct rcar_snd_info *info; + struct rsnd_priv *priv; + struct device *dev = &pdev->dev; + int ret; + + info = pdev->dev.platform_data; + if (!info) { + dev_err(dev, "driver needs R-Car sound information\n"); + return -ENODEV; + } + + /* + * init priv data + */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) { + dev_err(dev, "priv allocate failed\n"); + return -ENODEV; + } + + priv->dev = dev; + priv->info = info; + spin_lock_init(&priv->lock); + + /* + * init each module + */ + ret = rsnd_gen_probe(pdev, info, priv); + if (ret < 0) + return ret; + + ret = rsnd_scu_probe(pdev, info, priv); + if (ret < 0) + return ret; + + ret = rsnd_adg_probe(pdev, info, priv); + if (ret < 0) + return ret; + + ret = rsnd_ssi_probe(pdev, info, priv); + if (ret < 0) + return ret; + + ret = rsnd_dai_probe(pdev, info, priv); + if (ret < 0) + return ret; + + /* + * asoc register + */ + ret = snd_soc_register_platform(dev, &rsnd_soc_platform); + if (ret < 0) { + dev_err(dev, "cannot snd soc register\n"); + return ret; + } + + ret = snd_soc_register_component(dev, &rsnd_soc_component, + priv->daidrv, rsnd_dai_nr(priv)); + if (ret < 0) { + dev_err(dev, "cannot snd dai register\n"); + goto exit_snd_soc; + } + + dev_set_drvdata(dev, priv); + + pm_runtime_enable(dev); + + dev_info(dev, "probed\n"); + return ret; + +exit_snd_soc: + snd_soc_unregister_platform(dev); + + return ret; +} + +static int rsnd_remove(struct platform_device *pdev) +{ + struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + + /* + * remove each module + */ + rsnd_ssi_remove(pdev, priv); + rsnd_adg_remove(pdev, priv); + rsnd_scu_remove(pdev, priv); + rsnd_dai_remove(pdev, priv); + rsnd_gen_remove(pdev, priv); + + return 0; +} + +static struct platform_driver rsnd_driver = { + .driver = { + .name = "rcar_sound", + }, + .probe = rsnd_probe, + .remove = rsnd_remove, +}; +module_platform_driver(rsnd_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Renesas R-Car audio driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); +MODULE_ALIAS("platform:rcar-pcm-audio"); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c new file mode 100644 index 0000000..babb203 --- /dev/null +++ b/sound/soc/sh/rcar/gen.c @@ -0,0 +1,280 @@ +/* + * Renesas R-Car Gen1 SRU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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. + */ +#include "rsnd.h" + +struct rsnd_gen_ops { + int (*path_init)(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*path_exit)(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +}; + +struct rsnd_gen_reg_map { + int index; /* -1 : not supported */ + u32 offset_id; /* offset of ssi0, ssi1, ssi2... */ + u32 offset_adr; /* offset of SSICR, SSISR, ... */ +}; + +struct rsnd_gen { + void __iomem *base[RSND_BASE_MAX]; + + struct rsnd_gen_reg_map reg_map[RSND_REG_MAX]; + struct rsnd_gen_ops *ops; +}; + +#define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) + +/* + * Gen2 + * will be filled in the future + */ + +/* + * Gen1 + */ +static int rsnd_gen1_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *mod; + int ret; + int id; + + /* + * Gen1 is created by SRU/SSI, and this SRU is base module of + * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU) + * + * Easy image is.. + * Gen1 SRU = Gen2 SCU + SSIU + etc + * + * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is + * using fixed path. + * + * Then, SSI id = SCU id here + */ + + /* get SSI's ID */ + mod = rsnd_ssi_mod_get_frm_dai(priv, + rsnd_dai_id(priv, rdai), + rsnd_dai_is_play(rdai, io)); + id = rsnd_mod_id(mod); + + /* SSI */ + mod = rsnd_ssi_mod_get(priv, id); + ret = rsnd_dai_connect(rdai, mod, io); + if (ret < 0) + return ret; + + /* SCU */ + mod = rsnd_scu_mod_get(priv, id); + ret = rsnd_dai_connect(rdai, mod, io); + + return ret; +} + +static int rsnd_gen1_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *mod, *n; + int ret = 0; + + /* + * remove all mod from rdai + */ + for_each_rsnd_mod(mod, n, io) + ret |= rsnd_dai_disconnect(mod); + + return ret; +} + +static struct rsnd_gen_ops rsnd_gen1_ops = { + .path_init = rsnd_gen1_path_init, + .path_exit = rsnd_gen1_path_exit, +}; + +#define RSND_GEN1_REG_MAP(g, s, i, oi, oa) \ + do { \ + (g)->reg_map[RSND_REG_##i].index = RSND_GEN1_##s; \ + (g)->reg_map[RSND_REG_##i].offset_id = oi; \ + (g)->reg_map[RSND_REG_##i].offset_adr = oa; \ + } while (0) + +static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) +{ + RSND_GEN1_REG_MAP(gen, SRU, SRC_ROUTE_SEL, 0x0, 0x00); + RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL0, 0x0, 0x08); + RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL1, 0x0, 0x0c); + RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL2, 0x0, 0x10); + RSND_GEN1_REG_MAP(gen, SRU, SRC_CTRL, 0x0, 0xc0); + RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0); + RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4); + RSND_GEN1_REG_MAP(gen, SRU, BUSIF_MODE, 0x4, 0x20); + RSND_GEN1_REG_MAP(gen, SRU, BUSIF_ADINR, 0x40, 0x214); + + RSND_GEN1_REG_MAP(gen, ADG, BRRA, 0x0, 0x00); + RSND_GEN1_REG_MAP(gen, ADG, BRRB, 0x0, 0x04); + RSND_GEN1_REG_MAP(gen, ADG, SSICKR, 0x0, 0x08); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL0, 0x0, 0x0c); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL1, 0x0, 0x10); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL3, 0x0, 0x18); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL4, 0x0, 0x1c); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL5, 0x0, 0x20); + + RSND_GEN1_REG_MAP(gen, SSI, SSICR, 0x40, 0x00); + RSND_GEN1_REG_MAP(gen, SSI, SSISR, 0x40, 0x04); + RSND_GEN1_REG_MAP(gen, SSI, SSITDR, 0x40, 0x08); + RSND_GEN1_REG_MAP(gen, SSI, SSIRDR, 0x40, 0x0c); + RSND_GEN1_REG_MAP(gen, SSI, SSIWSR, 0x40, 0x20); +} + +static int rsnd_gen1_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + struct resource *sru_res; + struct resource *adg_res; + struct resource *ssi_res; + + /* + * map address + */ + sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); + adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG); + ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI); + + gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); + gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res); + gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res); + if (IS_ERR(gen->base[RSND_GEN1_SRU]) || + IS_ERR(gen->base[RSND_GEN1_ADG]) || + IS_ERR(gen->base[RSND_GEN1_SSI])) + return -ENODEV; + + gen->ops = &rsnd_gen1_ops; + rsnd_gen1_reg_map_init(gen); + + dev_dbg(dev, "Gen1 device probed\n"); + dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, + gen->base[RSND_GEN1_SRU]); + dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + gen->base[RSND_GEN1_ADG]); + dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + gen->base[RSND_GEN1_SSI]); + + return 0; + +} + +static void rsnd_gen1_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ +} + +/* + * Gen + */ +int rsnd_gen_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + return gen->ops->path_init(priv, rdai, io); +} + +int rsnd_gen_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + return gen->ops->path_exit(priv, rdai, io); +} + +void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int index; + u32 offset_id, offset_adr; + + if (reg >= RSND_REG_MAX) { + dev_err(dev, "rsnd_reg reg error\n"); + return NULL; + } + + index = gen->reg_map[reg].index; + offset_id = gen->reg_map[reg].offset_id; + offset_adr = gen->reg_map[reg].offset_adr; + + if (index < 0) { + dev_err(dev, "unsupported reg access %d\n", reg); + return NULL; + } + + if (offset_id && mod) + offset_id *= rsnd_mod_id(mod); + + /* + * index/offset were set on gen1/gen2 + */ + + return gen->base[index] + offset_id + offset_adr; +} + +int rsnd_gen_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen; + int i; + + gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); + if (!gen) { + dev_err(dev, "GEN allocate failed\n"); + return -ENOMEM; + } + + priv->gen = gen; + + /* + * see + * rsnd_reg_get() + * rsnd_gen_probe() + */ + for (i = 0; i < RSND_REG_MAX; i++) + gen->reg_map[i].index = -1; + + /* + * init each module + */ + if (rsnd_is_gen1(priv)) + return rsnd_gen1_probe(pdev, info, priv); + + dev_err(dev, "unknown generation R-Car sound device\n"); + + return -ENODEV; +} + +void rsnd_gen_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + if (rsnd_is_gen1(priv)) + rsnd_gen1_remove(pdev, priv); +} diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h new file mode 100644 index 0000000..9cc6986 --- /dev/null +++ b/sound/soc/sh/rcar/rsnd.h @@ -0,0 +1,302 @@ +/* + * Renesas R-Car + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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. + */ +#ifndef RSND_H +#define RSND_H + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/sh_dma.h> +#include <linux/workqueue.h> +#include <sound/rcar_snd.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +/* + * pseudo register + * + * The register address offsets SRU/SCU/SSIU on Gen1/Gen2 are very different. + * This driver uses pseudo register in order to hide it. + * see gen1/gen2 for detail + */ +enum rsnd_reg { + /* SRU/SCU */ + RSND_REG_SRC_ROUTE_SEL, + RSND_REG_SRC_TMG_SEL0, + RSND_REG_SRC_TMG_SEL1, + RSND_REG_SRC_TMG_SEL2, + RSND_REG_SRC_CTRL, + RSND_REG_SSI_MODE0, + RSND_REG_SSI_MODE1, + RSND_REG_BUSIF_MODE, + RSND_REG_BUSIF_ADINR, + + /* ADG */ + RSND_REG_BRRA, + RSND_REG_BRRB, + RSND_REG_SSICKR, + RSND_REG_AUDIO_CLK_SEL0, + RSND_REG_AUDIO_CLK_SEL1, + RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AUDIO_CLK_SEL3, + RSND_REG_AUDIO_CLK_SEL4, + RSND_REG_AUDIO_CLK_SEL5, + + /* SSI */ + RSND_REG_SSICR, + RSND_REG_SSISR, + RSND_REG_SSITDR, + RSND_REG_SSIRDR, + RSND_REG_SSIWSR, + + RSND_REG_MAX, +}; + +struct rsnd_priv; +struct rsnd_mod; +struct rsnd_dai; +struct rsnd_dai_stream; + +/* + * R-Car basic functions + */ +#define rsnd_mod_read(m, r) \ + rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r) +#define rsnd_mod_write(m, r, d) \ + rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) +#define rsnd_mod_bset(m, r, s, d) \ + rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) + +#define rsnd_priv_read(p, r) rsnd_read(p, NULL, RSND_REG_##r) +#define rsnd_priv_write(p, r, d) rsnd_write(p, NULL, RSND_REG_##r, d) +#define rsnd_priv_bset(p, r, s, d) rsnd_bset(p, NULL, RSND_REG_##r, s, d) + +u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); +void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data); +void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, + u32 mask, u32 data); + +/* + * R-Car DMA + */ +struct rsnd_dma { + struct rsnd_priv *priv; + struct sh_dmae_slave slave; + struct work_struct work; + struct dma_chan *chan; + enum dma_data_direction dir; + int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len); + int (*complete)(struct rsnd_dma *dma); + + int submit_loop; +}; + +void rsnd_dma_start(struct rsnd_dma *dma); +void rsnd_dma_stop(struct rsnd_dma *dma); +int rsnd_dma_available(struct rsnd_dma *dma); +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, + int is_play, int id, + int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len), + int (*complete)(struct rsnd_dma *dma)); +void rsnd_dma_quit(struct rsnd_priv *priv, + struct rsnd_dma *dma); + + +/* + * R-Car sound mod + */ + +struct rsnd_mod_ops { + char *name; + int (*init)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*quit)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*start)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*stop)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +}; + +struct rsnd_mod { + int id; + struct rsnd_priv *priv; + struct rsnd_mod_ops *ops; + struct list_head list; /* connect to rsnd_dai playback/capture */ + struct rsnd_dma dma; +}; + +#define rsnd_mod_to_priv(mod) ((mod)->priv) +#define rsnd_mod_to_dma(mod) (&(mod)->dma) +#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) +#define rsnd_mod_id(mod) ((mod)->id) +#define for_each_rsnd_mod(pos, n, io) \ + list_for_each_entry_safe(pos, n, &(io)->head, list) +#define rsnd_mod_call(mod, func, rdai, io) \ + (!(mod) ? -ENODEV : \ + !((mod)->ops->func) ? 0 : \ + (mod)->ops->func(mod, rdai, io)) + +void rsnd_mod_init(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_mod_ops *ops, + int id); +char *rsnd_mod_name(struct rsnd_mod *mod); + +/* + * R-Car sound DAI + */ +#define RSND_DAI_NAME_SIZE 16 +struct rsnd_dai_stream { + struct list_head head; /* head of rsnd_mod list */ + struct snd_pcm_substream *substream; + int byte_pos; + int period_pos; + int byte_per_period; + int next_period_byte; +}; + +struct rsnd_dai { + char name[RSND_DAI_NAME_SIZE]; + struct rsnd_dai_platform_info *info; /* rcar_snd.h */ + struct rsnd_dai_stream playback; + struct rsnd_dai_stream capture; + + int clk_master:1; + int bit_clk_inv:1; + int frm_clk_inv:1; + int sys_delay:1; + int data_alignment:1; +}; + +#define rsnd_dai_nr(priv) ((priv)->dai_nr) +#define for_each_rsnd_dai(rdai, priv, i) \ + for (i = 0, (rdai) = rsnd_dai_get(priv, i); \ + i < rsnd_dai_nr(priv); \ + i++, (rdai) = rsnd_dai_get(priv, i)) + +struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id); +int rsnd_dai_disconnect(struct rsnd_mod *mod); +int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod, + struct rsnd_dai_stream *io); +int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); +int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai); +#define rsnd_dai_get_platform_info(rdai) ((rdai)->info) +#define rsnd_io_to_runtime(io) ((io)->substream->runtime) + +void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt); +int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional); + +/* + * R-Car Gen1/Gen2 + */ +int rsnd_gen_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_gen_remove(struct platform_device *pdev, + struct rsnd_priv *priv); +int rsnd_gen_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +int rsnd_gen_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg); +#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) +#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) + +/* + * R-Car ADG + */ +int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod); +int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate); +int rsnd_adg_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_adg_remove(struct platform_device *pdev, + struct rsnd_priv *priv); + +/* + * R-Car sound priv + */ +struct rsnd_priv { + + struct device *dev; + struct rcar_snd_info *info; + spinlock_t lock; + + /* + * below value will be filled on rsnd_gen_probe() + */ + void *gen; + + /* + * below value will be filled on rsnd_scu_probe() + */ + void *scu; + int scu_nr; + + /* + * below value will be filled on rsnd_adg_probe() + */ + void *adg; + + /* + * below value will be filled on rsnd_ssi_probe() + */ + void *ssiu; + + /* + * below value will be filled on rsnd_dai_probe() + */ + struct snd_soc_dai_driver *daidrv; + struct rsnd_dai *rdai; + int dai_nr; +}; + +#define rsnd_priv_to_dev(priv) ((priv)->dev) +#define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags) +#define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags) + +/* + * R-Car SCU + */ +int rsnd_scu_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_scu_remove(struct platform_device *pdev, + struct rsnd_priv *priv); +struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); +#define rsnd_scu_nr(priv) ((priv)->scu_nr) + +/* + * R-Car SSI + */ +int rsnd_ssi_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_ssi_remove(struct platform_device *pdev, + struct rsnd_priv *priv); +struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); +struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv, + int dai_id, int is_play); + +#endif diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c new file mode 100644 index 0000000..2df2e91 --- /dev/null +++ b/sound/soc/sh/rcar/scu.c @@ -0,0 +1,236 @@ +/* + * Renesas R-Car SCU support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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. + */ +#include "rsnd.h" + +struct rsnd_scu { + struct rsnd_scu_platform_info *info; /* rcar_snd.h */ + struct rsnd_mod mod; +}; + +#define rsnd_scu_mode_flags(p) ((p)->info->flags) + +/* + * ADINR + */ +#define OTBL_24 (0 << 16) +#define OTBL_22 (2 << 16) +#define OTBL_20 (4 << 16) +#define OTBL_18 (6 << 16) +#define OTBL_16 (8 << 16) + + +#define rsnd_mod_to_scu(_mod) \ + container_of((_mod), struct rsnd_scu, mod) + +#define for_each_rsnd_scu(pos, priv, i) \ + for ((i) = 0; \ + ((i) < rsnd_scu_nr(priv)) && \ + ((pos) = (struct rsnd_scu *)(priv)->scu + i); \ + i++) + +static int rsnd_scu_set_route(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct scu_route_config { + u32 mask; + int shift; + } routes[] = { + { 0xF, 0, }, /* 0 */ + { 0xF, 4, }, /* 1 */ + { 0xF, 8, }, /* 2 */ + { 0x7, 12, }, /* 3 */ + { 0x7, 16, }, /* 4 */ + { 0x7, 20, }, /* 5 */ + { 0x7, 24, }, /* 6 */ + { 0x3, 28, }, /* 7 */ + { 0x3, 30, }, /* 8 */ + }; + + u32 mask; + u32 val; + int shift; + int id; + + /* + * Gen1 only + */ + if (!rsnd_is_gen1(priv)) + return 0; + + id = rsnd_mod_id(mod); + if (id < 0 || id > ARRAY_SIZE(routes)) + return -EIO; + + /* + * SRC_ROUTE_SELECT + */ + val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2; + val = val << routes[id].shift; + mask = routes[id].mask << routes[id].shift; + + rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val); + + /* + * SRC_TIMING_SELECT + */ + shift = (id % 4) * 8; + mask = 0x1F << shift; + if (8 == id) /* SRU8 is very special */ + val = id << shift; + else + val = (id + 1) << shift; + + switch (id / 4) { + case 0: + rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val); + break; + case 1: + rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val); + break; + case 2: + rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val); + break; + } + + return 0; +} + +static int rsnd_scu_set_mode(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + int id = rsnd_mod_id(mod); + u32 val; + + if (rsnd_is_gen1(priv)) { + val = (1 << id); + rsnd_mod_bset(mod, SRC_CTRL, val, val); + } + + return 0; +} + +static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 adinr = runtime->channels; + + switch (runtime->sample_bits) { + case 16: + adinr |= OTBL_16; + break; + case 32: + adinr |= OTBL_24; + break; + default: + return -EIO; + } + + rsnd_mod_write(mod, BUSIF_MODE, 1); + rsnd_mod_write(mod, BUSIF_ADINR, adinr); + + return 0; +} + +static int rsnd_scu_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 flags = rsnd_scu_mode_flags(scu); + int ret; + + /* + * SCU will be used if it has RSND_SCU_USE_HPBIF flags + */ + if (!(flags & RSND_SCU_USE_HPBIF)) { + /* it use PIO transter */ + dev_dbg(dev, "%s%d is not used\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; + } + + /* it use DMA transter */ + ret = rsnd_scu_set_route(priv, mod, rdai, io); + if (ret < 0) + return ret; + + ret = rsnd_scu_set_mode(priv, mod, rdai, io); + if (ret < 0) + return ret; + + ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); + if (ret < 0) + return ret; + + dev_dbg(dev, "%s%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static struct rsnd_mod_ops rsnd_scu_ops = { + .name = "scu", + .start = rsnd_scu_start, +}; + +struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id) +{ + BUG_ON(id < 0 || id >= rsnd_scu_nr(priv)); + + return &((struct rsnd_scu *)(priv->scu) + id)->mod; +} + +int rsnd_scu_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_scu *scu; + int i, nr; + + /* + * init SCU + */ + nr = info->scu_info_nr; + scu = devm_kzalloc(dev, sizeof(*scu) * nr, GFP_KERNEL); + if (!scu) { + dev_err(dev, "SCU allocate failed\n"); + return -ENOMEM; + } + + priv->scu_nr = nr; + priv->scu = scu; + + for_each_rsnd_scu(scu, priv, i) { + rsnd_mod_init(priv, &scu->mod, + &rsnd_scu_ops, i); + scu->info = &info->scu_info[i]; + + dev_dbg(dev, "SCU%d probed\n", i); + } + dev_dbg(dev, "scu probed\n"); + + return 0; +} + +void rsnd_scu_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ +} diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c new file mode 100644 index 0000000..fae26d3 --- /dev/null +++ b/sound/soc/sh/rcar/ssi.c @@ -0,0 +1,728 @@ +/* + * Renesas R-Car SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * Based on fsi.c + * Kuninori Morimoto <morimoto.kuninori@renesas.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. + */ +#include <linux/delay.h> +#include "rsnd.h" +#define RSND_SSI_NAME_SIZE 16 + +/* + * SSICR + */ +#define FORCE (1 << 31) /* Fixed */ +#define DMEN (1 << 28) /* DMA Enable */ +#define UIEN (1 << 27) /* Underflow Interrupt Enable */ +#define OIEN (1 << 26) /* Overflow Interrupt Enable */ +#define IIEN (1 << 25) /* Idle Mode Interrupt Enable */ +#define DIEN (1 << 24) /* Data Interrupt Enable */ + +#define DWL_8 (0 << 19) /* Data Word Length */ +#define DWL_16 (1 << 19) /* Data Word Length */ +#define DWL_18 (2 << 19) /* Data Word Length */ +#define DWL_20 (3 << 19) /* Data Word Length */ +#define DWL_22 (4 << 19) /* Data Word Length */ +#define DWL_24 (5 << 19) /* Data Word Length */ +#define DWL_32 (6 << 19) /* Data Word Length */ + +#define SWL_32 (3 << 16) /* R/W System Word Length */ +#define SCKD (1 << 15) /* Serial Bit Clock Direction */ +#define SWSD (1 << 14) /* Serial WS Direction */ +#define SCKP (1 << 13) /* Serial Bit Clock Polarity */ +#define SWSP (1 << 12) /* Serial WS Polarity */ +#define SDTA (1 << 10) /* Serial Data Alignment */ +#define DEL (1 << 8) /* Serial Data Delay */ +#define CKDV(v) (v << 4) /* Serial Clock Division Ratio */ +#define TRMD (1 << 1) /* Transmit/Receive Mode Select */ +#define EN (1 << 0) /* SSI Module Enable */ + +/* + * SSISR + */ +#define UIRQ (1 << 27) /* Underflow Error Interrupt Status */ +#define OIRQ (1 << 26) /* Overflow Error Interrupt Status */ +#define IIRQ (1 << 25) /* Idle Mode Interrupt Status */ +#define DIRQ (1 << 24) /* Data Interrupt Status Flag */ + +/* + * SSIWSR + */ +#define CONT (1 << 8) /* WS Continue Function */ + +struct rsnd_ssi { + struct clk *clk; + struct rsnd_ssi_platform_info *info; /* rcar_snd.h */ + struct rsnd_ssi *parent; + struct rsnd_mod mod; + + struct rsnd_dai *rdai; + struct rsnd_dai_stream *io; + u32 cr_own; + u32 cr_clk; + u32 cr_etc; + int err; + int dma_offset; + unsigned int usrcnt; + unsigned int rate; +}; + +struct rsnd_ssiu { + u32 ssi_mode0; + u32 ssi_mode1; + + int ssi_nr; + struct rsnd_ssi *ssi; +}; + +#define for_each_rsnd_ssi(pos, priv, i) \ + for (i = 0; \ + (i < rsnd_ssi_nr(priv)) && \ + ((pos) = ((struct rsnd_ssiu *)((priv)->ssiu))->ssi + i); \ + i++) + +#define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr) +#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) +#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma)) +#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0) +#define rsnd_ssi_dma_available(ssi) \ + rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod)) +#define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) +#define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master) +#define rsnd_ssi_mode_flags(p) ((p)->info->flags) +#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) +#define rsnd_ssi_to_ssiu(ssi)\ + (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1) + +static void rsnd_ssi_mode_init(struct rsnd_priv *priv, + struct rsnd_ssiu *ssiu) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_ssi *ssi; + u32 flags; + u32 val; + int i; + + /* + * SSI_MODE0 + */ + ssiu->ssi_mode0 = 0; + for_each_rsnd_ssi(ssi, priv, i) { + flags = rsnd_ssi_mode_flags(ssi); + + /* see also BUSIF_MODE */ + if (!(flags & RSND_SSI_DEPENDENT)) { + ssiu->ssi_mode0 |= (1 << i); + dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i); + } else { + dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i); + } + } + + /* + * SSI_MODE1 + */ +#define ssi_parent_set(p, sync, adg, ext) \ + do { \ + ssi->parent = ssiu->ssi + p; \ + if (flags & RSND_SSI_CLK_FROM_ADG) \ + val = adg; \ + else \ + val = ext; \ + if (flags & RSND_SSI_SYNC) \ + val |= sync; \ + } while (0) + + ssiu->ssi_mode1 = 0; + for_each_rsnd_ssi(ssi, priv, i) { + flags = rsnd_ssi_mode_flags(ssi); + + if (!(flags & RSND_SSI_CLK_PIN_SHARE)) + continue; + + val = 0; + switch (i) { + case 1: + ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0)); + break; + case 2: + ssi_parent_set(0, (1 << 4), (0x2 << 2), (0x1 << 2)); + break; + case 4: + ssi_parent_set(3, (1 << 20), (0x2 << 16), (0x1 << 16)); + break; + case 8: + ssi_parent_set(7, 0, 0, 0); + break; + } + + ssiu->ssi_mode1 |= val; + } +} + +static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi) +{ + struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi); + + rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0); + rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1); +} + +static void rsnd_ssi_status_check(struct rsnd_mod *mod, + u32 bit) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 status; + int i; + + for (i = 0; i < 1024; i++) { + status = rsnd_mod_read(mod, SSISR); + if (status & bit) + return; + + udelay(50); + } + + dev_warn(dev, "status check failed\n"); +} + +static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, + unsigned int rate) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct device *dev = rsnd_priv_to_dev(priv); + int i, j, ret; + int adg_clk_div_table[] = { + 1, 6, /* see adg.c */ + }; + int ssi_clk_mul_table[] = { + 1, 2, 4, 8, 16, 6, 12, + }; + unsigned int main_rate; + + /* + * Find best clock, and try to start ADG + */ + for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) { + for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { + + /* + * this driver is assuming that + * system word is 64fs (= 2 x 32bit) + * see rsnd_ssi_start() + */ + main_rate = rate / adg_clk_div_table[i] + * 32 * 2 * ssi_clk_mul_table[j]; + + ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate); + if (0 == ret) { + ssi->rate = rate; + ssi->cr_clk = FORCE | SWL_32 | + SCKD | SWSD | CKDV(j); + + dev_dbg(dev, "ssi%d outputs %u Hz\n", + rsnd_mod_id(&ssi->mod), rate); + + return 0; + } + } + } + + dev_err(dev, "unsupported clock rate\n"); + return -EIO; +} + +static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi) +{ + ssi->rate = 0; + ssi->cr_clk = 0; + rsnd_adg_ssi_clk_stop(&ssi->mod); +} + +static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 cr; + + if (0 == ssi->usrcnt) { + clk_enable(ssi->clk); + + if (rsnd_rdai_is_clk_master(rdai)) { + struct snd_pcm_runtime *runtime; + + runtime = rsnd_io_to_runtime(io); + + if (rsnd_ssi_clk_from_parent(ssi)) + rsnd_ssi_hw_start(ssi->parent, rdai, io); + else + rsnd_ssi_master_clk_start(ssi, runtime->rate); + } + } + + cr = ssi->cr_own | + ssi->cr_clk | + ssi->cr_etc | + EN; + + rsnd_mod_write(&ssi->mod, SSICR, cr); + + ssi->usrcnt++; + + dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod)); +} + +static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, + struct rsnd_dai *rdai) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 cr; + + if (0 == ssi->usrcnt) /* stop might be called without start */ + return; + + ssi->usrcnt--; + + if (0 == ssi->usrcnt) { + /* + * disable all IRQ, + * and, wait all data was sent + */ + cr = ssi->cr_own | + ssi->cr_clk; + + rsnd_mod_write(&ssi->mod, SSICR, cr | EN); + rsnd_ssi_status_check(&ssi->mod, DIRQ); + + /* + * disable SSI, + * and, wait idle state + */ + rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */ + rsnd_ssi_status_check(&ssi->mod, IIRQ); + + if (rsnd_rdai_is_clk_master(rdai)) { + if (rsnd_ssi_clk_from_parent(ssi)) + rsnd_ssi_hw_stop(ssi->parent, rdai); + else + rsnd_ssi_master_clk_stop(ssi); + } + + clk_disable(ssi->clk); + } + + dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); +} + +/* + * SSI mod common functions + */ +static int rsnd_ssi_init(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 cr; + + cr = FORCE; + + /* + * always use 32bit system word for easy clock calculation. + * see also rsnd_ssi_master_clk_enable() + */ + cr |= SWL_32; + + /* + * init clock settings for SSICR + */ + switch (runtime->sample_bits) { + case 16: + cr |= DWL_16; + break; + case 32: + cr |= DWL_24; + break; + default: + return -EIO; + } + + if (rdai->bit_clk_inv) + cr |= SCKP; + if (rdai->frm_clk_inv) + cr |= SWSP; + if (rdai->data_alignment) + cr |= SDTA; + if (rdai->sys_delay) + cr |= DEL; + if (rsnd_dai_is_play(rdai, io)) + cr |= TRMD; + + /* + * set ssi parameter + */ + ssi->rdai = rdai; + ssi->io = io; + ssi->cr_own = cr; + ssi->err = -1; /* ignore 1st error */ + + rsnd_ssi_mode_set(ssi); + + dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_ssi_quit(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + if (ssi->err > 0) + dev_warn(dev, "ssi under/over flow err = %d\n", ssi->err); + + ssi->rdai = NULL; + ssi->io = NULL; + ssi->cr_own = 0; + ssi->err = 0; + + return 0; +} + +static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status) +{ + /* under/over flow error */ + if (status & (UIRQ | OIRQ)) { + ssi->err++; + + /* clear error status */ + rsnd_mod_write(&ssi->mod, SSISR, 0); + } +} + +/* + * SSI PIO + */ +static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data) +{ + struct rsnd_ssi *ssi = data; + struct rsnd_dai_stream *io = ssi->io; + u32 status = rsnd_mod_read(&ssi->mod, SSISR); + irqreturn_t ret = IRQ_NONE; + + if (io && (status & DIRQ)) { + struct rsnd_dai *rdai = ssi->rdai; + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 *buf = (u32 *)(runtime->dma_area + + rsnd_dai_pointer_offset(io, 0)); + + rsnd_ssi_record_error(ssi, status); + + /* + * 8/16/32 data can be assesse to TDR/RDR register + * directly as 32bit data + * see rsnd_ssi_init() + */ + if (rsnd_dai_is_play(rdai, io)) + rsnd_mod_write(&ssi->mod, SSITDR, *buf); + else + *buf = rsnd_mod_read(&ssi->mod, SSIRDR); + + rsnd_dai_pointer_update(io, sizeof(*buf)); + + ret = IRQ_HANDLED; + } + + return ret; +} + +static int rsnd_ssi_pio_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + /* enable PIO IRQ */ + ssi->cr_etc = UIEN | OIEN | DIEN; + + rsnd_ssi_hw_start(ssi, rdai, io); + + dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + + dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + ssi->cr_etc = 0; + + rsnd_ssi_hw_stop(ssi, rdai); + + return 0; +} + +static struct rsnd_mod_ops rsnd_ssi_pio_ops = { + .name = "ssi (pio)", + .init = rsnd_ssi_init, + .quit = rsnd_ssi_quit, + .start = rsnd_ssi_pio_start, + .stop = rsnd_ssi_pio_stop, +}; + +static int rsnd_ssi_dma_inquiry(struct rsnd_dma *dma, dma_addr_t *buf, int *len) +{ + struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma); + struct rsnd_dai_stream *io = ssi->io; + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + *len = io->byte_per_period; + *buf = runtime->dma_addr + + rsnd_dai_pointer_offset(io, ssi->dma_offset + *len); + ssi->dma_offset = *len; /* it cares A/B plane */ + + return 0; +} + +static int rsnd_ssi_dma_complete(struct rsnd_dma *dma) +{ + struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma); + struct rsnd_dai_stream *io = ssi->io; + u32 status = rsnd_mod_read(&ssi->mod, SSISR); + + rsnd_ssi_record_error(ssi, status); + + rsnd_dai_pointer_update(ssi->io, io->byte_per_period); + + return 0; +} + +static int rsnd_ssi_dma_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); + + /* enable DMA transfer */ + ssi->cr_etc = DMEN; + ssi->dma_offset = 0; + + rsnd_dma_start(dma); + + rsnd_ssi_hw_start(ssi, ssi->rdai, io); + + /* enable WS continue */ + if (rsnd_rdai_is_clk_master(rdai)) + rsnd_mod_write(&ssi->mod, SSIWSR, CONT); + + return 0; +} + +static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); + + ssi->cr_etc = 0; + + rsnd_ssi_hw_stop(ssi, rdai); + + rsnd_dma_stop(dma); + + return 0; +} + +static struct rsnd_mod_ops rsnd_ssi_dma_ops = { + .name = "ssi (dma)", + .init = rsnd_ssi_init, + .quit = rsnd_ssi_quit, + .start = rsnd_ssi_dma_start, + .stop = rsnd_ssi_dma_stop, +}; + +/* + * Non SSI + */ +static int rsnd_ssi_non(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s\n", __func__); + + return 0; +} + +static struct rsnd_mod_ops rsnd_ssi_non_ops = { + .name = "ssi (non)", + .init = rsnd_ssi_non, + .quit = rsnd_ssi_non, + .start = rsnd_ssi_non, + .stop = rsnd_ssi_non, +}; + +/* + * ssi mod function + */ +struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv, + int dai_id, int is_play) +{ + struct rsnd_ssi *ssi; + int i, has_play; + + is_play = !!is_play; + + for_each_rsnd_ssi(ssi, priv, i) { + if (rsnd_ssi_dai_id(ssi) != dai_id) + continue; + + has_play = !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY); + + if (is_play == has_play) + return &ssi->mod; + } + + return NULL; +} + +struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) +{ + BUG_ON(id < 0 || id >= rsnd_ssi_nr(priv)); + + return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod; +} + +int rsnd_ssi_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct rsnd_ssi_platform_info *pinfo; + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_mod_ops *ops; + struct clk *clk; + struct rsnd_ssiu *ssiu; + struct rsnd_ssi *ssi; + char name[RSND_SSI_NAME_SIZE]; + int i, nr, ret; + + /* + * init SSI + */ + nr = info->ssi_info_nr; + ssiu = devm_kzalloc(dev, sizeof(*ssiu) + (sizeof(*ssi) * nr), + GFP_KERNEL); + if (!ssiu) { + dev_err(dev, "SSI allocate failed\n"); + return -ENOMEM; + } + + priv->ssiu = ssiu; + ssiu->ssi = (struct rsnd_ssi *)(ssiu + 1); + ssiu->ssi_nr = nr; + + for_each_rsnd_ssi(ssi, priv, i) { + pinfo = &info->ssi_info[i]; + + snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); + + clk = clk_get(dev, name); + if (IS_ERR(clk)) + return PTR_ERR(clk); + + ssi->info = pinfo; + ssi->clk = clk; + + ops = &rsnd_ssi_non_ops; + + /* + * SSI DMA case + */ + if (pinfo->dma_id > 0) { + ret = rsnd_dma_init( + priv, rsnd_mod_to_dma(&ssi->mod), + (rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY), + pinfo->dma_id, + rsnd_ssi_dma_inquiry, + rsnd_ssi_dma_complete); + if (ret < 0) + dev_info(dev, "SSI DMA failed. try PIO transter\n"); + else + ops = &rsnd_ssi_dma_ops; + + dev_dbg(dev, "SSI%d use DMA transfer\n", i); + } + + /* + * SSI PIO case + */ + if (!rsnd_ssi_dma_available(ssi) && + rsnd_ssi_pio_available(ssi)) { + ret = devm_request_irq(dev, pinfo->pio_irq, + &rsnd_ssi_pio_interrupt, + IRQF_SHARED, + dev_name(dev), ssi); + if (ret) { + dev_err(dev, "SSI request interrupt failed\n"); + return ret; + } + + ops = &rsnd_ssi_pio_ops; + + dev_dbg(dev, "SSI%d use PIO transfer\n", i); + } + + rsnd_mod_init(priv, &ssi->mod, ops, i); + } + + rsnd_ssi_mode_init(priv, ssiu); + + dev_dbg(dev, "ssi probed\n"); + + return 0; +} + +void rsnd_ssi_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_ssi *ssi; + int i; + + for_each_rsnd_ssi(ssi, priv, i) { + clk_put(ssi->clk); + if (rsnd_ssi_dma_available(ssi)) + rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); + } + +} diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 06a8000..53c9ecd 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,8 +149,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } else { rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ @@ -334,7 +335,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_set_metadata(struct snd_compr_stream *cstream, +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -347,7 +348,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_get_metadata(struct snd_compr_stream *cstream, +static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -364,8 +365,8 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, - .set_metadata = sst_compr_set_metadata, - .get_metadata = sst_compr_get_metadata, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d56bbea..4d05613 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -30,9 +30,12 @@ #include <linux/bitops.h> #include <linux/debugfs.h> #include <linux/platform_device.h> +#include <linux/pinctrl/consumer.h> #include <linux/ctype.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -47,8 +50,6 @@ #define NAME_SIZE 32 -static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); - #ifdef CONFIG_DEBUG_FS struct dentry *snd_soc_debugfs_root; EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); @@ -69,6 +70,16 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +struct snd_ac97_reset_cfg { + struct pinctrl *pctl; + struct pinctrl_state *pstate_reset; + struct pinctrl_state *pstate_warm_reset; + struct pinctrl_state *pstate_run; + int gpio_sdata; + int gpio_sync; + int gpio_reset; +}; + /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -192,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev, struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - ret = strict_strtol(buf, 10, &rtd->pmdown_time); + ret = kstrtol(buf, 10, &rtd->pmdown_time); if (ret) return ret; @@ -237,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file, char *start = buf; unsigned long reg, value; struct snd_soc_codec *codec = file->private_data; + int ret; buf_size = min(count, (sizeof(buf)-1)); if (copy_from_user(buf, user_buf, buf_size)) @@ -248,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file, reg = simple_strtoul(start, &start, 16); while (*start == ' ') start++; - if (strict_strtoul(start, 16, &value)) - return -EINVAL; + ret = kstrtoul(start, 16, &value); + if (ret) + return ret; /* Userspace has been fiddling around behind the kernel's back */ add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE); @@ -272,8 +285,8 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) codec->debugfs_codec_root = debugfs_create_dir(codec->name, debugfs_card_root); if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, "ASoC: Failed to create codec debugfs" - " directory\n"); + dev_warn(codec->dev, + "ASoC: Failed to create codec debugfs directory\n"); return; } @@ -286,8 +299,8 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) codec->debugfs_codec_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) - dev_warn(codec->dev, "ASoC: Failed to create codec register" - " debugfs file\n"); + dev_warn(codec->dev, + "ASoC: Failed to create codec register debugfs file\n"); snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); } @@ -530,6 +543,15 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif +static void codec2codec_close_delayed_work(struct work_struct *work) +{ + /* Currently nothing to do for c2c links + * Since c2c links are internal nodes in the DAPM graph and + * don't interface with the outside world or application layer + * we don't have to do any special handling on close. + */ +} + #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -631,8 +653,7 @@ int snd_soc_suspend(struct device *dev) */ if (codec->dapm.idle_bias_off) { dev_dbg(codec->dev, - "ASoC: idle_bias_off CODEC on" - " over suspend\n"); + "ASoC: idle_bias_off CODEC on over suspend\n"); break; } case SND_SOC_BIAS_OFF: @@ -643,8 +664,8 @@ int snd_soc_suspend(struct device *dev) regcache_mark_dirty(codec->control_data); break; default: - dev_dbg(codec->dev, "ASoC: CODEC is on" - " over suspend\n"); + dev_dbg(codec->dev, + "ASoC: CODEC is on over suspend\n"); break; } } @@ -713,8 +734,8 @@ static void soc_resume_deferred(struct work_struct *work) codec->suspended = 0; break; default: - dev_dbg(codec->dev, "ASoC: CODEC was on over" - " suspend\n"); + dev_dbg(codec->dev, + "ASoC: CODEC was on over suspend\n"); break; } } @@ -1110,8 +1131,8 @@ static int soc_probe_codec(struct snd_soc_card *card, } WARN(codec->dapm.idle_bias_off && codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias" - " with idle_bias_off==1\n", codec->name); + "codec %s can not start from non-off bias with idle_bias_off==1\n", + codec->name); } /* If the driver didn't set I/O up try regmap */ @@ -1224,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card, } rtd->card = card; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; codec->name_prefix = NULL; @@ -1429,6 +1447,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return ret; } } else { + INIT_DELAYED_WORK(&rtd->delayed_work, + codec2codec_close_delayed_work); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; @@ -1582,8 +1603,9 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, codec->compress_type = compress_type; ret = snd_soc_cache_init(codec); if (ret < 0) { - dev_err(codec->dev, "ASoC: Failed to set cache compression" - " type: %d\n", ret); + dev_err(codec->dev, + "ASoC: Failed to set cache compression type: %d\n", + ret); return ret; } codec->cache_init = 1; @@ -1639,8 +1661,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, card->owner, 0, &card->snd_card); if (ret < 0) { - dev_err(card->dev, "ASoC: can't create sound card for" - " card %s: %d\n", card->name, ret); + dev_err(card->dev, + "ASoC: can't create sound card for card %s: %d\n", + card->name, ret); goto base_error; } card->snd_card->dev = card->dev; @@ -1717,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); - snd_soc_dapm_new_widgets(&card->dapm); - for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; dai_fmt = dai_link->dai_fmt; @@ -1797,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - snd_soc_dapm_new_widgets(&card->dapm); - if (card->fully_routed) list_for_each_entry(codec, &card->codec_dev_list, card_list) snd_soc_dapm_auto_nc_codec_pins(codec); + snd_soc_dapm_new_widgets(card); + ret = snd_card_register(card->snd_card); if (ret < 0) { dev_err(card->dev, "ASoC: failed to register soundcard %d\n", @@ -1815,8 +1836,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_rtd; i++) { ret = soc_register_ac97_dai_link(&card->rtd[i]); if (ret < 0) { - dev_err(card->dev, "ASoC: failed to register AC97:" - " %d\n", ret); + dev_err(card->dev, + "ASoC: failed to register AC97: %d\n", ret); while (--i >= 0) soc_unregister_ac97_dai_link(card->rtd[i].codec); goto probe_aux_dev_err; @@ -2079,6 +2100,163 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +static struct snd_ac97_reset_cfg snd_ac97_rst_cfg; + +static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1); + + udelay(10); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); + msleep(2); +} + +static void snd_soc_ac97_reset(struct snd_ac97 *ac97) +{ + struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0); + gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0); + + udelay(10); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1); + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); + msleep(2); +} + +static int snd_soc_ac97_parse_pinctl(struct device *dev, + struct snd_ac97_reset_cfg *cfg) +{ + struct pinctrl *p; + struct pinctrl_state *state; + int gpio; + int ret; + + p = devm_pinctrl_get(dev); + if (IS_ERR(p)) { + dev_err(dev, "Failed to get pinctrl\n"); + return PTR_RET(p); + } + cfg->pctl = p; + + state = pinctrl_lookup_state(p, "ac97-reset"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-reset\n"); + return PTR_RET(state); + } + cfg->pstate_reset = state; + + state = pinctrl_lookup_state(p, "ac97-warm-reset"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n"); + return PTR_RET(state); + } + cfg->pstate_warm_reset = state; + + state = pinctrl_lookup_state(p, "ac97-running"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-running\n"); + return PTR_RET(state); + } + cfg->pstate_run = state; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-sync gpio\n"); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link sync"); + if (ret) { + dev_err(dev, "Failed requesting ac97-sync gpio\n"); + return ret; + } + cfg->gpio_sync = gpio; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link sdata"); + if (ret) { + dev_err(dev, "Failed requesting ac97-sdata gpio\n"); + return ret; + } + cfg->gpio_sdata = gpio; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-reset gpio\n"); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link reset"); + if (ret) { + dev_err(dev, "Failed requesting ac97-reset gpio\n"); + return ret; + } + cfg->gpio_reset = gpio; + + return 0; +} + +struct snd_ac97_bus_ops *soc_ac97_ops; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) +{ + if (ops == soc_ac97_ops) + return 0; + + if (soc_ac97_ops && ops) + return -EBUSY; + + soc_ac97_ops = ops; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); + +/** + * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * + * This function sets the reset and warm_reset properties of ops and parses + * the device node of pdev to get pinctrl states and gpio numbers to use. + */ +int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, + struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_ac97_reset_cfg cfg; + int ret; + + ret = snd_soc_ac97_parse_pinctl(dev, &cfg); + if (ret) + return ret; + + ret = snd_soc_set_ac97_ops(ops); + if (ret) + return ret; + + ops->warm_reset = snd_soc_ac97_warm_reset; + ops->reset = snd_soc_ac97_reset; + + snd_ac97_rst_cfg = cfg; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); + /** * snd_soc_free_ac97_codec - free AC97 codec device * @codec: audio codec @@ -2219,29 +2397,6 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, EXPORT_SYMBOL_GPL(snd_soc_test_bits); /** - * snd_soc_set_runtime_hwparams - set the runtime hardware parameters - * @substream: the pcm substream - * @hw: the hardware parameters - * - * Sets the substream runtime hardware parameters. - */ -int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, - const struct snd_pcm_hardware *hw) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.info = hw->info; - runtime->hw.formats = hw->formats; - runtime->hw.period_bytes_min = hw->period_bytes_min; - runtime->hw.period_bytes_max = hw->period_bytes_max; - runtime->hw.periods_min = hw->periods_min; - runtime->hw.periods_max = hw->periods_max; - runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; - runtime->hw.fifo_size = hw->fifo_size; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); - -/** * snd_soc_cnew - create new control * @_template: control template * @data: control private data @@ -2259,7 +2414,6 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, struct snd_kcontrol_new template; struct snd_kcontrol *kcontrol; char *name = NULL; - int name_len; memcpy(&template, _template, sizeof(template)); template.index = 0; @@ -2268,13 +2422,10 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, long_name = template.name; if (prefix) { - name_len = strlen(long_name) + strlen(prefix) + 2; - name = kmalloc(name_len, GFP_KERNEL); + name = kasprintf(GFP_KERNEL, "%s %s", prefix, long_name); if (!name) return NULL; - snprintf(name, name_len, "%s %s", prefix, long_name); - template.name = name; } else { template.name = long_name; @@ -2308,6 +2459,22 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, return 0; } +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + if (unlikely(!name)) + return NULL; + + list_for_each_entry(kctl, &card->controls, list) + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + return kctl; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); + /** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were @@ -2550,59 +2717,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); /** - * snd_soc_info_enum_ext - external enumerated single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about an external enumerated - * single mixer. - * - * Returns 0 for success. - */ -int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = e->max; - - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); - -/** - * snd_soc_info_volsw_ext - external single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single external mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int max = kcontrol->private_value; - - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); - -/** * snd_soc_info_volsw - single mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -3586,14 +3700,16 @@ int snd_soc_register_card(struct snd_soc_card *card) * not both or neither. */ if (!!link->codec_name == !!link->codec_of_node) { - dev_err(card->dev, "ASoC: Neither/both codec" - " name/of_node are set for %s\n", link->name); + dev_err(card->dev, + "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); return -EINVAL; } /* Codec DAI name must be specified */ if (!link->codec_dai_name) { - dev_err(card->dev, "ASoC: codec_dai_name not" - " set for %s\n", link->name); + dev_err(card->dev, + "ASoC: codec_dai_name not set for %s\n", + link->name); return -EINVAL; } @@ -3602,8 +3718,9 @@ int snd_soc_register_card(struct snd_soc_card *card) * can be left unspecified, and a dummy platform will be used. */ if (link->platform_name && link->platform_of_node) { - dev_err(card->dev, "ASoC: Both platform name/of_node" - " are set for %s\n", link->name); + dev_err(card->dev, + "ASoC: Both platform name/of_node are set for %s\n", + link->name); return -EINVAL; } @@ -3613,8 +3730,9 @@ int snd_soc_register_card(struct snd_soc_card *card) * name alone.. */ if (link->cpu_name && link->cpu_of_node) { - dev_err(card->dev, "ASoC: Neither/both " - "cpu name/of_node are set for %s\n",link->name); + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); return -EINVAL; } /* @@ -3623,8 +3741,9 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (!link->cpu_dai_name && !(link->cpu_name || link->cpu_of_node)) { - dev_err(card->dev, "ASoC: Neither cpu_dai_name nor " - "cpu_name/of_node are set for %s\n", link->name); + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); return -EINVAL; } } @@ -3728,8 +3847,9 @@ static inline char *fmt_multiple_name(struct device *dev, struct snd_soc_dai_driver *dai_drv) { if (dai_drv->name == NULL) { - dev_err(dev, "ASoC: error - multiple DAI %s registered with" - " no name\n", dev_name(dev)); + dev_err(dev, + "ASoC: error - multiple DAI %s registered with no name\n", + dev_name(dev)); return NULL; } @@ -3859,8 +3979,9 @@ static int snd_soc_register_dais(struct device *dev, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to " - "CODEC %s\n", dai->name, codec->name); + dev_dbg(dev, + "ASoC: Mapped DAI %s to CODEC %s\n", + dai->name, codec->name); dai->codec = codec; break; } @@ -3910,10 +4031,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) { - kfree(platform); + if (platform->name == NULL) return -ENOMEM; - } platform->dev = dev; platform->driver = platform_drv; @@ -4296,8 +4415,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, num_routes = of_property_count_strings(np, propname); if (num_routes < 0 || num_routes & 1) { - dev_err(card->dev, "ASoC: Property '%s' does not exist or its" - " length is not even\n", propname); + dev_err(card->dev, + "ASoC: Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c7051c4..c17c14c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -47,6 +47,15 @@ #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)); +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -64,6 +73,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_virt_mux] = 5, [snd_soc_dapm_value_mux] = 5, [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_switch] = 7, [snd_soc_dapm_mixer] = 7, [snd_soc_dapm_mixer_named_ctl] = 7, [snd_soc_dapm_pga] = 8, @@ -72,17 +82,20 @@ static int dapm_up_seq[] = { [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, [snd_soc_dapm_line] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_kcontrol] = 11, + [snd_soc_dapm_post] = 12, }; static int dapm_down_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_adc] = 1, - [snd_soc_dapm_hp] = 2, - [snd_soc_dapm_spk] = 2, - [snd_soc_dapm_line] = 2, - [snd_soc_dapm_out_drv] = 2, + [snd_soc_dapm_kcontrol] = 1, + [snd_soc_dapm_adc] = 2, + [snd_soc_dapm_hp] = 3, + [snd_soc_dapm_spk] = 3, + [snd_soc_dapm_line] = 3, + [snd_soc_dapm_out_drv] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, @@ -172,36 +185,178 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } -/* get snd_card from DAPM context */ -static inline struct snd_card *dapm_get_snd_card( - struct snd_soc_dapm_context *dapm) +struct dapm_kcontrol_data { + unsigned int value; + struct snd_soc_dapm_widget *widget; + struct list_head paths; + struct snd_soc_dapm_widget_list *wlist; +}; + +static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol) { - if (dapm->codec) - return dapm->codec->card->snd_card; - else if (dapm->platform) - return dapm->platform->card->snd_card; - else - BUG(); + struct dapm_kcontrol_data *data; + struct soc_mixer_control *mc; - /* unreachable */ - return NULL; + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(widget->dapm->dev, + "ASoC: can't allocate kcontrol data for %s\n", + widget->name); + return -ENOMEM; + } + + INIT_LIST_HEAD(&data->paths); + + switch (widget->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + mc = (struct soc_mixer_control *)kcontrol->private_value; + + if (mc->autodisable) { + struct snd_soc_dapm_widget template; + + memset(&template, 0, sizeof(template)); + template.reg = mc->reg; + template.mask = (1 << fls(mc->max)) - 1; + template.shift = mc->shift; + if (mc->invert) + template.off_val = mc->max; + else + template.off_val = 0; + template.on_val = template.off_val; + template.id = snd_soc_dapm_kcontrol; + template.name = kcontrol->id.name; + + data->value = template.on_val; + + data->widget = snd_soc_dapm_new_control(widget->dapm, + &template); + if (!data->widget) { + kfree(data); + return -ENOMEM; + } + } + break; + default: + break; + } + + kcontrol->private_data = data; + + return 0; } -/* get soc_card from DAPM context */ -static inline struct snd_soc_card *dapm_get_soc_card( - struct snd_soc_dapm_context *dapm) +static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { - if (dapm->codec) - return dapm->codec->card; - else if (dapm->platform) - return dapm->platform->card; + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data->widget); + kfree(data->wlist); + kfree(data); +} + +static struct snd_soc_dapm_widget_list *dapm_kcontrol_get_wlist( + const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + return data->wlist; +} + +static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, + struct snd_soc_dapm_widget *widget) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *new_wlist; + unsigned int n; + + if (data->wlist) + n = data->wlist->num_widgets + 1; else - BUG(); + n = 1; - /* unreachable */ - return NULL; + new_wlist = krealloc(data->wlist, + sizeof(*new_wlist) + sizeof(widget) * n, GFP_KERNEL); + if (!new_wlist) + return -ENOMEM; + + new_wlist->widgets[n - 1] = widget; + new_wlist->num_widgets = n; + + data->wlist = new_wlist; + + return 0; +} + +static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol, + struct snd_soc_dapm_path *path) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + list_add_tail(&path->list_kcontrol, &data->paths); + + if (data->widget) { + snd_soc_dapm_add_path(data->widget->dapm, data->widget, + path->source, NULL, NULL); + } +} + +static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (!data->widget) + return true; + + return data->widget->power; +} + +static struct list_head *dapm_kcontrol_get_path_list( + const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + return &data->paths; +} + +#define dapm_kcontrol_for_each_path(path, kcontrol) \ + list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ + list_kcontrol) + +static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + return data->value; } +static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (data->value == value) + return false; + + if (data->widget) + data->widget->on_val = value; + + data->value = value; + + return true; +} + +/** + * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol + * @kcontrol: The kcontrol + */ +struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) +{ + return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->codec; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); + static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; @@ -209,6 +364,7 @@ static void dapm_reset(struct snd_soc_card *card) memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); list_for_each_entry(w, &card->widgets, list) { + w->new_power = w->power; w->power_checked = false; w->inputs = -1; w->outputs = -1; @@ -365,11 +521,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & e->mask; - p->connect = 0; - for (i = 0; i < e->max; i++) { - if (!(strcmp(p->name, e->texts[i])) && item == i) - p->connect = 1; - } + if (item < e->max && !strcmp(p->name, e->texts[item])) + p->connect = 1; + else + p->connect = 0; } break; case snd_soc_dapm_virt_mux: { @@ -399,11 +554,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, break; } - p->connect = 0; - for (i = 0; i < e->max; i++) { - if (!(strcmp(p->name, e->texts[i])) && item == i) - p->connect = 1; - } + if (item < e->max && !strcmp(p->name, e->texts[item])) + p->connect = 1; + else + p->connect = 0; } break; /* does not affect routing - always connected */ @@ -428,6 +582,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_spk: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -512,7 +667,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, * create it. Either way, add the widget into the control's widget list */ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, - int kci, struct snd_soc_dapm_path *path) + int kci) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; @@ -520,11 +675,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, size_t prefix_len; int shared; struct snd_kcontrol *kcontrol; - struct snd_soc_dapm_widget_list *wlist; - int wlistentries; - size_t wlistsize; bool wname_in_long_name, kcname_in_long_name; - size_t name_len; char *long_name; const char *name; int ret; @@ -542,25 +693,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[kci], &kcontrol); - if (kcontrol) { - wlist = kcontrol->private_data; - wlistentries = wlist->num_widgets + 1; - } else { - wlist = NULL; - wlistentries = 1; - } - - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - wlistentries * sizeof(struct snd_soc_dapm_widget *); - wlist = krealloc(wlist, wlistsize, GFP_KERNEL); - if (wlist == NULL) { - dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n", - w->name); - return -ENOMEM; - } - wlist->num_widgets = wlistentries; - wlist->widgets[wlistentries - 1] = w; - if (!kcontrol) { if (shared) { wname_in_long_name = false; @@ -583,31 +715,22 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcname_in_long_name = false; break; default: - kfree(wlist); return -EINVAL; } } if (wname_in_long_name && kcname_in_long_name) { - name_len = strlen(w->name) - prefix_len + 1 + - strlen(w->kcontrol_news[kci].name) + 1; - - long_name = kmalloc(name_len, GFP_KERNEL); - if (long_name == NULL) { - kfree(wlist); - return -ENOMEM; - } - /* * The control will get a prefix from the control * creation process but we're also using the same * prefix for widgets so cut the prefix off the * front of the widget name. */ - snprintf(long_name, name_len, "%s %s", + long_name = kasprintf(GFP_KERNEL, "%s %s", w->name + prefix_len, w->kcontrol_news[kci].name); - long_name[name_len - 1] = '\0'; + if (long_name == NULL) + return -ENOMEM; name = long_name; } else if (wname_in_long_name) { @@ -618,24 +741,33 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, name = w->kcontrol_news[kci].name; } - kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name, + kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); + kfree(long_name); + if (!kcontrol) + return -ENOMEM; + kcontrol->private_free = dapm_kcontrol_free; + + ret = dapm_kcontrol_data_alloc(w, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + return ret; + } + ret = snd_ctl_add(card, kcontrol); if (ret < 0) { dev_err(dapm->dev, "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", w->name, name, ret); - kfree(wlist); - kfree(long_name); return ret; } - - path->long_name = long_name; } - kcontrol->private_data = wlist; + ret = dapm_kcontrol_add_widget(kcontrol, w); + if (ret) + return ret; + w->kcontrols[kci] = kcontrol; - path->kcontrol = kcontrol; return 0; } @@ -655,13 +787,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; if (w->kcontrols[i]) { - path->kcontrol = w->kcontrols[i]; + dapm_kcontrol_add_path(w->kcontrols[i], path); continue; } - ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, i); if (ret < 0) return ret; + + dapm_kcontrol_add_path(w->kcontrols[i], path); } } @@ -682,19 +816,17 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } - ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0); if (ret < 0) return ret; list_for_each_entry(path, &w->sources, list_sink) - path->kcontrol = w->kcontrols[0]; + dapm_kcontrol_add_path(w->kcontrols[0], path); return 0; } @@ -815,6 +947,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -910,6 +1043,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -1064,7 +1198,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, int ret; if (SND_SOC_DAPM_EVENT_ON(event)) { - if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, @@ -1074,7 +1208,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { - if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, @@ -1246,10 +1380,9 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, list_add_tail(&new_widget->power_list, list); } -static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, +static void dapm_seq_check_event(struct snd_soc_card *card, struct snd_soc_dapm_widget *w, int event) { - struct snd_soc_card *card = dapm->card; const char *ev_name; int power, ret; @@ -1270,60 +1403,63 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, ev_name = "POST_PMD"; power = 0; break; + case SND_SOC_DAPM_WILL_PMU: + ev_name = "WILL_PMU"; + power = 1; + break; + case SND_SOC_DAPM_WILL_PMD: + ev_name = "WILL_PMD"; + power = 0; + break; default: BUG(); return; } - if (w->power != power) + if (w->new_power != power) return; if (w->event && (w->event_flags & event)) { - pop_dbg(dapm->dev, card->pop_time, "pop test : %s %s\n", + pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n", w->name, ev_name); trace_snd_soc_dapm_widget_event_start(w, event); ret = w->event(w, NULL, event); trace_snd_soc_dapm_widget_event_done(w, event); if (ret < 0) - dev_err(dapm->dev, "ASoC: %s: %s event failed: %d\n", + dev_err(w->dapm->dev, "ASoC: %s: %s event failed: %d\n", ev_name, w->name, ret); } } /* Apply the coalesced changes from a DAPM sequence */ -static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, +static void dapm_seq_run_coalesced(struct snd_soc_card *card, struct list_head *pending) { - struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; - int reg, power; + int reg; unsigned int value = 0; unsigned int mask = 0; - unsigned int cur_mask; reg = list_first_entry(pending, struct snd_soc_dapm_widget, power_list)->reg; list_for_each_entry(w, pending, power_list) { - cur_mask = 1 << w->shift; BUG_ON(reg != w->reg); + w->power = w->new_power; - if (w->invert) - power = !w->power; + mask |= w->mask << w->shift; + if (w->power) + value |= w->on_val << w->shift; else - power = w->power; + value |= w->off_val << w->shift; - mask |= cur_mask; - if (power) - value |= cur_mask; - - pop_dbg(dapm->dev, card->pop_time, + pop_dbg(w->dapm->dev, card->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", w->name, reg, value, mask); /* Check for events */ - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMU); - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMD); + dapm_seq_check_event(card, w, SND_SOC_DAPM_PRE_PMU); + dapm_seq_check_event(card, w, SND_SOC_DAPM_PRE_PMD); } if (reg >= 0) { @@ -1333,7 +1469,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, w = list_first_entry(pending, struct snd_soc_dapm_widget, power_list); - pop_dbg(dapm->dev, card->pop_time, + pop_dbg(w->dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); @@ -1341,8 +1477,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, } list_for_each_entry(w, pending, power_list) { - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMU); - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMD); + dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); + dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); } } @@ -1354,8 +1490,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, * Currently anything that requires more than a single write is not * handled. */ -static void dapm_seq_run(struct snd_soc_dapm_context *dapm, - struct list_head *list, int event, bool power_up) +static void dapm_seq_run(struct snd_soc_card *card, + struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); @@ -1378,7 +1514,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, if (sort[w->id] != cur_sort || w->reg != cur_reg || w->dapm != cur_dapm || w->subseq != cur_subseq) { if (!list_empty(&pending)) - dapm_seq_run_coalesced(cur_dapm, &pending); + dapm_seq_run_coalesced(card, &pending); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) @@ -1438,7 +1574,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(cur_dapm, &pending); + dapm_seq_run_coalesced(card, &pending); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) @@ -1448,37 +1584,48 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } } -static void dapm_widget_update(struct snd_soc_dapm_context *dapm) +static void dapm_widget_update(struct snd_soc_card *card) { - struct snd_soc_dapm_update *update = dapm->update; - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_update *update = card->update; + struct snd_soc_dapm_widget_list *wlist; + struct snd_soc_dapm_widget *w = NULL; + unsigned int wi; int ret; - if (!update) + if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; - w = update->widget; + wlist = dapm_kcontrol_get_wlist(update->kcontrol); - if (w->event && - (w->event_flags & SND_SOC_DAPM_PRE_REG)) { - ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret != 0) - dev_err(dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n", - w->name, ret); + for (wi = 0; wi < wlist->num_widgets; wi++) { + w = wlist->widgets[wi]; + + if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) { + ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret != 0) + dev_err(w->dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n", + w->name, ret); + } } + if (!w) + return; + ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) - dev_err(dapm->dev, "ASoC: %s DAPM update failed: %d\n", + dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); - if (w->event && - (w->event_flags & SND_SOC_DAPM_POST_REG)) { - ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); - if (ret != 0) - dev_err(dapm->dev, "ASoC: %s DAPM post-event failed: %d\n", - w->name, ret); + for (wi = 0; wi < wlist->num_widgets; wi++) { + w = wlist->widgets[wi]; + + if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) { + ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); + if (ret != 0) + dev_err(w->dapm->dev, "ASoC: %s DAPM post-event failed: %d\n", + w->name, ret); + } } } @@ -1590,6 +1737,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1606,8 +1754,6 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, dapm_seq_insert(w, up_list, true); else dapm_seq_insert(w, down_list, false); - - w->power = power; } static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, @@ -1641,9 +1787,8 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, * o Input pin to Output pin (bypass, sidetone) * o DAC to ADC (loopback). */ -static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) +static int dapm_power_widgets(struct snd_soc_card *card, int event) { - struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; struct snd_soc_dapm_context *d; LIST_HEAD(up_list); @@ -1683,7 +1828,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; } - if (w->power) { + if (w->new_power) { d = w->dapm; /* Supplies and micbiases only bring the @@ -1725,21 +1870,29 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_walk_done(card); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &dapm->card->dapm_list, list) + list_for_each_entry(d, &card->dapm_list, list) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); async_synchronize_full_domain(&async_domain); + list_for_each_entry(w, &down_list, power_list) { + dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMD); + } + + list_for_each_entry(w, &up_list, power_list) { + dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMU); + } + /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(dapm, &down_list, event, false); + dapm_seq_run(card, &down_list, event, false); - dapm_widget_update(dapm); + dapm_widget_update(card); /* Now power up. */ - dapm_seq_run(dapm, &up_list, event, true); + dapm_seq_run(card, &up_list, event, true); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &dapm->card->dapm_list, list) + list_for_each_entry(d, &card->dapm_list, list) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); async_synchronize_full_domain(&async_domain); @@ -1750,7 +1903,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) d->stream_event(d, event); } - pop_dbg(dapm->dev, card->pop_time, + pop_dbg(card->dev, card->pop_time, "DAPM sequencing finished, waiting %dms\n", card->pop_time); pop_wait(card->pop_time); @@ -1785,8 +1938,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (w->reg >= 0) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " - R%d(0x%x) bit %d", - w->reg, w->reg, w->shift); + " - R%d(0x%x) mask 0x%x", + w->reg, w->reg, w->mask << w->shift); ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); @@ -1923,22 +2076,14 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; - if (widget->id != snd_soc_dapm_mux && - widget->id != snd_soc_dapm_virt_mux && - widget->id != snd_soc_dapm_value_mux) - return -ENODEV; - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { - if (path->kcontrol != kcontrol) - continue; - + dapm_kcontrol_for_each_path(path, kcontrol) { if (!path->name || !e->texts[mux]) continue; @@ -1953,73 +2098,68 @@ static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, "mux disconnection"); path->connect = 0; /* old connection must be powered down */ } + dapm_mark_dirty(path->sink, "mux change"); } - if (found) { - dapm_mark_dirty(widget, "mux change"); - dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); - } + if (found) + dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); return found; } -int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e, + struct snd_soc_dapm_update *update) { - struct snd_soc_card *card = widget->dapm->card; + struct snd_soc_card *card = dapm->card; int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); + card->update = update; + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(widget); + soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mixer_update_power(struct snd_soc_card *card, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; int found = 0; - if (widget->id != snd_soc_dapm_mixer && - widget->id != snd_soc_dapm_mixer_named_ctl && - widget->id != snd_soc_dapm_switch) - return -ENODEV; - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { - if (path->kcontrol != kcontrol) - continue; - - /* found, now check type */ + dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; path->connect = connect; dapm_mark_dirty(path->source, "mixer connection"); + dapm_mark_dirty(path->sink, "mixer update"); } - if (found) { - dapm_mark_dirty(widget, "mixer update"); - dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); - } + if (found) + dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); return found; } -int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int connect) +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, + struct snd_kcontrol *kcontrol, int connect, + struct snd_soc_dapm_update *update) { - struct snd_soc_card *card = widget->dapm->card; + struct snd_soc_card *card = dapm->card; int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); + card->update = update; + ret = soc_dapm_mixer_update_power(card, kcontrol, connect); + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(widget); + soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); @@ -2094,6 +2234,15 @@ static void snd_soc_dapm_sys_remove(struct device *dev) device_remove_file(dev, &dev_attr_dapm_widget); } +static void dapm_free_path(struct snd_soc_dapm_path *path) +{ + list_del(&path->list_sink); + list_del(&path->list_source); + list_del(&path->list_kcontrol); + list_del(&path->list); + kfree(path); +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2109,20 +2258,12 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) * While removing the path, remove reference to it from both * source and sink widgets so that path is removed only once. */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p->long_name); - kfree(p); - } - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p->long_name); - kfree(p); - } + list_for_each_entry_safe(p, next_p, &w->sources, list_sink) + dapm_free_path(p); + + list_for_each_entry_safe(p, next_p, &w->sinks, list_source) + dapm_free_path(p); + kfree(w->kcontrols); kfree(w->name); kfree(w); @@ -2192,70 +2333,20 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) return 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + ret = dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&dapm->card->dapm_mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_route *route) +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)) { struct snd_soc_dapm_path *path; - struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; - struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; - const char *sink; - const char *control = route->control; - const char *source; - char prefixed_sink[80]; - char prefixed_source[80]; - int ret = 0; - - if (dapm->codec && dapm->codec->name_prefix) { - snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", - dapm->codec->name_prefix, route->sink); - sink = prefixed_sink; - snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", - dapm->codec->name_prefix, route->source); - source = prefixed_source; - } else { - sink = route->sink; - source = route->source; - } - - /* - * find src and dest widgets over all widgets but favor a widget from - * current DAPM context - */ - list_for_each_entry(w, &dapm->card->widgets, list) { - if (!wsink && !(strcmp(w->name, sink))) { - wtsink = w; - if (w->dapm == dapm) - wsink = w; - continue; - } - if (!wsource && !(strcmp(w->name, source))) { - wtsource = w; - if (w->dapm == dapm) - wsource = w; - } - } - /* use widget from another DAPM context if not found from this */ - if (!wsink) - wsink = wtsink; - if (!wsource) - wsource = wtsource; - - if (wsource == NULL) { - dev_err(dapm->dev, "ASoC: no source widget found for %s\n", - route->source); - return -ENODEV; - } - if (wsink == NULL) { - dev_err(dapm->dev, "ASoC: no sink widget found for %s\n", - route->sink); - return -ENODEV; - } + int ret; path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL); if (!path) @@ -2263,8 +2354,9 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->source = wsource; path->sink = wsink; - path->connected = route->connected; + path->connected = connected; INIT_LIST_HEAD(&path->list); + INIT_LIST_HEAD(&path->list_kcontrol); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); @@ -2284,6 +2376,9 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, wsource->ext = 1; } + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + /* connect static paths */ if (control == NULL) { list_add(&path->list, &dapm->card->paths); @@ -2314,6 +2409,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2345,15 +2441,78 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, return 0; } - dapm_mark_dirty(wsource, "Route added"); - dapm_mark_dirty(wsink, "Route added"); - return 0; +err: + kfree(path); + return ret; +} + +static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; + struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; + const char *sink; + const char *source; + char prefixed_sink[80]; + char prefixed_source[80]; + int ret; + if (dapm->codec && dapm->codec->name_prefix) { + snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", + dapm->codec->name_prefix, route->sink); + sink = prefixed_sink; + snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", + dapm->codec->name_prefix, route->source); + source = prefixed_source; + } else { + sink = route->sink; + source = route->source; + } + + /* + * find src and dest widgets over all widgets but favor a widget from + * current DAPM context + */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!wsink && !(strcmp(w->name, sink))) { + wtsink = w; + if (w->dapm == dapm) + wsink = w; + continue; + } + if (!wsource && !(strcmp(w->name, source))) { + wtsource = w; + if (w->dapm == dapm) + wsource = w; + } + } + /* use widget from another DAPM context if not found from this */ + if (!wsink) + wsink = wtsink; + if (!wsource) + wsource = wtsource; + + if (wsource == NULL) { + dev_err(dapm->dev, "ASoC: no source widget found for %s\n", + route->source); + return -ENODEV; + } + if (wsink == NULL) { + dev_err(dapm->dev, "ASoC: no sink widget found for %s\n", + route->sink); + return -ENODEV; + } + + ret = snd_soc_dapm_add_path(dapm, wsource, wsink, route->control, + route->connected); + if (ret) + goto err; + + return 0; err: dev_warn(dapm->dev, "ASoC: no dapm match for %s --> %s --> %s\n", - source, control, sink); - kfree(path); + source, route->control, sink); return ret; } @@ -2398,10 +2557,7 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(path->source, "Route removed"); dapm_mark_dirty(path->sink, "Route removed"); - list_del(&path->list); - list_del(&path->list_sink); - list_del(&path->list_source); - kfree(path); + dapm_free_path(path); } else { dev_warn(dapm->dev, "ASoC: Route %s->%s does not exist\n", source, sink); @@ -2558,14 +2714,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); * * Returns 0 for success. */ -int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) +int snd_soc_dapm_new_widgets(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; unsigned int val; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); - list_for_each_entry(w, &dapm->card->widgets, list) + list_for_each_entry(w, &card->widgets, list) { if (w->new) continue; @@ -2575,7 +2731,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { - mutex_unlock(&dapm->card->dapm_mutex); + mutex_unlock(&card->dapm_mutex); return -ENOMEM; } } @@ -2601,12 +2757,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = soc_widget_read(w, w->reg); - val &= 1 << w->shift; - if (w->invert) - val = !val; - - if (val) + val = soc_widget_read(w, w->reg) >> w->shift; + val &= w->mask; + if (val == w->on_val) w->power = 1; } @@ -2616,8 +2769,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) dapm_debugfs_add_widget(w); } - dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); - mutex_unlock(&dapm->card->dapm_mutex); + dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); @@ -2634,8 +2787,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2643,17 +2796,24 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + unsigned int val; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(widget->dapm->dev, + dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); - ucontrol->value.integer.value[0] = - (snd_soc_read(widget->codec, reg) >> shift) & mask; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (dapm_kcontrol_is_powered(kcontrol)) + val = (snd_soc_read(codec, reg) >> shift) & mask; + else + val = dapm_kcontrol_get_value(kcontrol); + mutex_unlock(&card->dapm_mutex); + if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = max - val; + else + ucontrol->value.integer.value[0] = val; return 0; } @@ -2671,9 +2831,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; @@ -2685,10 +2843,9 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val; int connect, change; struct snd_soc_dapm_update update; - int wi; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(widget->dapm->dev, + dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); @@ -2697,33 +2854,30 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - mask = mask << shift; - val = val << shift; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(widget->codec, reg, mask, val); - if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; + dapm_kcontrol_set_value(kcontrol, val); - widget->value = val; + mask = mask << shift; + val = val << shift; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + change = snd_soc_test_bits(codec, reg, mask, val); + if (change) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - soc_dapm_mixer_update_power(widget, kcontrol, connect); + card->update = &update; - widget->dapm->update = NULL; - } + soc_dapm_mixer_update_power(card, kcontrol, connect); + + card->update = NULL; } mutex_unlock(&card->dapm_mutex); - return 0; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2739,12 +2893,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; - val = snd_soc_read(widget->codec, e->reg); + val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = @@ -2766,15 +2919,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; - int wi; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2790,24 +2940,17 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(codec, e->reg, mask, val); if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + card->update = &update; - widget->value = val; + soc_dapm_mux_update_power(card, kcontrol, mux, e); - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = e->reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; - - soc_dapm_mux_update_power(widget, kcontrol, mux, e); - - widget->dapm->update = NULL; - } + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2825,11 +2968,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - - ucontrol->value.enumerated.item[0] = widget->value; - + ucontrol->value.enumerated.item[0] = dapm_kcontrol_get_value(kcontrol); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); @@ -2844,34 +2983,25 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; + unsigned int value; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; - int ret = 0; - int wi; if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = widget->value != ucontrol->value.enumerated.item[0]; - if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; - - widget->value = ucontrol->value.enumerated.item[0]; - - soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); - } - } + value = ucontrol->value.enumerated.item[0]; + change = dapm_kcontrol_set_value(kcontrol, value); + if (change) + soc_dapm_mux_update_power(card, kcontrol, value, e); mutex_unlock(&card->dapm_mutex); - return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -2891,12 +3021,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, mux; - reg_val = snd_soc_read(widget->codec, e->reg); + reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; for (mux = 0; mux < e->max; mux++) { if (val == e->values[mux]) @@ -2932,15 +3061,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double); int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; - int wi; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2956,24 +3082,17 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(codec, e->reg, mask, val); if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + card->update = &update; - widget->value = val; + soc_dapm_mux_update_power(card, kcontrol, mux, e); - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = e->reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; - - soc_dapm_mux_update_power(widget, kcontrol, mux, e); - - widget->dapm->update = NULL; - } + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -3055,7 +3174,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; - size_t name_len; int ret; if ((w = dapm_cnew_widget(widget)) == NULL) @@ -3071,7 +3189,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } - if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, @@ -3096,19 +3214,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; } - name_len = strlen(widget->name) + 1; if (dapm->codec && dapm->codec->name_prefix) - name_len += 1 + strlen(dapm->codec->name_prefix); - w->name = kmalloc(name_len, GFP_KERNEL); + w->name = kasprintf(GFP_KERNEL, "%s %s", + dapm->codec->name_prefix, widget->name); + else + w->name = kasprintf(GFP_KERNEL, "%s", widget->name); + if (w->name == NULL) { kfree(w); return NULL; } - if (dapm->codec && dapm->codec->name_prefix) - snprintf((char *)w->name, name_len, "%s %s", - dapm->codec->name_prefix, widget->name); - else - snprintf((char *)w->name, name_len, "%s", widget->name); switch (w->id) { case snd_soc_dapm_switch: @@ -3121,16 +3236,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; break; - case snd_soc_dapm_adc: - case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; - case snd_soc_dapm_dac: - case snd_soc_dapm_aif_in: case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: case snd_soc_dapm_input: @@ -3146,6 +3261,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: w->power_check = dapm_supply_check_power; break; default: @@ -3410,9 +3526,6 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) { struct snd_soc_dapm_widget *dai_w, *w; struct snd_soc_dai *dai; - struct snd_soc_dapm_route r; - - memset(&r, 0, sizeof(r)); /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { @@ -3439,29 +3552,27 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname) + if (!w->sname || !strstr(w->sname, dai_w->name)) continue; if (dai->driver->playback.stream_name && strstr(w->sname, dai->driver->playback.stream_name)) { - r.source = dai->playback_widget->name; - r.sink = w->name; dev_dbg(dai->dev, "%s -> %s\n", - r.source, r.sink); + dai->playback_widget->name, w->name); - snd_soc_dapm_add_route(w->dapm, &r); + snd_soc_dapm_add_path(w->dapm, + dai->playback_widget, w, NULL, NULL); } if (dai->driver->capture.stream_name && strstr(w->sname, dai->driver->capture.stream_name)) { - r.source = w->name; - r.sink = dai->capture_widget->name; dev_dbg(dai->dev, "%s -> %s\n", - r.source, r.sink); + w->name, dai->capture_widget->name); - snd_soc_dapm_add_route(w->dapm, &r); + snd_soc_dapm_add_path(w->dapm, w, + dai->capture_widget, NULL, NULL); } } } @@ -3523,7 +3634,7 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, } } - dapm_power_widgets(&rtd->card->dapm, event); + dapm_power_widgets(rtd->card, event); } /** @@ -3792,7 +3903,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) if (dapm->bias_level == SND_SOC_BIAS_ON) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); - dapm_seq_run(dapm, &down_list, 0, false); + dapm_seq_run(card, &down_list, 0, false); if (dapm->bias_level == SND_SOC_BIAS_PREPARE) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 8ca9ecc..122c0c1 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -158,7 +158,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - return PTR_RET(codec->control_data); + return PTR_ERR_OR_ZERO(codec->control_data); } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..71358e3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, list_add(&(pins[i].list), &jack->pins); } - snd_soc_dapm_new_widgets(&jack->codec->card->dapm); - /* Update to reflect the last reported status; canned jack * implementations are likely to set their state before the * card has an opportunity to associate pins. @@ -263,7 +261,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ccb6be4..330c9a6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -33,6 +33,29 @@ #define DPCM_MAX_BE_USERS 8 +/** + * snd_soc_set_runtime_hwparams - set the runtime hardware parameters + * @substream: the pcm substream + * @hw: the hardware parameters + * + * Sets the substream runtime hardware parameters. + */ +int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, + const struct snd_pcm_hardware *hw) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + runtime->hw.info = hw->info; + runtime->hw.formats = hw->formats; + runtime->hw.period_bytes_min = hw->period_bytes_min; + runtime->hw.period_bytes_max = hw->period_bytes_max; + runtime->hw.periods_min = hw->periods_min; + runtime->hw.periods_max = hw->periods_max; + runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; + runtime->hw.fifo_size = hw->fifo_size; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); + /* DPCM stream event, send event to FE and all active BEs. */ static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) @@ -124,6 +147,26 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } +static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, + struct snd_soc_pcm_stream *codec_stream, + struct snd_soc_pcm_stream *cpu_stream) +{ + hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); + hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max); + hw->channels_min = max(codec_stream->channels_min, + cpu_stream->channels_min); + hw->channels_max = min(codec_stream->channels_max, + cpu_stream->channels_max); + hw->formats = codec_stream->formats & cpu_stream->formats; + hw->rates = codec_stream->rates & cpu_stream->rates; + if (codec_stream->rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + hw->rates |= cpu_stream->rates; + if (cpu_stream->rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + hw->rates |= codec_stream->rates; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -189,51 +232,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; + soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback, + &cpu_dai_drv->playback); } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; + soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture, + &cpu_dai_drv->capture); } ret = -EINVAL; @@ -408,8 +411,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ @@ -1829,18 +1833,10 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) /* Called by DAPM mixer/mux changes to update audio routing between PCMs and * any DAI links. */ -int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget) +int soc_dpcm_runtime_update(struct snd_soc_card *card) { - struct snd_soc_card *card; int i, old, new, paths; - if (widget->codec) - card = widget->codec->card; - else if (widget->platform) - card = widget->platform->card; - else - return -EINVAL; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dapm_widget_list *list; @@ -2024,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4b3be6c..29b211e 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -159,15 +159,10 @@ int __init snd_soc_util_init(void) { int ret; - soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1); - if (!soc_dummy_dev) - return -ENOMEM; - - ret = platform_device_add(soc_dummy_dev); - if (ret != 0) { - platform_device_put(soc_dummy_dev); - return ret; - } + soc_dummy_dev = + platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); + if (IS_ERR(soc_dummy_dev)) + return PTR_ERR(soc_dummy_dev); ret = platform_driver_register(&soc_dummy_driver); if (ret != 0) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig new file mode 100644 index 0000000..0a53053 --- /dev/null +++ b/sound/soc/spear/Kconfig @@ -0,0 +1,9 @@ +config SND_SPEAR_SOC + tristate + select SND_DMAENGINE_PCM + +config SND_SPEAR_SPDIF_OUT + tristate + +config SND_SPEAR_SPDIF_IN + tristate diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile new file mode 100644 index 0000000..c4ea716 --- /dev/null +++ b/sound/soc/spear/Makefile @@ -0,0 +1,8 @@ +# SPEAR Platform Support +snd-soc-spear-pcm-objs := spear_pcm.o +snd-soc-spear-spdif-in-objs := spdif_in.o +snd-soc-spear-spdif-out-objs := spdif_out.o + +obj-$(CONFIG_SND_SPEAR_SOC) += snd-soc-spear-pcm.o +obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += snd-soc-spear-spdif-in.o +obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += snd-soc-spear-spdif-out.o diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 14d57e8..63acfeb 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -49,15 +49,12 @@ static void spdif_in_configure(struct spdif_in_dev *host) writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); } -static int spdif_in_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int spdif_in_dai_probe(struct snd_soc_dai *dai) { - struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); - if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) - return -EINVAL; + dai->capture_dma_data = &host->dma_params; - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); return 0; } @@ -70,7 +67,6 @@ static void spdif_in_shutdown(struct snd_pcm_substream *substream, return; writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); - snd_soc_dai_set_dma_data(dai, substream, NULL); } static void spdif_in_format(struct spdif_in_dev *host, u32 format) @@ -151,13 +147,13 @@ static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, } static struct snd_soc_dai_ops spdif_in_dai_ops = { - .startup = spdif_in_startup, .shutdown = spdif_in_shutdown, .trigger = spdif_in_trigger, .hw_params = spdif_in_hw_params, }; -struct snd_soc_dai_driver spdif_in_dai = { +static struct snd_soc_dai_driver spdif_in_dai = { + .probe = spdif_in_dai_probe, .capture = { .channels_min = 2, .channels_max = 2, @@ -235,7 +231,7 @@ static int spdif_in_probe(struct platform_device *pdev) if (host->irq < 0) return -EINVAL; - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -257,34 +253,21 @@ static int spdif_in_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, "spdif-in", host); if (ret) { - clk_put(host->clk); dev_warn(&pdev->dev, "request_irq failed\n"); return ret; } - ret = snd_soc_register_component(&pdev->dev, &spdif_in_component, + return snd_soc_register_component(&pdev->dev, &spdif_in_component, &spdif_in_dai, 1); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; } static int spdif_in_remove(struct platform_device *pdev) { - struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } - static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, .remove = spdif_in_remove, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 1e3c3dd..2fdf68c 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -62,8 +62,6 @@ static int spdif_out_startup(struct snd_pcm_substream *substream, if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -EINVAL; - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); - ret = clk_enable(host->clk); if (ret) return ret; @@ -84,7 +82,6 @@ static void spdif_out_shutdown(struct snd_pcm_substream *substream, clk_disable(host->clk); host->running = false; - snd_soc_dai_set_dma_data(dai, substream, NULL); } static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq, @@ -243,8 +240,12 @@ static const struct snd_kcontrol_new spdif_out_controls[] = { spdif_mute_get, spdif_mute_put), }; -int spdif_soc_dai_probe(struct snd_soc_dai *dai) +static int spdif_soc_dai_probe(struct snd_soc_dai *dai) { + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &host->dma_params; + return snd_soc_add_dai_controls(dai, spdif_out_controls, ARRAY_SIZE(spdif_out_controls)); } @@ -281,30 +282,18 @@ static int spdif_out_probe(struct platform_device *pdev) struct resource *res; int ret; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -EINVAL; - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_warn(&pdev->dev, "Failed to get memory resourse\n"); - return -ENOENT; - } - host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - host->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!host->io_base) { - dev_warn(&pdev->dev, "ioremap failed\n"); - return -ENOMEM; - } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + host->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(host->io_base)) + return PTR_ERR(host->io_base); - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -320,22 +309,12 @@ static int spdif_out_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &spdif_out_component, &spdif_out_dai, 1); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; + return ret; } static int spdif_out_remove(struct platform_device *pdev) { - struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 2fbd489..4707f2b 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -13,19 +13,13 @@ #include <linux/module.h> #include <linux/dmaengine.h> -#include <linux/dma-mapping.h> -#include <linux/init.h> #include <linux/platform_device.h> -#include <linux/scatterlist.h> -#include <linux/slab.h> -#include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/spear_dma.h> -static struct snd_pcm_hardware spear_pcm_hardware = { +static const struct snd_pcm_hardware spear_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), @@ -37,149 +31,33 @@ static struct snd_pcm_hardware spear_pcm_hardware = { .fifo_size = 0, /* fifo size in bytes */ }; -static int spear_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static struct dma_chan *spear_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) { - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + struct spear_dma_data *dma_data; - return 0; -} - -static int spear_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - - return 0; -} - -static int spear_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - struct spear_dma_data *dma_data = (struct spear_dma_data *) - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - int ret; - - ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware); - if (ret) - return ret; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - return snd_dmaengine_pcm_open_request_chan(substream, dma_data->filter, - dma_data); + return snd_dmaengine_pcm_request_channel(dma_data->filter, dma_data); } -static int spear_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops spear_pcm_ops = { - .open = spear_pcm_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = spear_pcm_hw_params, - .hw_free = spear_pcm_hw_free, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, - .mmap = spear_pcm_mmap, -}; - -static int -spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - - dev_info(buf->dev.dev, - " preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *)buf->area, (void *)buf->addr, size); - - buf->bytes = size; - return 0; -} - -static void spear_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf || !buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); - -static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - int ret; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &spear_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (rtd->cpu_dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, - SNDRV_PCM_STREAM_PLAYBACK, - spear_pcm_hardware.buffer_bytes_max); - if (ret) - return ret; - } - - if (rtd->cpu_dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, - SNDRV_PCM_STREAM_CAPTURE, - spear_pcm_hardware.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver spear_soc_platform = { - .ops = &spear_pcm_ops, - .pcm_new = spear_pcm_new, - .pcm_free = spear_pcm_free, +static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { + .pcm_hardware = &spear_pcm_hardware, + .compat_request_channel = spear_pcm_request_chan, + .prealloc_buffer_size = 16 * 1024, }; static int spear_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &spear_soc_platform); + return snd_dmaengine_pcm_register(&pdev->dev, + &spear_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); } static int spear_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_platform(&pdev->dev); - + snd_dmaengine_pcm_unregister(&pdev->dev); return 0; } diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index b1c9d57..8fc653c 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,8 +1,8 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" - depends on ARCH_TEGRA && TEGRA20_APB_DMA + depends on (ARCH_TEGRA && TEGRA20_APB_DMA) || COMPILE_TEST select REGMAP_MMIO - select SND_SOC_GENERIC_DMAENGINE_PCM if TEGRA20_APB_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want support for SoC audio on Tegra. @@ -59,9 +59,19 @@ config SND_SOC_TEGRA30_I2S Tegra30 I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_RT5640 + tristate "SoC Audio support for Tegra boards using an RT5640 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5640 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5640 codec, such as Dalmore. + config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 @@ -71,7 +81,7 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 @@ -82,7 +92,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -100,7 +110,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 416a14b..21d2550 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -18,12 +18,14 @@ obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support +snd-soc-tegra-rt5640-objs := tegra_rt5640.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 2f70ea7..ae27bcd 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -142,13 +142,12 @@ static void tegra20_ac97_codec_write(struct snd_ac97 *ac97_snd, } while (!time_after(jiffies, timeout)); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops tegra20_ac97_ops = { .read = tegra20_ac97_codec_read, .write = tegra20_ac97_codec_write, .reset = tegra20_ac97_codec_reset, .warm_reset = tegra20_ac97_codec_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) { @@ -313,7 +312,7 @@ static const struct regmap_config tegra20_ac97_regmap_config = { static int tegra20_ac97_platform_probe(struct platform_device *pdev) { struct tegra20_ac97 *ac97; - struct resource *mem, *memregion; + struct resource *mem; u32 of_dma[2]; void __iomem *regs; int ret = 0; @@ -327,7 +326,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ac97); - ac97->clk_ac97 = clk_get(&pdev->dev, NULL); + ac97->clk_ac97 = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ac97->clk_ac97)) { dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); ret = PTR_ERR(ac97->clk_ac97); @@ -335,24 +334,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err_clk_put; } @@ -399,27 +383,13 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.slave_id = of_dma[1]; ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - ac97->capture_dma_data.maxburst = 4; - ac97->capture_dma_data.slave_id = of_dma[0]; - - ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, - &tegra20_ac97_dai, 1); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - ret = tegra_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; - } + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + ac97->playback_dma_data.maxburst = 4; + ac97->playback_dma_data.slave_id = of_dma[1]; ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); if (ret) - goto err_unregister_pcm; + goto err_clk_put; ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); if (ret) @@ -431,20 +401,38 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); + if (ret) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err_asoc_utils_fini; + } + + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, + &tegra20_ac97_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_asoc_utils_fini; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_component; + } + /* XXX: crufty ASoC AC97 API - only one AC97 codec allowed */ workdata = ac97; return 0; -err_asoc_utils_fini: - tegra_asoc_utils_fini(&ac97->util_data); -err_unregister_pcm: - tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_asoc_utils_fini: + tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: - clk_put(ac97->clk_ac97); err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -458,7 +446,8 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&ac97->util_data); clk_disable_unprepare(ac97->clk_ac97); - clk_put(ac97->clk_ac97); + + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 5eaa12c..551b3c9 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - spdif->capture_dma_data.maxburst = 4; + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->playback_dma_data.maxburst = 4; spdif->playback_dma_data.slave_id = dmareq->start; pm_runtime_enable(&pdev->dev); diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 23e592f..d554d46 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -627,9 +627,34 @@ static int tegra30_ahub_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int tegra30_ahub_suspend(struct device *dev) +{ + regcache_mark_dirty(ahub->regmap_ahub); + regcache_mark_dirty(ahub->regmap_apbif); + + return 0; +} + +static int tegra30_ahub_resume(struct device *dev) +{ + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(ahub->regmap_ahub); + ret |= regcache_sync(ahub->regmap_apbif); + pm_runtime_put(dev); + + return ret; +} +#endif + static const struct dev_pm_ops tegra30_ahub_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, tegra30_ahub_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(tegra30_ahub_suspend, tegra30_ahub_resume) }; static struct platform_driver tegra30_ahub_driver = { diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 31d092d..47565fd04 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); @@ -514,6 +514,31 @@ static int tegra30_i2s_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int tegra30_i2s_suspend(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + + regcache_mark_dirty(i2s->regmap); + + return 0; +} + +static int tegra30_i2s_resume(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; +} +#endif + static const struct of_device_id tegra30_i2s_of_match[] = { { .compatible = "nvidia,tegra30-i2s", }, {}, @@ -522,6 +547,7 @@ static const struct of_device_id tegra30_i2s_of_match[] = { static const struct dev_pm_ops tegra30_i2s_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, tegra30_i2s_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(tegra30_i2s_suspend, tegra30_i2s_resume) }; static struct platform_driver tegra30_i2s_driver = { diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9..c61ea3a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -13,8 +13,6 @@ * published by the Free Software Foundation. */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 24fb001b..d173880 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -173,7 +173,6 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) { int ret; - bool new_clocks = false; data->dev = dev; @@ -181,40 +180,28 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (of_machine_is_compatible("nvidia,tegra114")) { + else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; - new_clocks = true; - } else { + else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; } - if (new_clocks) - data->clk_pll_a = clk_get(dev, "pll_a"); - else - data->clk_pll_a = clk_get_sys(NULL, "pll_a"); + data->clk_pll_a = clk_get(dev, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); ret = PTR_ERR(data->clk_pll_a); goto err; } - if (new_clocks) - data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); - else - data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); if (IS_ERR(data->clk_pll_a_out0)) { dev_err(data->dev, "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(data->clk_pll_a_out0); goto err_put_pll_a; } - if (new_clocks) - data->clk_cdev1 = clk_get(dev, "mclk"); - else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); - else - data->clk_cdev1 = clk_get_sys("extern1", NULL); + data->clk_cdev1 = clk_get(dev, "mclk"); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c new file mode 100644 index 0000000..4511c5a --- /dev/null +++ b/sound/soc/tegra/tegra_rt5640.c @@ -0,0 +1,258 @@ +/* +* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec. + * + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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 <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5640.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5640" + +struct tegra_rt5640 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; +}; + +static int tegra_rt5640_asoc_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->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_rt5640_ops = { + .hw_params = tegra_rt5640_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5640_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5640_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, +}; + +static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5640_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + + snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins), + tegra_rt5640_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5640_hp_jack, + 1, + &tegra_rt5640_hp_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5640_dai = { + .name = "RT5640", + .stream_name = "RT5640 PCM", + .codec_dai_name = "rt5640-aif1", + .init = tegra_rt5640_asoc_init, + .ops = &tegra_rt5640_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5640 = { + .name = "tegra-rt5640", + .owner = THIS_MODULE, + .dai_link = &tegra_rt5640_dai, + .num_links = 1, + .controls = tegra_rt5640_controls, + .num_controls = ARRAY_SIZE(tegra_rt5640_controls), + .dapm_widgets = tegra_rt5640_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5640_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5640_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5640; + struct tegra_rt5640 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5640), GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_rt5640\n"); + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5640_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5640_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5640_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5640_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5640_dai.platform_of_node = tegra_rt5640_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5640_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, + &tegra_rt5640_hp_jack_gpio); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5640_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5640", }, + {}, +}; + +static struct platform_driver tegra_rt5640_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5640_of_match, + }, + .probe = tegra_rt5640_probe, + .remove = tegra_rt5640_remove, +}; +module_platform_driver(tegra_rt5640_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra+RT5640 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5640_of_match); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f87fc53..8e774d1 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -28,8 +28,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 05c68aa..734bfcd 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -24,8 +24,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8a28403..e0305a1 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -119,12 +119,11 @@ static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops txx9aclc_ac97_ops = { .read = txx9aclc_ac97_read, .write = txx9aclc_ac97_write, .reset = txx9aclc_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) { @@ -185,12 +184,9 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (irq < 0) return irq; r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) - return -EBUSY; - - if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), - dev_name(&pdev->dev))) - return -EBUSY; + drvdata->base = devm_ioremap_resource(&pdev->dev, r); + if (IS_ERR(drvdata->base)) + return PTR_ERR(drvdata->base); drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); if (!drvdata) @@ -201,14 +197,15 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) r->start >= TXX9_DIRECTMAP_BASE && r->start < TXX9_DIRECTMAP_BASE + 0x400000) drvdata->physbase |= 0xf00000000ull; - drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); - if (!drvdata->base) - return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, 0, dev_name(&pdev->dev), drvdata); if (err < 0) return err; + err = snd_soc_set_ac97_ops(&txx9aclc_ac97_ops); + if (err < 0) + return err; + return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } @@ -216,6 +213,7 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 204b899..178d1ba 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -27,7 +27,7 @@ #include "mop500_ab8500.h" /* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ -struct snd_soc_dai_link mop500_dai_links[] = { +static struct snd_soc_dai_link mop500_dai_links[] = { { .name = "ab8500_0", .stream_name = "ab8500_0", @@ -52,6 +52,7 @@ struct snd_soc_dai_link mop500_dai_links[] = { static struct snd_soc_card mop500_card = { .name = "MOP500-card", + .owner = THIS_MODULE, .probe = NULL, .dai_link = mop500_dai_links, .num_links = ARRAY_SIZE(mop500_dai_links), diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 892ad9a..7e923ec 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -16,6 +16,7 @@ #include <linux/device.h> #include <linux/io.h> #include <linux/clk.h> +#include <linux/mutex.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -24,6 +25,7 @@ #include "ux500_pcm.h" #include "ux500_msp_dai.h" +#include "mop500_ab8500.h" #include "../codecs/ab8500-codec.h" #define TX_SLOT_MONO 0x0008 @@ -43,6 +45,12 @@ static unsigned int tx_slots = DEF_TX_SLOTS; static unsigned int rx_slots = DEF_RX_SLOTS; +/* Configuration consistency parameters */ +static DEFINE_MUTEX(mop500_ab8500_params_lock); +static unsigned long mop500_ab8500_usage; +static int mop500_ab8500_rate; +static int mop500_ab8500_channels; + /* Clocks */ static const char * const enum_mclk[] = { "SYSCLK", @@ -125,9 +133,9 @@ static int mop500_ab8500_set_mclk(struct device *dev, static int mclk_input_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct mop500_ab8500_drvdata *drvdata = - snd_soc_card_get_drvdata(codec->card); + snd_soc_card_get_drvdata(card); ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; @@ -137,9 +145,9 @@ static int mclk_input_control_get(struct snd_kcontrol *kcontrol, static int mclk_input_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct mop500_ab8500_drvdata *drvdata = - snd_soc_card_get_drvdata(codec->card); + snd_soc_card_get_drvdata(card); unsigned int val = ucontrol->value.enumerated.item[0]; if (val > (unsigned int)MCLK_ULPCLK) @@ -160,16 +168,6 @@ static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { SOC_ENUM_EXT("Master Clock Select", soc_enum_mclk, mclk_input_control_get, mclk_input_control_put), - /* Digital interface - Clocks */ - SOC_SINGLE("Digital Interface Master Generator Switch", - AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, - 1, 0), - SOC_SINGLE("Digital Interface 0 Bit-clock Switch", - AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, - 1, 0), - SOC_SINGLE("Digital Interface 1 Bit-clock Switch", - AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, - 1, 0), SOC_DAPM_PIN_SWITCH("Headset Left"), SOC_DAPM_PIN_SWITCH("Headset Right"), SOC_DAPM_PIN_SWITCH("Earpiece"), @@ -193,7 +191,7 @@ static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { /* ASoC */ -int mop500_ab8500_startup(struct snd_pcm_substream *substream) +static int mop500_ab8500_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -202,7 +200,7 @@ int mop500_ab8500_startup(struct snd_pcm_substream *substream) snd_soc_card_get_drvdata(rtd->card)); } -void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +static void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct device *dev = rtd->card->dev; @@ -216,7 +214,7 @@ void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) rx_slots = DEF_RX_SLOTS; } -int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, +static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -240,6 +238,21 @@ int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, substream->name, substream->number); + /* Ensure configuration consistency between DAIs */ + mutex_lock(&mop500_ab8500_params_lock); + if (mop500_ab8500_usage) { + if (mop500_ab8500_rate != params_rate(params) || + mop500_ab8500_channels != params_channels(params)) { + mutex_unlock(&mop500_ab8500_params_lock); + return -EBUSY; + } + } else { + mop500_ab8500_rate = params_rate(params); + mop500_ab8500_channels = params_channels(params); + } + __set_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + channels = params_channels(params); switch (params_format(params)) { @@ -338,9 +351,22 @@ int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, return 0; } +static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + mutex_lock(&mop500_ab8500_params_lock); + __clear_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + + return 0; +} + struct snd_soc_ops mop500_ab8500_ops[] = { { .hw_params = mop500_ab8500_hw_params, + .hw_free = mop500_ab8500_hw_free, .startup = mop500_ab8500_startup, .shutdown = mop500_ab8500_shutdown, } @@ -385,7 +411,7 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) drvdata->mclk_sel = MCLK_ULPCLK; /* Add controls */ - ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ret = snd_soc_add_card_controls(codec->card, mop500_ab8500_ctrls, ARRAY_SIZE(mop500_ab8500_ctrls)); if (ret < 0) { pr_err("%s: Failed to add machine-controls (%d)!\n", diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 7d5fc13..c6fb5cc 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -658,14 +658,11 @@ static int ux500_msp_dai_probe(struct snd_soc_dai *dai) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); - drvdata->playback_dma_data.dma_cfg = drvdata->msp->dma_cfg_tx; - drvdata->capture_dma_data.dma_cfg = drvdata->msp->dma_cfg_rx; + dai->playback_dma_data = &drvdata->msp->playback_dma_data; + dai->capture_dma_data = &drvdata->msp->capture_dma_data; - dai->playback_dma_data = &drvdata->playback_dma_data; - dai->capture_dma_data = &drvdata->capture_dma_data; - - drvdata->playback_dma_data.data_size = drvdata->slot_width; - drvdata->capture_dma_data.data_size = drvdata->slot_width; + drvdata->msp->playback_dma_data.data_size = drvdata->slot_width; + drvdata->msp->capture_dma_data.data_size = drvdata->slot_width; return 0; } diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index f531043..312ae53 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -51,15 +51,11 @@ enum ux500_msp_clock_id { struct ux500_msp_i2s_drvdata { struct ux500_msp *msp; struct regulator *reg_vape; - struct ux500_msp_dma_params playback_dma_data; - struct ux500_msp_dma_params capture_dma_data; unsigned int fmt; unsigned int tx_mask; unsigned int rx_mask; int slots; int slot_width; - u8 configured; - int data_delay; /* Clocks */ unsigned int master_clk; diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index f2db6c9..1ca8b08 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -15,7 +15,6 @@ #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/pinctrl/consumer.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/io.h> @@ -26,9 +25,6 @@ #include "ux500_msp_i2s.h" -/* MSP1/3 Tx/Rx usage protection */ -static DEFINE_SPINLOCK(msp_rxtx_lock); - /* Protocol desciptors */ static const struct msp_protdesc prot_descs[] = { { /* I2S */ @@ -356,24 +352,8 @@ static int configure_multichannel(struct ux500_msp *msp, static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) { - int status = 0, retval = 0; + int status = 0; u32 reg_val_DMACR, reg_val_GCR; - unsigned long flags; - - /* Check msp state whether in RUN or CONFIGURED Mode */ - if (msp->msp_state == MSP_STATE_IDLE) { - spin_lock_irqsave(&msp_rxtx_lock, flags); - if (msp->pinctrl_rxtx_ref == 0 && - !(IS_ERR(msp->pinctrl_p) || IS_ERR(msp->pinctrl_def))) { - retval = pinctrl_select_state(msp->pinctrl_p, - msp->pinctrl_def); - if (retval) - pr_err("could not set MSP defstate\n"); - } - if (!retval) - msp->pinctrl_rxtx_ref++; - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - } /* Configure msp with protocol dependent settings */ configure_protocol(msp, config); @@ -387,12 +367,14 @@ static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) } /* Make sure the correct DMA-directions are configured */ - if ((config->direction & MSP_DIR_RX) && (!msp->dma_cfg_rx)) { + if ((config->direction & MSP_DIR_RX) && + !msp->capture_dma_data.dma_cfg) { dev_err(msp->dev, "%s: ERROR: MSP RX-mode is not configured!", __func__); return -EINVAL; } - if ((config->direction == MSP_DIR_TX) && (!msp->dma_cfg_tx)) { + if ((config->direction == MSP_DIR_TX) && + !msp->playback_dma_data.dma_cfg) { dev_err(msp->dev, "%s: ERROR: MSP TX-mode is not configured!", __func__); return -EINVAL; @@ -630,8 +612,7 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) { - int status = 0, retval = 0; - unsigned long flags; + int status = 0; dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); @@ -643,18 +624,6 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) (~(FRAME_GEN_ENABLE | SRG_ENABLE))), msp->registers + MSP_GCR); - spin_lock_irqsave(&msp_rxtx_lock, flags); - WARN_ON(!msp->pinctrl_rxtx_ref); - msp->pinctrl_rxtx_ref--; - if (msp->pinctrl_rxtx_ref == 0 && - !(IS_ERR(msp->pinctrl_p) || IS_ERR(msp->pinctrl_sleep))) { - retval = pinctrl_select_state(msp->pinctrl_p, - msp->pinctrl_sleep); - if (retval) - pr_err("could not set MSP sleepstate\n"); - } - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - writel(0, msp->registers + MSP_GCR); writel(0, msp->registers + MSP_TCF); writel(0, msp->registers + MSP_RCF); @@ -682,7 +651,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct msp_i2s_platform_data *platform_data) { struct resource *res = NULL; - struct i2s_controller *i2s_cont; struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; @@ -707,8 +675,8 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->id = platform_data->id; msp->dev = &pdev->dev; - msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; - msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; + msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx; + msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { @@ -717,6 +685,9 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, return -ENOMEM; } + msp->playback_dma_data.tx_rx_addr = res->start + MSP_DR; + msp->capture_dma_data.tx_rx_addr = res->start + MSP_DR; + msp->registers = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (msp->registers == NULL) { @@ -727,41 +698,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->msp_state = MSP_STATE_IDLE; msp->loopback_enable = 0; - /* I2S-controller is allocated and added in I2S controller class. */ - i2s_cont = devm_kzalloc(&pdev->dev, sizeof(*i2s_cont), GFP_KERNEL); - if (!i2s_cont) { - dev_err(&pdev->dev, - "%s: ERROR: Failed to allocate I2S-controller!\n", - __func__); - return -ENOMEM; - } - i2s_cont->dev.parent = &pdev->dev; - i2s_cont->data = (void *)msp; - i2s_cont->id = (s16)msp->id; - snprintf(i2s_cont->name, sizeof(i2s_cont->name), "ux500-msp-i2s.%04x", - msp->id); - dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); - msp->i2s_cont = i2s_cont; - - msp->pinctrl_p = pinctrl_get(msp->dev); - if (IS_ERR(msp->pinctrl_p)) - dev_err(&pdev->dev, "could not get MSP pinctrl\n"); - else { - msp->pinctrl_def = pinctrl_lookup_state(msp->pinctrl_p, - PINCTRL_STATE_DEFAULT); - if (IS_ERR(msp->pinctrl_def)) { - dev_err(&pdev->dev, - "could not get MSP defstate (%li)\n", - PTR_ERR(msp->pinctrl_def)); - } - msp->pinctrl_sleep = pinctrl_lookup_state(msp->pinctrl_p, - PINCTRL_STATE_SLEEP); - if (IS_ERR(msp->pinctrl_sleep)) - dev_err(&pdev->dev, - "could not get MSP idlestate (%li)\n", - PTR_ERR(msp->pinctrl_def)); - } - return 0; } @@ -769,8 +705,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, struct ux500_msp *msp) { dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); - - device_unregister(&msp->i2s_cont->dev); } MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index e5cd105..258d0bc 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -16,6 +16,7 @@ #define UX500_MSP_I2S_H #include <linux/platform_device.h> +#include <linux/platform_data/asoc-ux500-msp.h> #define MSP_INPUT_FREQ_APB 48000000 @@ -341,11 +342,6 @@ enum msp_compress_mode { MSP_COMPRESS_MODE_A_LAW = 3 }; -enum msp_spi_burst_mode { - MSP_SPI_BURST_MODE_DISABLE = 0, - MSP_SPI_BURST_MODE_ENABLE = 1 -}; - enum msp_expand_mode { MSP_EXPAND_MODE_LINEAR = 0, MSP_EXPAND_MODE_LINEAR_SIGNED = 1, @@ -370,13 +366,6 @@ enum msp_protocol { */ #define MAX_MSP_BACKUP_REGS 36 -enum enum_i2s_controller { - MSP_0_I2S_CONTROLLER = 0, - MSP_1_I2S_CONTROLLER, - MSP_2_I2S_CONTROLLER, - MSP_3_I2S_CONTROLLER, -}; - enum i2s_direction_t { MSP_DIR_TX = 0x01, MSP_DIR_RX = 0x02, @@ -454,32 +443,6 @@ struct msp_protdesc { u32 clocks_per_frame; }; -struct i2s_message { - enum i2s_direction_t i2s_direction; - void *txdata; - void *rxdata; - size_t txbytes; - size_t rxbytes; - int dma_flag; - int tx_offset; - int rx_offset; - bool cyclic_dma; - dma_addr_t buf_addr; - size_t buf_len; - size_t period_len; -}; - -struct i2s_controller { - struct module *owner; - unsigned int id; - unsigned int class; - const struct i2s_algorithm *algo; /* the algorithm to access the bus */ - void *data; - struct mutex bus_lock; - struct device dev; /* the controller device */ - char name[48]; -}; - struct ux500_msp_config { unsigned int f_inputclk; unsigned int rx_clk_sel; @@ -491,8 +454,6 @@ struct ux500_msp_config { unsigned int tx_fsync_sel; unsigned int rx_fifo_config; unsigned int tx_fifo_config; - unsigned int spi_clk_mode; - unsigned int spi_burst_mode; unsigned int loopback_enable; unsigned int tx_data_enable; unsigned int default_protdesc; @@ -502,43 +463,28 @@ struct ux500_msp_config { unsigned int direction; unsigned int protocol; unsigned int frame_freq; - unsigned int frame_size; enum msp_data_size data_size; unsigned int def_elem_len; unsigned int iodelay; - void (*handler) (void *data); - void *tx_callback_data; - void *rx_callback_data; +}; + +struct ux500_msp_dma_params { + unsigned int data_size; + dma_addr_t tx_rx_addr; + struct stedma40_chan_cfg *dma_cfg; }; struct ux500_msp { - enum enum_i2s_controller id; + enum msp_i2s_id id; void __iomem *registers; struct device *dev; - struct i2s_controller *i2s_cont; - struct stedma40_chan_cfg *dma_cfg_rx; - struct stedma40_chan_cfg *dma_cfg_tx; - struct dma_chan *tx_pipeid; - struct dma_chan *rx_pipeid; + struct ux500_msp_dma_params playback_dma_data; + struct ux500_msp_dma_params capture_dma_data; enum msp_state msp_state; - int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); - struct timer_list notify_timer; int def_elem_len; unsigned int dir_busy; int loopback_enable; - u32 backup_regs[MAX_MSP_BACKUP_REGS]; unsigned int f_bitclk; - /* Pin modes */ - struct pinctrl *pinctrl_p; - struct pinctrl_state *pinctrl_def; - struct pinctrl_state *pinctrl_sleep; - /* Reference Count */ - int pinctrl_rxtx_ref; -}; - -struct ux500_msp_dma_params { - unsigned int data_size; - struct stedma40_chan_cfg *dma_cfg; }; struct msp_i2s_platform_data; diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index b6e5ae2..ce554de 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -76,20 +76,20 @@ static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, dma_params = snd_soc_dai_get_dma_data(dai, substream); dma_cfg = dma_params->dma_cfg; - mem_data_width = STEDMA40_HALFWORD_WIDTH; + mem_data_width = DMA_SLAVE_BUSWIDTH_2_BYTES; switch (dma_params->data_size) { case 32: - per_data_width = STEDMA40_WORD_WIDTH; + per_data_width = DMA_SLAVE_BUSWIDTH_4_BYTES; break; case 16: - per_data_width = STEDMA40_HALFWORD_WIDTH; + per_data_width = DMA_SLAVE_BUSWIDTH_2_BYTES; break; case 8: - per_data_width = STEDMA40_BYTE_WIDTH; + per_data_width = DMA_SLAVE_BUSWIDTH_1_BYTE; break; default: - per_data_width = STEDMA40_WORD_WIDTH; + per_data_width = DMA_SLAVE_BUSWIDTH_4_BYTES; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -103,10 +103,40 @@ static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, return snd_dmaengine_pcm_request_channel(stedma40_filter, dma_cfg); } +static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ux500_msp_dma_params *dma_params; + struct stedma40_chan_cfg *dma_cfg; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_cfg = dma_params->dma_cfg; + + ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); + if (ret) + return ret; + + slave_config->dst_maxburst = 4; + slave_config->dst_addr_width = dma_cfg->dst_info.data_width; + slave_config->src_maxburst = 4; + slave_config->src_addr_width = dma_cfg->src_info.data_width; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + slave_config->dst_addr = dma_params->tx_rx_addr; + else + slave_config->src_addr = dma_params->tx_rx_addr; + + return 0; +} + static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .pcm_hardware = &ux500_pcm_hw, .compat_request_channel = ux500_pcm_request_chan, .prealloc_buffer_size = 128 * 1024, + .prepare_slave_config = ux500_pcm_prepare_slave_config, }; int ux500_pcm_register_platform(struct platform_device *pdev) diff --git a/sound/sound_core.c b/sound/sound_core.c index 359753f..45759f4 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -292,7 +292,7 @@ retry: } device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, s->name+6); + NULL, "%s", s->name+6); return s->unit_minor; fail: diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 75e6016..eee7afc 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2670,8 +2670,6 @@ static int dbri_remove(struct platform_device *op) snd_dbri_free(card->private_data); snd_card_free(card); - dev_set_drvdata(&op->dev, NULL); - return 0; } diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index a1a24b9..8e3d9a6 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1070,7 +1070,6 @@ out: ssc_free(chip->ssc); snd_card_free(card); - dev_set_drvdata(&spi->dev, NULL); return 0; } diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index 4394ae7..c39c779 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -30,7 +30,7 @@ MODULE_AUTHOR("Torsten Schenk <torsten.schenk@zoho.com>"); MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); +MODULE_SUPPORTED_DEVICE("{{TerraTec,DMX 6Fire USB}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ff..23452ee 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -110,19 +110,37 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) static int usb6fire_comm_write8(struct comm_runtime *rt, u8 request, u8 reg, u8 value) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, value, 0x00); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } static int usb6fire_comm_write16(struct comm_runtime *rt, u8 request, u8 reg, u8 vl, u8 vh) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, vl, vh); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } int usb6fire_comm_init(struct sfire_chip *chip) @@ -135,6 +153,12 @@ int usb6fire_comm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->receiver_buffer = kzalloc(COMM_RECEIVER_BUFSIZE, GFP_KERNEL); + if (!rt->receiver_buffer) { + kfree(rt); + return -ENOMEM; + } + urb = &rt->receiver; rt->serial = 1; rt->chip = chip; @@ -153,6 +177,7 @@ int usb6fire_comm_init(struct sfire_chip *chip) urb->interval = 1; ret = usb_submit_urb(urb, GFP_KERNEL); if (ret < 0) { + kfree(rt->receiver_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create comm data receiver."); return ret; @@ -171,6 +196,9 @@ void usb6fire_comm_abort(struct sfire_chip *chip) void usb6fire_comm_destroy(struct sfire_chip *chip) { - kfree(chip->comm); + struct comm_runtime *rt = chip->comm; + + kfree(rt->receiver_buffer); + kfree(rt); chip->comm = NULL; } diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 6a0840b..780d5ed 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -24,7 +24,7 @@ struct comm_runtime { struct sfire_chip *chip; struct urb receiver; - u8 receiver_buffer[COMM_RECEIVER_BUFSIZE]; + u8 *receiver_buffer; u8 serial; /* urb serial */ diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index b9defcd..780bf3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version) if (!memcmp(version, known_fw_versions + i, 2)) return 0; - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. " "please reconnect to power. if this failure " "still happens, check your firmware installation.", - 4, version); + version); return -EINVAL; } diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c index 2672242..f3dd726 100644 --- a/sound/usb/6fire/midi.c +++ b/sound/usb/6fire/midi.c @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ int usb6fire_midi_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ int usb6fire_midi_init(struct sfire_chip *chip) ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_abort(struct sfire_chip *chip) void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index c321006..84851b9 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct midi_runtime { struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length); diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 40dd50a..b5eb97f 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -450,13 +450,13 @@ static int usb6fire_pcm_close(struct snd_pcm_substream *alsa_sub) static int usb6fire_pcm_hw_params(struct snd_pcm_substream *alsa_sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(alsa_sub, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(alsa_sub, + params_buffer_bytes(hw_params)); } static int usb6fire_pcm_hw_free(struct snd_pcm_substream *alsa_sub) { - return snd_pcm_lib_free_pages(alsa_sub); + return snd_pcm_lib_free_vmalloc_buffer(alsa_sub); } static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub) @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; @@ -560,6 +560,8 @@ static struct snd_pcm_ops pcm_ops = { .prepare = usb6fire_pcm_prepare, .trigger = usb6fire_pcm_trigger, .pointer = usb6fire_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static void usb6fire_pcm_init_urb(struct pcm_urb *urb, @@ -580,6 +582,33 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; } +static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->out_urbs[i].buffer) + return -ENOMEM; + rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->in_urbs[i].buffer) + return -ENOMEM; + } + return 0; +} + +static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + kfree(rt->out_urbs[i].buffer); + kfree(rt->in_urbs[i].buffer); + } +} + int usb6fire_pcm_init(struct sfire_chip *chip) { int i; @@ -591,6 +620,13 @@ int usb6fire_pcm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + ret = usb6fire_pcm_buffers_init(rt); + if (ret) { + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); + return ret; + } + rt->chip = chip; rt->stream_state = STREAM_DISABLED; rt->rate = ARRAY_SIZE(rates); @@ -612,6 +648,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm); if (ret < 0) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n"); return ret; @@ -622,11 +659,8 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); - ret = snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - MAX_BUFSIZE, MAX_BUFSIZE); if (ret) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "error preallocating pcm buffers.\n"); @@ -641,17 +675,25 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; + unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) + if (rt->playback.instance) { + snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); snd_pcm_stop(rt->playback.instance, SNDRV_PCM_STATE_XRUN); - if (rt->capture.instance) + snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); + } + + if (rt->capture.instance) { + snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); snd_pcm_stop(rt->capture.instance, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); + } for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); @@ -663,6 +705,9 @@ void usb6fire_pcm_abort(struct sfire_chip *chip) void usb6fire_pcm_destroy(struct sfire_chip *chip) { - kfree(chip->pcm); + struct pcm_runtime *rt = chip->pcm; + + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); chip->pcm = NULL; } diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h index 9b01133..f5779d6 100644 --- a/sound/usb/6fire/pcm.h +++ b/sound/usb/6fire/pcm.h @@ -32,7 +32,7 @@ struct pcm_urb { struct urb instance; struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB]; /* END DO NOT SEPARATE */ - u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE]; + u8 *buffer; struct pcm_urb *peer; }; diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd7..de9408b 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,36 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Select this option to include support for M2Tech hiFace USB-SPDIF + interface. + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc..abe668f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index c191618..7103b09 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -183,14 +183,15 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(sub, + params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); deactivate_substream(cdev, sub); - return snd_pcm_lib_free_pages(sub); + return snd_pcm_lib_free_vmalloc_buffer(sub); } /* this should probably go upstream */ @@ -345,7 +346,9 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .hw_free = snd_usb_caiaq_pcm_hw_free, .prepare = snd_usb_caiaq_pcm_prepare, .trigger = snd_usb_caiaq_pcm_trigger, - .pointer = snd_usb_caiaq_pcm_pointer + .pointer = snd_usb_caiaq_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, @@ -852,11 +855,6 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) snd_pcm_set_ops(cdev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); - snd_pcm_lib_preallocate_pages_for_all(cdev->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); - cdev->data_cb_info = kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 48b63cc..1a61dd1 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -39,25 +39,24 @@ MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); MODULE_DESCRIPTION("caiaq USB audio"); MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," - "{Native Instruments, RigKontrol3}," - "{Native Instruments, Kore Controller}," - "{Native Instruments, Kore Controller 2}," - "{Native Instruments, Audio Kontrol 1}," - "{Native Instruments, Audio 2 DJ}," - "{Native Instruments, Audio 4 DJ}," - "{Native Instruments, Audio 8 DJ}," - "{Native Instruments, Traktor Audio 2}," - "{Native Instruments, Session I/O}," - "{Native Instruments, GuitarRig mobile}," - "{Native Instruments, Traktor Kontrol X1}," - "{Native Instruments, Traktor Kontrol S4}," - "{Native Instruments, Maschine Controller}}"); +MODULE_SUPPORTED_DEVICE("{{Native Instruments,RigKontrol2}," + "{Native Instruments,RigKontrol3}," + "{Native Instruments,Kore Controller}," + "{Native Instruments,Kore Controller 2}," + "{Native Instruments,Audio Kontrol 1}," + "{Native Instruments,Audio 2 DJ}," + "{Native Instruments,Audio 4 DJ}," + "{Native Instruments,Audio 8 DJ}," + "{Native Instruments,Traktor Audio 2}," + "{Native Instruments,Session I/O}," + "{Native Instruments,GuitarRig mobile}," + "{Native Instruments,Traktor Kontrol X1}," + "{Native Instruments,Traktor Kontrol S4}," + "{Native Instruments,Maschine Controller}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int snd_card_used[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the caiaq sound device"); @@ -388,7 +387,7 @@ static int create_card(struct usb_device *usb_dev, struct snd_usb_caiaqdev *cdev; for (devnum = 0; devnum < SNDRV_CARDS; devnum++) - if (enable[devnum] && !snd_card_used[devnum]) + if (enable[devnum]) break; if (devnum >= SNDRV_CARDS) diff --git a/sound/usb/card.h b/sound/usb/card.h index bf2889a..5ecacaa 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -21,6 +21,7 @@ struct audioformat { unsigned char endpoint; /* endpoint */ unsigned char ep_attr; /* endpoint attributes */ unsigned char datainterval; /* log_2 of data packet interval */ + unsigned char protocol; /* UAC_VERSION_1/2 */ unsigned int maxpacksize; /* max. packet size */ unsigned int rates; /* rate bitmasks */ unsigned int rate_min, rate_max; /* min/max rates */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 3a2ce39..86f80c6 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -407,9 +407,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate) { - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - - switch (altsd->bInterfaceProtocol) { + switch (fmt->protocol) { case UAC_VERSION_1: default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7a444b5..93e970f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -418,6 +418,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep; int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + if (WARN_ON(!alts)) + return NULL; + mutex_lock(&chip->mutex); list_for_each_entry(ep, &chip->ep_list, list) { @@ -591,17 +594,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate max. frequency */ - if (ep->maxpacksize) { + /* assume max. frequency is 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ maxsize = ep->maxpacksize; ep->freqmax = (maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); } if (ep->fill_max) diff --git a/sound/usb/format.c b/sound/usb/format.c index 99299ff..3525231 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -43,13 +43,12 @@ */ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned int format, void *_fmt, - int protocol) + unsigned int format, void *_fmt) { int sample_width, sample_bytes; u64 pcm_formats = 0; - switch (protocol) { + switch (fp->protocol) { case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; @@ -360,11 +359,8 @@ err: */ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int format, - struct uac_format_type_i_continuous_descriptor *fmt, - struct usb_host_interface *iface) + struct uac_format_type_i_continuous_descriptor *fmt) { - struct usb_interface_descriptor *altsd = get_iface_desc(iface); - int protocol = altsd->bInterfaceProtocol; snd_pcm_format_t pcm_format; int ret; @@ -387,8 +383,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, } fp->formats = pcm_format_to_bits(pcm_format); } else { - fp->formats = parse_audio_format_i_type(chip, fp, format, - fmt, protocol); + fp->formats = parse_audio_format_i_type(chip, fp, format, fmt); if (!fp->formats) return -EINVAL; } @@ -398,11 +393,8 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * proprietary class specific descriptor. * audio class v2 uses class specific EP0 range requests for that. */ - switch (protocol) { + switch (fp->protocol) { default: - snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); @@ -427,12 +419,9 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *_fmt, - struct usb_host_interface *iface) + int format, void *_fmt) { int brate, framesize, ret; - struct usb_interface_descriptor *altsd = get_iface_desc(iface); - int protocol = altsd->bInterfaceProtocol; switch (format) { case UAC_FORMAT_TYPE_II_AC3: @@ -452,11 +441,8 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; - switch (protocol) { + switch (fp->protocol) { default: - snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -483,17 +469,17 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int format, struct uac_format_type_i_continuous_descriptor *fmt, - int stream, struct usb_host_interface *iface) + int stream) { int err; switch (fmt->bFormatType) { case UAC_FORMAT_TYPE_I: case UAC_FORMAT_TYPE_III: - err = parse_audio_format_i(chip, fp, format, fmt, iface); + err = parse_audio_format_i(chip, fp, format, fmt); break; case UAC_FORMAT_TYPE_II: - err = parse_audio_format_ii(chip, fp, format, fmt, iface); + err = parse_audio_format_ii(chip, fp, format, fmt); break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", diff --git a/sound/usb/format.h b/sound/usb/format.h index 6f31522..4b8a011 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -4,6 +4,6 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int format, struct uac_format_type_i_continuous_descriptor *fmt, - int stream, struct usb_host_interface *iface); + int stream); #endif /* __USBAUDIO_FORMAT_H */ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 0000000..463b136 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 0000000..b0dcb39 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,297 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael@amarulasolutions.com> + * Antonio Ospite <ao2@amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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/slab.h> +#include <sound/initval.h> + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi <michael@amarulasolutions.com>"); +MODULE_AUTHOR("Antonio Ospite <ao2@amarulasolutions.com>"); +MODULE_DESCRIPTION("M2Tech hiFace USB-SPDIF audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{M2Tech,Young}," + "{M2Tech,hiFace}," + "{M2Tech,North Star}," + "{M2Tech,W4S Young}," + "{M2Tech,Corrson}," + "{M2Tech,AUDIA}," + "{M2Tech,SL Audio}," + "{M2Tech,Empirical}," + "{M2Tech,Rockna}," + "{M2Tech,Pathos}," + "{M2Tech,Metronome}," + "{M2Tech,CAD}," + "{M2Tech,Audio Esclusive}," + "{M2Tech,Rotel}," + "{M2Tech,Eeaudio}," + "{The Chord Company,CHORD}," + "{AVA Group A/S,Vitus}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "hiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + int len; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, sizeof(*chip), &card); + if (ret < 0) { + dev_err(&device->dev, "cannot create alsa card.\n"); + return ret; + } + + strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); + + if (quirk && quirk->device_name) + strlcpy(card->shortname, quirk->device_name, sizeof(card->shortname)); + else + strlcpy(card->shortname, "M2Tech generic audio", sizeof(card->shortname)); + + strlcat(card->longname, card->shortname, sizeof(card->longname)); + len = strlcat(card->longname, " at ", sizeof(card->longname)); + if (len < sizeof(card->longname)) + usb_make_path(device, card->longname + len, + sizeof(card->longname) - len); + + chip = card->private_data; + chip->dev = device; + chip->card = card; + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + dev_err(&device->dev, "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(®ister_mutex); + + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i]) + break; + + if (i >= SNDRV_CARDS) { + dev_err(&device->dev, "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + + ret = hiface_chip_create(device, i, quirk, &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + dev_err(&device->dev, "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + mutex_unlock(®ister_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(®ister_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); +} + +static const struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 0000000..189a137 --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,30 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael@amarulasolutions.com> + * Antonio Ospite <ao2@amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include <linux/usb.h> +#include <sound/core.h> + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 0000000..c21a3df --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,621 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael@amarulasolutions.com> + * Antonio Ospite <ao2@amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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/slab.h> +#include <sound/pcm.h> + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_PACKET_SIZE 4096 +#define PCM_BUFFER_SIZE (2 * PCM_N_URBS * PCM_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static const struct snd_pcm_hw_constraint_list constraints_extra_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = PCM_BUFFER_SIZE, + .period_bytes_min = PCM_PACKET_SIZE, + .period_bytes_max = PCM_BUFFER_SIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +/* message values used to change the sample rate */ +#define HIFACE_SET_RATE_REQUEST 0xb0 + +#define HIFACE_RATE_44100 0x43 +#define HIFACE_RATE_48000 0x4b +#define HIFACE_RATE_88200 0x42 +#define HIFACE_RATE_96000 0x4a +#define HIFACE_RATE_176400 0x40 +#define HIFACE_RATE_192000 0x48 +#define HIFACE_RATE_352000 0x58 +#define HIFACE_RATE_384000 0x68 + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + struct usb_device *device = rt->chip->dev; + u16 rate_value; + int ret; + + /* We are already sure that the rate is supported here thanks to + * ALSA constraints + */ + switch (rate) { + case 44100: + rate_value = HIFACE_RATE_44100; + break; + case 48000: + rate_value = HIFACE_RATE_48000; + break; + case 88200: + rate_value = HIFACE_RATE_88200; + break; + case 96000: + rate_value = HIFACE_RATE_96000; + break; + case 176400: + rate_value = HIFACE_RATE_176400; + break; + case 192000: + rate_value = HIFACE_RATE_192000; + break; + case 352000: + rate_value = HIFACE_RATE_352000; + break; + case 384000: + rate_value = HIFACE_RATE_384000; + break; + default: + dev_err(&device->dev, "Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + HIFACE_SET_RATE_REQUEST, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value, 0, NULL, 0, 100); + if (ret < 0) { + dev_err(&device->dev, "Error setting samplerate %d.\n", rate); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream(struct snd_pcm_substream + *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct device *device = &rt->chip->dev->dev; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + dev_err(device, "Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0; + int i; + + if (rt->stream_state == STREAM_DISABLED) { + + /* reset panic state when starting a new stream */ + rt->panic = false; + + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + struct device *device = &rt->chip->dev->dev; + dev_dbg(device, "%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + +/* The hardware wants word-swapped 32-bit values */ +static void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +/* returns true if a period elapsed */ +static bool hiface_pcm_playback(struct pcm_substream *sub, struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + struct device *device = &urb->chip->dev->dev; + u8 *source; + unsigned int pcm_buffer_size; + + WARN_ON(alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE); + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_PACKET_SIZE <= pcm_buffer_size) { + dev_dbg(device, "%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + dev_dbg(device, "%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_PACKET_SIZE - len); + } + sub->dma_off += PCM_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_PACKET_SIZE; + if (sub->period_off >= alsa_rt->period_size) { + sub->period_off %= alsa_rt->period_size; + return true; + } + return false; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + bool do_period_elapsed = false; + unsigned long flags; + int ret; + + if (rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (unlikely(usb_urb->status == -ENOENT || /* unlinked */ + usb_urb->status == -ENODEV || /* device removed */ + usb_urb->status == -ECONNRESET || /* unlinked */ + usb_urb->status == -ESHUTDOWN)) { /* device disabled */ + goto out_fail; + } + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) + do_period_elapsed = hiface_pcm_playback(sub, out_urb); + else + memset(out_urb->buffer, 0, PCM_PACKET_SIZE); + + spin_unlock_irqrestore(&sub->lock, flags); + + if (do_period_elapsed) + snd_pcm_period_elapsed(sub->instance); + + ret = usb_submit_urb(&out_urb->instance, GFP_ATOMIC); + if (ret < 0) + goto out_fail; + + return; + +out_fail: + rt->panic = true; +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + sub = &rt->playback; + + if (!sub) { + struct device *device = &rt->chip->dev->dev; + mutex_unlock(&rt->stream_mutex); + dev_err(device, "Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rates |= SNDRV_PCM_RATE_KNOT; + alsa_rt->hw.rate_max = 384000; + + /* explicit constraints needed as we added SNDRV_PCM_RATE_KNOT */ + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_extra_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + mutex_lock(&rt->stream_mutex); + if (sub) { + hiface_pcm_stream_stop(rt); + + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_alloc_vmalloc_buffer(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + return snd_pcm_lib_free_vmalloc_buffer(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irq(&sub->lock); + sub->active = true; + spin_unlock_irq(&sub->lock); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irq(&sub->lock); + sub->active = false; + spin_unlock_irq(&sub->lock); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_POS_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strlcpy(pcm->name, "USB-SPDIF Audio", sizeof(pcm->name)); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 0000000..77edd7c --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael@amarulasolutions.com> + * Antonio Ospite <ao2@amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 8e01fa4..b901f46 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1575,8 +1575,41 @@ static struct port_info { EXTERNAL_PORT(0x0582, 0x004d, 0, "%s MIDI"), EXTERNAL_PORT(0x0582, 0x004d, 1, "%s 1"), EXTERNAL_PORT(0x0582, 0x004d, 2, "%s 2"), + /* BOSS GT-PRO */ + CONTROL_PORT(0x0582, 0x0089, 0, "%s Control"), /* Edirol UM-3EX */ CONTROL_PORT(0x0582, 0x009a, 3, "%s Control"), + /* Roland VG-99 */ + CONTROL_PORT(0x0582, 0x00b2, 0, "%s Control"), + EXTERNAL_PORT(0x0582, 0x00b2, 1, "%s MIDI"), + /* Cakewalk Sonar V-Studio 100 */ + EXTERNAL_PORT(0x0582, 0x00eb, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x00eb, 1, "%s Control"), + /* Roland VB-99 */ + CONTROL_PORT(0x0582, 0x0102, 0, "%s Control"), + EXTERNAL_PORT(0x0582, 0x0102, 1, "%s MIDI"), + /* Roland A-PRO */ + EXTERNAL_PORT(0x0582, 0x010f, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x010f, 1, "%s 1"), + CONTROL_PORT(0x0582, 0x010f, 2, "%s 2"), + /* Roland SD-50 */ + ROLAND_SYNTH_PORT(0x0582, 0x0114, 0, "%s Synth", 128), + EXTERNAL_PORT(0x0582, 0x0114, 1, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0114, 2, "%s Control"), + /* Roland OCTA-CAPTURE */ + EXTERNAL_PORT(0x0582, 0x0120, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0120, 1, "%s Control"), + EXTERNAL_PORT(0x0582, 0x0121, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0121, 1, "%s Control"), + /* Roland SPD-SX */ + CONTROL_PORT(0x0582, 0x0145, 0, "%s Control"), + EXTERNAL_PORT(0x0582, 0x0145, 1, "%s MIDI"), + /* Roland A-Series */ + CONTROL_PORT(0x0582, 0x0156, 0, "%s Keyboard"), + EXTERNAL_PORT(0x0582, 0x0156, 1, "%s MIDI"), + /* Roland INTEGRA-7 */ + ROLAND_SYNTH_PORT(0x0582, 0x015b, 0, "%s Synth", 128), + CONTROL_PORT(0x0582, 0x015b, 1, "%s Control"), /* M-Audio MidiSport 8x8 */ CONTROL_PORT(0x0763, 0x1031, 8, "%s Control"), CONTROL_PORT(0x0763, 0x1033, 8, "%s Control"), @@ -1948,6 +1981,44 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, } /* + * Detects the endpoints and ports of Roland devices. + */ +static int snd_usbmidi_detect_roland(struct snd_usb_midi* umidi, + struct snd_usb_midi_endpoint_info* endpoint) +{ + struct usb_interface* intf; + struct usb_host_interface *hostif; + u8* cs_desc; + + intf = umidi->iface; + if (!intf) + return -ENOENT; + hostif = intf->altsetting; + /* + * Some devices have a descriptor <06 24 F1 02 <inputs> <outputs>>, + * some have standard class descriptors, or both kinds, or neither. + */ + for (cs_desc = hostif->extra; + cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; + cs_desc += cs_desc[0]) { + if (cs_desc[0] >= 6 && + cs_desc[1] == USB_DT_CS_INTERFACE && + cs_desc[2] == 0xf1 && + cs_desc[3] == 0x02) { + endpoint->in_cables = (1 << cs_desc[4]) - 1; + endpoint->out_cables = (1 << cs_desc[5]) - 1; + return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); + } else if (cs_desc[0] >= 7 && + cs_desc[1] == USB_DT_CS_INTERFACE && + cs_desc[2] == UAC_HEADER) { + return snd_usbmidi_get_ms_info(umidi, endpoint); + } + } + + return -ENODEV; +} + +/* * Creates the endpoints and their ports for Midiman devices. */ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, @@ -2162,6 +2233,9 @@ int snd_usbmidi_create(struct snd_card *card, case QUIRK_MIDI_YAMAHA: err = snd_usbmidi_detect_yamaha(umidi, &endpoints[0]); break; + case QUIRK_MIDI_ROLAND: + err = snd_usbmidi_detect_roland(umidi, &endpoints[0]); + break; case QUIRK_MIDI_MIDIMAN: umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; memcpy(&endpoints[0], quirk->data, diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 6ad617b..5093159 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,14 +613,24 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); + } } static void abort_alsa_playback(struct ua101 *ua) { - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); + } } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, @@ -1349,7 +1359,7 @@ static void ua101_disconnect(struct usb_interface *interface) snd_card_disconnect(ua->card); /* make sure that there are no pending USB requests */ - __list_for_each(midi, &ua->midi_list) + list_for_each(midi, &ua->midi_list) snd_usbmidi_disconnect(midi); abort_alsa_playback(ua); abort_alsa_capture(ua); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d543808..95558ef 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -888,6 +888,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..d42a584 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -9,6 +9,8 @@ * Alan Cox (alan@lxorguk.ukuu.org.uk) * Thomas Sailer (sailer@ife.ee.ethz.ch) * + * Audio Advantage Micro II support added by: + * Przemek Rudy (prudy1@o2.pl) * * 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 @@ -30,6 +32,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/control.h> #include <sound/hwdep.h> @@ -1315,6 +1318,211 @@ static struct std_mono_table ebox44_table[] = { {} }; +/* Audio Advantage Micro II findings: + * + * Mapping spdif AES bits to vendor register.bit: + * AES0: [0 0 0 0 2.3 2.2 2.1 2.0] - default 0x00 + * AES1: [3.3 3.2.3.1.3.0 2.7 2.6 2.5 2.4] - default: 0x01 + * AES2: [0 0 0 0 0 0 0 0] + * AES3: [0 0 0 0 0 0 x 0] - 'x' bit is set basing on standard usb request + * (UAC_EP_CS_ATTR_SAMPLE_RATE) for Audio Devices + * + * power on values: + * r2: 0x10 + * r3: 0x20 (b7 is zeroed just before playback (except IEC61937) and set + * just after it to 0xa0, presumably it disables/mutes some analog + * parts when there is no audio.) + * r9: 0x28 + * + * Optical transmitter on/off: + * vendor register.bit: 9.1 + * 0 - on (0x28 register value) + * 1 - off (0x2a register value) + * + */ +static int snd_microii_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned int ep; + unsigned char data[3]; + int rate; + + ucontrol->value.iec958.status[0] = kcontrol->private_value & 0xff; + ucontrol->value.iec958.status[1] = (kcontrol->private_value >> 8) & 0xff; + ucontrol->value.iec958.status[2] = 0x00; + + /* use known values for that card: interface#1 altsetting#1 */ + iface = usb_ifnum_to_if(mixer->chip->dev, 1); + alts = &iface->altsetting[1]; + ep = get_endpoint(alts, 0)->bEndpointAddress; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + ep, + data, + sizeof(data)); + if (err < 0) + goto end; + + rate = data[0] | (data[1] << 8) | (data[2] << 16); + ucontrol->value.iec958.status[3] = (rate == 48000) ? + IEC958_AES3_CON_FS_48000 : IEC958_AES3_CON_FS_44100; + + err = 0; +end: + return err; +} + +static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg; + unsigned long priv_backup = kcontrol->private_value; + + reg = ((ucontrol->value.iec958.status[1] & 0x0f) << 4) | + (ucontrol->value.iec958.status[0] & 0x0f); + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 2, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xfffff0f0; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; + kcontrol->private_value |= (ucontrol->value.iec958.status[0] & 0x0f); + + reg = (ucontrol->value.iec958.status[0] & IEC958_AES0_NONAUDIO) ? + 0xa0 : 0x20; + reg |= (ucontrol->value.iec958.status[1] >> 4) & 0x0f; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 3, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xffff0fff; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; + + /* The frequency bits in AES3 cannot be set via register access. */ + + /* Silently ignore any bits from the request that cannot be set. */ + + err = (priv_backup != kcontrol->private_value); +end: + return err; +} + +static int snd_microii_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0x0f; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0x00; + ucontrol->value.iec958.status[3] = 0x00; + + return 0; +} + +static int snd_microii_spdif_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = !(kcontrol->private_value & 0x02); + + return 0; +} + +static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 9, + NULL, + 0); + + if (!err) { + err = (reg != (kcontrol->private_value & 0x0ff)); + if (err) + kcontrol->private_value = reg; + } + + return err; +} + +static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_default_get, + .put = snd_microii_spdif_default_put, + .private_value = 0x00000100UL,/* reset value */ + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = snd_microii_spdif_switch_get, + .put = snd_microii_spdif_switch_put, + .private_value = 0x00000028UL,/* reset value */ + } +}; + +static int snd_microii_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_microii_mixer_spdif); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_microii_mixer_spdif[i], mixer)); + if (err < 0) + return err; + } + + return err; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1353,6 +1561,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; + case USB_ID(0x0d8c, 0x0103): /* Audio Advantage Micro II */ + err = snd_microii_controls_create(mixer); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 93b6e32..b375d58 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -202,13 +202,11 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt) { - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - /* if endpoint doesn't have pitch control, bail out */ if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; - switch (altsd->bInterfaceProtocol) { + switch (fmt->protocol) { case UAC_VERSION_1: default: return init_pitch_v1(chip, iface, alts, fmt); @@ -300,6 +298,166 @@ static int deactivate_endpoints(struct snd_usb_substream *subs) return 0; } +static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, + unsigned int altsetting, + struct usb_host_interface **alts, + unsigned int *ep) +{ + struct usb_interface *iface; + struct usb_interface_descriptor *altsd; + struct usb_endpoint_descriptor *epd; + + iface = usb_ifnum_to_if(dev, ifnum); + if (!iface || iface->num_altsetting < altsetting + 1) + return -ENOENT; + *alts = &iface->altsetting[altsetting]; + altsd = get_iface_desc(*alts); + if (altsd->bAlternateSetting != altsetting || + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC || + (altsd->bInterfaceSubClass != 2 && + altsd->bInterfaceProtocol != 2 ) || + altsd->bNumEndpoints < 1) + return -ENOENT; + epd = get_endpoint(*alts, 0); + if (!usb_endpoint_is_isoc_in(epd) || + (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_IMPLICIT_FB) + return -ENOENT; + *ep = epd->bEndpointAddress; + return 0; +} + +static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, + struct usb_device *dev, + struct usb_interface_descriptor *altsd, + unsigned int attr) +{ + struct usb_host_interface *alts; + struct usb_interface *iface; + unsigned int ep; + + /* Implicit feedback sync EPs consumers are always playback EPs */ + if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ + ep = 0x81; + iface = usb_ifnum_to_if(dev, 3); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + break; + case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ + case USB_ID(0x0763, 0x2081): + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + if (attr == USB_ENDPOINT_SYNC_ASYNC && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + altsd->bInterfaceProtocol == 2 && + altsd->bNumEndpoints == 1 && + USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ && + search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1, + altsd->bAlternateSetting, + &alts, &ep) >= 0) { + goto add_sync_ep; + } + + /* No quirk */ + return 0; + +add_sync_ep: + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + SND_USB_ENDPOINT_TYPE_DATA); + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + + return 0; +} + +static int set_sync_endpoint(struct snd_usb_substream *subs, + struct audioformat *fmt, + struct usb_device *dev, + struct usb_host_interface *alts, + struct usb_interface_descriptor *altsd) +{ + int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int ep, attr; + bool implicit_fb; + int err; + + /* we need a sync pipe in async OUT or adaptive IN mode */ + /* check the number of EP, since some devices have broken + * descriptors which fool us. if it has only one EP, + * assume it as adaptive-out or sync-in. + */ + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + + err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr); + if (err < 0) + return err; + + if (altsd->bNumEndpoints < 2) + return 0; + + if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) || + (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE)) + return 0; + + /* check sync-pipe endpoint */ + /* ... and check descriptor size before accessing bSynchAddress + because there is a version of the SB Audigy 2 NX firmware lacking + the audio fields in the endpoint descriptors */ + if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || + (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bSynchAddress != 0)) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); + return -EINVAL; + } + ep = get_endpoint(alts, 1)->bEndpointAddress; + if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); + return -EINVAL; + } + + implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) + == USB_ENDPOINT_USAGE_IMPLICIT_FB; + + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + + return 0; +} + /* * find a matching format and set up the interface */ @@ -309,9 +467,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; struct usb_interface *iface; - unsigned int ep, attr; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - int err, implicit_fb = 0; + int err; iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) @@ -356,106 +512,22 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, fmt->endpoint, subs->direction, SND_USB_ENDPOINT_TYPE_DATA); + if (!subs->data_endpoint) return -EINVAL; - /* we need a sync pipe in async OUT or adaptive IN mode */ - /* check the number of EP, since some devices have broken - * descriptors which fool us. if it has only one EP, - * assume it as adaptive-out or sync-in. - */ - attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; - - switch (subs->stream->chip->usb_id) { - case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ - case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 3); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - } - break; - case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ - case USB_ID(0x0763, 0x2081): - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 2); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - } - } - - if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || - (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && - altsd->bNumEndpoints >= 2) { - /* check sync-pipe endpoint */ - /* ... and check descriptor size before accessing bSynchAddress - because there is a version of the SB Audigy 2 NX firmware lacking - the audio fields in the endpoint descriptors */ - if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || - (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0 && - !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", - dev->devnum, fmt->iface, fmt->altsetting, - get_endpoint(alts, 1)->bmAttributes, - get_endpoint(alts, 1)->bLength, - get_endpoint(alts, 1)->bSynchAddress); - return -EINVAL; - } - ep = get_endpoint(alts, 1)->bEndpointAddress; - if (!implicit_fb && - get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", - dev->devnum, fmt->iface, fmt->altsetting, - is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); - return -EINVAL; - } - - implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) - == USB_ENDPOINT_USAGE_IMPLICIT_FB; - -add_sync_ep: - subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, - alts, ep, !subs->direction, - implicit_fb ? - SND_USB_ENDPOINT_TYPE_DATA : - SND_USB_ENDPOINT_TYPE_SYNC); - if (!subs->sync_endpoint) - return -EINVAL; - - subs->data_endpoint->sync_master = subs->sync_endpoint; - } + err = set_sync_endpoint(subs, fmt, dev, alts, altsd); + if (err < 0) + return err; - if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0) + err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt); + if (err < 0) return err; subs->cur_audiofmt = fmt; snd_usb_set_format_quirk(subs, fmt); -#if 0 - printk(KERN_DEBUG - "setting done: format = %d, rate = %d..%d, channels = %d\n", - fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); - printk(KERN_DEBUG - " datapipe = 0x%0x, syncpipe = 0x%0x\n", - subs->datapipe, subs->syncpipe); -#endif - return 0; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8b75bcf..f5f0595 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -461,6 +461,17 @@ YAMAHA_DEVICE(0x7000, "DTX"), YAMAHA_DEVICE(0x7010, "UB99"), #undef YAMAHA_DEVICE #undef YAMAHA_INTERFACE +/* this catches most recent vendor-specific Yamaha devices */ +{ + .match_flags = USB_DEVICE_ID_MATCH_VENDOR | + USB_DEVICE_ID_MATCH_INT_CLASS, + .idVendor = 0x0499, + .bInterfaceClass = USB_CLASS_VENDOR_SPEC, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUTODETECT + } +}, /* * Roland/RolandED/Edirol/BOSS devices @@ -1136,7 +1147,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Roland M-1000 support */ { /* * Has ID 0x0038 when not in "Advanced Driver" mode; @@ -1251,7 +1261,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Edirol M-100FX support */ { /* has ID 0x004e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x004c), @@ -1371,20 +1380,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* has ID 0x006b when not in "Advanced Driver" mode */ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x006a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "SP-606", - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, -{ /* has ID 0x006e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x006d), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { @@ -1471,8 +1466,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Roland V-SYNTH XT support */ - /* TODO: add BOSS GT-PRO support */ { /* has ID 0x008c when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x008b), @@ -1487,42 +1480,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Edirol PC-80 support */ -{ - USB_DEVICE(0x0582, 0x0096), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "UA-1EX", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x009a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "UM-3EX", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x000f, - .in_cables = 0x000f - } - } -}, { /* * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX @@ -1553,124 +1510,8 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Edirol MD-P1 support */ -{ - USB_DEVICE(0x582, 0x00a6), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "Juno-G", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, -{ - /* Roland SH-201 */ - USB_DEVICE(0x0582, 0x00ad), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "SH-201", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 - * kHz. In standard mode, the device has ID 0582:00b3, and offers - * 16-bit PCM at 44.1 kHz with no MIDI. - */ - USB_DEVICE(0x0582, 0x00b2), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "VG-99", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* Roland SonicCell */ - USB_DEVICE(0x0582, 0x00c2), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "SonicCell", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, { /* Edirol M-16DX */ - /* FIXME: This quirk gives a good-working capture stream but the - * playback seems problematic because of lacking of sync - * with capture stream. It needs to sync with the capture - * clock. As now, you'll get frequent sound distortions - * via the playback. - */ USB_DEVICE(0x0582, 0x00c4), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, @@ -1699,35 +1540,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* BOSS GT-10 */ - USB_DEVICE(0x0582, 0x00da), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ /* 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. @@ -1758,42 +1570,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* has ID 0x00ea when not in Advanced Driver mode */ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e9), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "UA-1G", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x0104), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "UM-1G", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, -{ /* Edirol UM-3G */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { @@ -1806,92 +1582,49 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* Boss JS-8 Jam Station */ - USB_DEVICE(0x0582, 0x0109), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "JS-8", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* has ID 0x0110 when not in Advanced Driver mode */ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), + /* only 44.1 kHz works at the moment */ + USB_DEVICE(0x0582, 0x0120), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Roland", */ - /* .product_name = "A-PRO", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0007 - } - } -}, -{ - /* Roland GAIA SH-01 */ - USB_DEVICE(0x0582, 0x0111), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "GAIA", + /* .product_name = "OCTO-CAPTURE", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &(const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } } }, { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x0113), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "ME-25", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 12, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x85, + .ep_attr = 0x25, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } }, { .ifnum = 2, @@ -1902,30 +1635,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x0127), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "GR-55", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE }, { - .ifnum = 2, - .type = QUIRK_MIDI_STANDARD_INTERFACE + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = -1 @@ -1934,34 +1649,49 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* Added support for Roland UM-ONE which differs from UM-1 */ - USB_DEVICE(0x0582, 0x012a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "ROLAND", */ - /* .product_name = "UM-ONE", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0003 - } - } -}, -{ - USB_DEVICE(0x0582, 0x011e), + /* only 44.1 kHz works at the moment */ + USB_DEVICE(0x0582, 0x012f), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "BR-800", */ + /* .vendor_name = "Roland", */ + /* .product_name = "QUAD-CAPTURE", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 6, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x85, + .ep_attr = 0x25, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } }, { .ifnum = 2, @@ -1972,38 +1702,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x0130), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "MICRO BR-80", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, + .ifnum = 3, .type = QUIRK_IGNORE_INTERFACE }, { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = -1 @@ -2011,34 +1715,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* this catches most recent vendor-specific Roland devices */ { - USB_DEVICE(0x0582, 0x014d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "GT-100", */ + .match_flags = USB_DEVICE_ID_MATCH_VENDOR | + USB_DEVICE_ID_MATCH_INT_CLASS, + .idVendor = 0x0582, + .bInterfaceClass = USB_CLASS_VENDOR_SPEC, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } + .type = QUIRK_AUTODETECT } }, @@ -3434,4 +3119,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ + USB_DEVICE(0x0d8c, 0x0103), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .product_name = "Audio Advantage MicroII", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3879eae..0df9ede 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -18,6 +18,7 @@ #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> +#include <linux/usb/midi.h> #include <sound/control.h> #include <sound/core.h> @@ -128,6 +129,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, { struct audioformat *fp; struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; int stream, err; unsigned *rate_table = NULL; @@ -165,6 +167,9 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + altsd = get_iface_desc(alts); + fp->protocol = altsd->bInterfaceProtocol; + if (fp->datainterval == 0) fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) @@ -175,6 +180,212 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return 0; } +static int create_auto_pcm_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver) +{ + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_endpoint_descriptor *epd; + struct uac1_as_header_descriptor *ashd; + struct uac_format_type_i_discrete_descriptor *fmtd; + + /* + * Most Roland/Yamaha audio streaming interfaces have more or less + * standard descriptors, but older devices might lack descriptors, and + * future ones might change, so ensure that we fail silently if the + * interface doesn't look exactly right. + */ + + /* must have a non-zero altsetting for streaming */ + if (iface->num_altsetting < 2) + return -ENODEV; + alts = &iface->altsetting[1]; + altsd = get_iface_desc(alts); + + /* must have an isochronous endpoint for streaming */ + if (altsd->bNumEndpoints < 1) + return -ENODEV; + epd = get_endpoint(alts, 0); + if (!usb_endpoint_xfer_isoc(epd)) + return -ENODEV; + + /* must have format descriptors */ + ashd = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, + UAC_AS_GENERAL); + fmtd = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, + UAC_FORMAT_TYPE); + if (!ashd || ashd->bLength < 7 || + !fmtd || fmtd->bLength < 8) + return -ENODEV; + + return create_standard_audio_quirk(chip, iface, driver, NULL); +} + +static int create_yamaha_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + struct usb_host_interface *alts) +{ + static const struct snd_usb_audio_quirk yamaha_midi_quirk = { + .type = QUIRK_MIDI_YAMAHA + }; + struct usb_midi_in_jack_descriptor *injd; + struct usb_midi_out_jack_descriptor *outjd; + + /* must have some valid jack descriptors */ + injd = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, USB_MS_MIDI_IN_JACK); + outjd = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, USB_MS_MIDI_OUT_JACK); + if (!injd && !outjd) + return -ENODEV; + if (injd && (injd->bLength < 5 || + (injd->bJackType != USB_MS_EMBEDDED && + injd->bJackType != USB_MS_EXTERNAL))) + return -ENODEV; + if (outjd && (outjd->bLength < 6 || + (outjd->bJackType != USB_MS_EMBEDDED && + outjd->bJackType != USB_MS_EXTERNAL))) + return -ENODEV; + return create_any_midi_quirk(chip, iface, driver, &yamaha_midi_quirk); +} + +static int create_roland_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + struct usb_host_interface *alts) +{ + static const struct snd_usb_audio_quirk roland_midi_quirk = { + .type = QUIRK_MIDI_ROLAND + }; + u8 *roland_desc = NULL; + + /* might have a vendor-specific descriptor <06 24 F1 02 ...> */ + for (;;) { + roland_desc = snd_usb_find_csint_desc(alts->extra, + alts->extralen, + roland_desc, 0xf1); + if (!roland_desc) + return -ENODEV; + if (roland_desc[0] < 6 || roland_desc[3] != 2) + continue; + return create_any_midi_quirk(chip, iface, driver, + &roland_midi_quirk); + } +} + +static int create_std_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + struct usb_host_interface *alts) +{ + struct usb_ms_header_descriptor *mshd; + struct usb_ms_endpoint_descriptor *msepd; + + /* must have the MIDIStreaming interface header descriptor*/ + mshd = (struct usb_ms_header_descriptor *)alts->extra; + if (alts->extralen < 7 || + mshd->bLength < 7 || + mshd->bDescriptorType != USB_DT_CS_INTERFACE || + mshd->bDescriptorSubtype != USB_MS_HEADER) + return -ENODEV; + /* must have the MIDIStreaming endpoint descriptor*/ + msepd = (struct usb_ms_endpoint_descriptor *)alts->endpoint[0].extra; + if (alts->endpoint[0].extralen < 4 || + msepd->bLength < 4 || + msepd->bDescriptorType != USB_DT_CS_ENDPOINT || + msepd->bDescriptorSubtype != UAC_MS_GENERAL || + msepd->bNumEmbMIDIJack < 1 || + msepd->bNumEmbMIDIJack > 16) + return -ENODEV; + + return create_any_midi_quirk(chip, iface, driver, NULL); +} + +static int create_auto_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver) +{ + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_endpoint_descriptor *epd; + int err; + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + + /* must have at least one bulk/interrupt endpoint for streaming */ + if (altsd->bNumEndpoints < 1) + return -ENODEV; + epd = get_endpoint(alts, 0); + if (!usb_endpoint_xfer_bulk(epd) && + !usb_endpoint_xfer_int(epd)) + return -ENODEV; + + switch (USB_ID_VENDOR(chip->usb_id)) { + case 0x0499: /* Yamaha */ + err = create_yamaha_midi_quirk(chip, iface, driver, alts); + if (err != -ENODEV) + return err; + break; + case 0x0582: /* Roland */ + err = create_roland_midi_quirk(chip, iface, driver, alts); + if (err != -ENODEV) + return err; + break; + } + + return create_std_midi_quirk(chip, iface, driver, alts); +} + +static int create_autodetect_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver) +{ + int err; + + err = create_auto_pcm_quirk(chip, iface, driver); + if (err == -ENODEV) + err = create_auto_midi_quirk(chip, iface, driver); + return err; +} + +static int create_autodetect_quirks(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + int probed_ifnum = get_iface_desc(iface->altsetting)->bInterfaceNumber; + int ifcount, ifnum, err; + + err = create_autodetect_quirk(chip, iface, driver); + if (err < 0) + return err; + + /* + * ALSA PCM playback/capture devices cannot be registered in two steps, + * so we have to claim the other corresponding interface here. + */ + ifcount = chip->dev->actconfig->desc.bNumInterfaces; + for (ifnum = 0; ifnum < ifcount; ifnum++) { + if (ifnum == probed_ifnum || quirk->ifnum >= 0) + continue; + iface = usb_ifnum_to_if(chip->dev, ifnum); + if (!iface || + usb_interface_claimed(iface) || + get_iface_desc(iface->altsetting)->bInterfaceClass != + USB_CLASS_VENDOR_SPEC) + continue; + + err = create_autodetect_quirk(chip, iface, driver); + if (err >= 0) + usb_driver_claim_interface(driver, iface, (void *)-1L); + } + + return 0; +} + /* * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. @@ -303,9 +514,11 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, static const quirk_func_t quirk_funcs[] = { [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, [QUIRK_COMPOSITE] = create_composite_quirk, + [QUIRK_AUTODETECT] = create_autodetect_quirks, [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, + [QUIRK_MIDI_ROLAND] = create_any_midi_quirk, [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, [QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk, diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 7db2f89..c4339f9 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -493,10 +493,10 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) altsd = get_iface_desc(alts); protocol = altsd->bInterfaceProtocol; /* skip invalid one */ - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + if (((altsd->bInterfaceClass != USB_CLASS_AUDIO || + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && + altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC)) && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && - altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || altsd->bNumEndpoints < 1 || le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) continue; @@ -512,6 +512,15 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) continue; + /* + * Roland audio streaming interfaces are marked with protocols + * 0/1/2, but are UAC 1 compatible. + */ + if (USB_ID_VENDOR(chip->usb_id) == 0x0582 && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + protocol <= 2) + protocol = UAC_VERSION_1; + chconfig = 0; /* get audio formats */ switch (protocol) { @@ -635,6 +644,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->protocol = protocol; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->channels = num_channels; if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) @@ -676,7 +686,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) } /* ok, let's parse further... */ - if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { + if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream) < 0) { kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index bc43bca..caabe9b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -72,9 +72,11 @@ struct snd_usb_audio { enum quirk_type { QUIRK_IGNORE_INTERFACE, QUIRK_COMPOSITE, + QUIRK_AUTODETECT, QUIRK_MIDI_STANDARD_INTERFACE, QUIRK_MIDI_FIXED_ENDPOINT, QUIRK_MIDI_YAMAHA, + QUIRK_MIDI_ROLAND, QUIRK_MIDI_MIDIMAN, QUIRK_MIDI_NOVATION, QUIRK_MIDI_RAW_BYTES, diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 9af7c1f..5a51b18 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -150,7 +150,7 @@ MODULE_AUTHOR("Karsten Wiese <annabellesgarden@yahoo.de>"); MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.8.7.2"); MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604), "NAME_ALLCAPS"(0x8001)(0x8005)(0x8007) }}"); +MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604),"NAME_ALLCAPS"(0x8001)(0x8005)(0x8007)}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S) { int i; for (i = 0; i < URBS_AsyncSeq; ++i) { - if (S[i].urb) { - usb_kill_urb(S->urb[i]); - usb_free_urb(S->urb[i]); - S->urb[i] = NULL; - } + usb_kill_urb(S->urb[i]); + usb_free_urb(S->urb[i]); + S->urb[i] = NULL; } kfree(S->buffer); } diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index b376532..63fb521 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { if (atomic_read(&subs->state) >= state_PRERUNNING) { + unsigned long flags; + + snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); } for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; @@ -695,9 +699,6 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) ((char*)(usbdata + i))[1] = ra[i].c2; usb_fill_bulk_urb(us->urb[i], usX2Y->dev, usb_sndbulkpipe(usX2Y->dev, 4), usbdata + i, 2, i_usX2Y_04Int, usX2Y); -#ifdef OLD_USB - us->urb[i]->transfer_flags = USB_QUEUE_BULK; -#endif } us->submitted = 0; us->len = NOOF_SETRATE_URBS; |